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Goip+Asterisk.Проблемы с распознаванием DTMF

Вопросы по использованию и настройке IP телефонов, шлюзов и всего прочего

Модераторы: april22, Zavr2008

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Roman_R
Сообщения: 18
Зарегистрирован: 18 мар 2015, 20:23

Goip+Asterisk.Проблемы с распознаванием DTMF

Сообщение Roman_R »

дравствуйте!

Столкнулся с такими проблемами при настройке связки Goip+Asterisk. Goip регистрируется на сервере Asterisk. Я звоню на шлюз по GSM. Он кидает на IVR, где осуществляется донабор номера. Так вот сигналы DTMF цифр 4 и 5 (да только эти ))) не распознаются сервером. Наткнулся на кучу статей и рекомендаций в гугле. Вот одна из них:

http://subnets.ru/blog/?p=1278

Перепробовал все что можно. за исключением инструкций по смене частот кодов и перекомпиляции сервера (Asterisk работает на NAS Synology ).

Настройки на сервере:

Код: Выделить всё

 [users](!)
            type=friend
            host=dynamic
            nat=yes
            canreinvite=no
            transport=udp
            qualify=yes
            disallow=all
            allow=alaw
            allow=ulaw
            callgroup=1
            pickupgroup=1
            deny=0.0.0.0/0.0.0.0
            permit=172.126.0.0/255.255.255.0
    [230](users)
            username=230
            secret=1234
            fromuser=230
            insecure=port,invite
            context=default
            dtmfmode=rfc2833
Вот что происходит при наборе цифр:

Код: Выделить всё

      == CDR updated on SIP/230-00000078
        -- Executing [0@menu:1] Goto("SIP/230-00000078", "dtmf-analys,s,1") in new stack
        -- Goto (dtmf-analys,s,1)
        -- Executing [s@dtmf-analys:1] SayDigits("SIP/230-00000078", "0") in new stack
        -- <SIP/230-00000078> Playing 'digits/0.alaw' (language 'en')
        -- Executing [s@dtmf-analys:2] Answer("SIP/230-00000078", "") in new stack
        -- Executing [s@dtmf-analys:3] BackGround("SIP/230-00000078", "beep") in new stack
        -- <SIP/230-00000078> Playing 'beep.alaw' (language 'en')
        -- Executing [s@dtmf-analys:4] Wait("SIP/230-00000078", "1") in new stack
        -- Executing [s@dtmf-analys:5] Read("SIP/230-00000078", "digito,,9") in new stack
        -- Accepting a maximum of 9 digits.
    [Mar 22 16:33:41] DTMF[20485]: channel.c:4151 __ast_read: DTMF begin '1' received on SIP/230-00000078
    [Mar 22 16:33:41] DTMF[20485]: channel.c:4155 __ast_read: DTMF begin ignored '1' on SIP/230-00000078
    [Mar 22 16:33:41] DTMF[20485]: channel.c:4066 __ast_read: DTMF end '1' received on SIP/230-00000078, duration 100 ms
    [Mar 22 16:33:41] DTMF[20485]: channel.c:4135 __ast_read: DTMF end passthrough '1' on SIP/230-00000078
    [Mar 22 16:33:42] DTMF[20485]: channel.c:4151 __ast_read: DTMF begin '2' received on SIP/230-00000078
    [Mar 22 16:33:42] DTMF[20485]: channel.c:4155 __ast_read: DTMF begin ignored '2' on SIP/230-00000078
    [Mar 22 16:33:42] DTMF[20485]: channel.c:4066 __ast_read: DTMF end '2' received on SIP/230-00000078, duration 100 ms
    [Mar 22 16:33:42] DTMF[20485]: channel.c:4135 __ast_read: DTMF end passthrough '2' on SIP/230-00000078
    [Mar 22 16:33:43] DTMF[20485]: channel.c:4151 __ast_read: DTMF begin '3' received on SIP/230-00000078
    [Mar 22 16:33:43] DTMF[20485]: channel.c:4155 __ast_read: DTMF begin ignored '3' on SIP/230-00000078
    [Mar 22 16:33:43] DTMF[20485]: channel.c:4066 __ast_read: DTMF end '3' received on SIP/230-00000078, duration 100 ms
    [Mar 22 16:33:43] DTMF[20485]: channel.c:4135 __ast_read: DTMF end passthrough '3' on SIP/230-00000078
    [Mar 22 16:33:47] DTMF[20485]: channel.c:4151 __ast_read: DTMF begin '6' received on SIP/230-00000078
    [Mar 22 16:33:47] DTMF[20485]: channel.c:4155 __ast_read: DTMF begin ignored '6' on SIP/230-00000078
    [Mar 22 16:33:47] DTMF[20485]: channel.c:4066 __ast_read: DTMF end '6' received on SIP/230-00000078, duration 100 ms
    [Mar 22 16:33:47] DTMF[20485]: channel.c:4135 __ast_read: DTMF end passthrough '6' on SIP/230-00000078
    [Mar 22 16:33:51] DTMF[20485]: channel.c:4151 __ast_read: DTMF begin '7' received on SIP/230-00000078
    [Mar 22 16:33:51] DTMF[20485]: channel.c:4155 __ast_read: DTMF begin ignored '7' on SIP/230-00000078
    [Mar 22 16:33:51] DTMF[20485]: channel.c:4066 __ast_read: DTMF end '7' received on SIP/230-00000078, duration 100 ms
    [Mar 22 16:33:51] DTMF[20485]: channel.c:4135 __ast_read: DTMF end passthrough '7' on SIP/230-00000078
    [Mar 22 16:33:52] DTMF[20485]: channel.c:4151 __ast_read: DTMF begin '8' received on SIP/230-00000078
    [Mar 22 16:33:52] DTMF[20485]: channel.c:4155 __ast_read: DTMF begin ignored '8' on SIP/230-00000078
    [Mar 22 16:33:52] DTMF[20485]: channel.c:4066 __ast_read: DTMF end '8' received on SIP/230-00000078, duration 100 ms
    [Mar 22 16:33:52] DTMF[20485]: channel.c:4135 __ast_read: DTMF end passthrough '8' on SIP/230-00000078
    [Mar 22 16:33:56] DTMF[20485]: channel.c:4151 __ast_read: DTMF begin '9' received on SIP/230-00000078
    [Mar 22 16:33:56] DTMF[20485]: channel.c:4155 __ast_read: DTMF begin ignored '9' on SIP/230-00000078
    [Mar 22 16:33:56] DTMF[20485]: channel.c:4066 __ast_read: DTMF end '9' received on SIP/230-00000078, duration 100 ms
    [Mar 22 16:33:56] DTMF[20485]: channel.c:4135 __ast_read: DTMF end passthrough '9' on SIP/230-00000078
        -- User entered '1236789'
        -- Executing [s@dtmf-analys:6] SayDigits("SIP/230-00000078", "1236789") in new stack
        -- <SIP/230-00000078> Playing 'digits/1.alaw' (language 'en')
        -- <SIP/230-00000078> Playing 'digits/2.alaw' (language 'en')
        -- <SIP/230-00000078> Playing 'digits/3.alaw' (language 'en')
        -- <SIP/230-00000078> Playing 'digits/6.alaw' (language 'en')
        -- <SIP/230-00000078> Playing 'digits/7.alaw' (language 'en')
        -- <SIP/230-00000078> Playing 'digits/8.alaw' (language 'en')
        -- <SIP/230-00000078> Playing 'digits/9.alaw' (language 'en')
        -- Executing [s@dtmf-analys:7] Goto("SIP/230-00000078", "collect") in new stack
        -- Goto (dtmf-analys,s,5)
        -- Executing [s@dtmf-analys:5] Read("SIP/230-00000078", "digito,,9") in new stack
        -- Accepting a maximum of 9 digits.
        -- User entered nothing.
        -- Executing [s@dtmf-analys:6] SayDigits("SIP/230-00000078", "") in new stack
        -- Executing [s@dtmf-analys:7] Goto("SIP/230-00000078", "collect") in new stack
        -- Goto (dtmf-analys,s,5)
        -- Executing [s@dtmf-analys:5] Read("SIP/230-00000078", "digito,,9") in new stack
        -- Accepting a maximum of 9 digits.
        -- User entered nothing.
        -- Executing [s@dtmf-analys:6] SayDigits("SIP/230-00000078", "") in new stack
        -- Executing [s@dtmf-analys:7] Goto("SIP/230-00000078", "collect") in new stack
        -- Goto (dtmf-analys,s,5)
        -- Executing [s@dtmf-analys:5] Read("SIP/230-00000078", "digito,,9") in new stack
        -- Accepting a maximum of 9 digits.
Так обрабатываются нажатия:

Код: Выделить всё

[dtmf-analys]
exten => s,1,SayDigits(0) [0-чтобы знать что мы в нужном контексте]
; exten => s,n,Answer()
exten => s,n,Wait(1)
exten => s,n(collect),Read(digito,,9)
exten => s,n,SayDigits(${digito})
exten => s,n,GoTo(collect)
exten => s,n,Hangup
; exten => t,1,Goto(s,1)
Замечание: если звонить с очень древнего мобильного телефона, все корректно работает. Также если звонить со софтфонов внутри сети. Проблема только при звонке с мобильного на Goip. Пробовал разные типы DTMF - не помогло. Может ли дело быть в частотах DTMF?
Вложения
goip-3.JPG
goip-3.JPG (16.8 КБ) 16591 просмотр
goip-2.JPG
goip-1.JPG
goip-1.JPG (56.75 КБ) 16591 просмотр
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ded
Сообщения: 15618
Зарегистрирован: 26 авг 2010, 19:00

Re: Goip+Asterisk.Проблемы с распознаванием DTMF

Сообщение ded »

Можно попытаться поднять громкость на входящем (если GoIP позволяет)
Вернуться к началу
Roman_R
Сообщения: 18
Зарегистрирован: 18 мар 2015, 20:23

Re: Goip+Asterisk.Проблемы с распознаванием DTMF

Сообщение Roman_R »

Регулировок Input Gain и Output Gain не нашел. Прошивка: GHSFVT-1.1-22. GSM1 Model G610.
Опция relaxdtmf=yes. Не могу понять чего такого есть в старом древнем мобильнике и что может быть в современном.
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MIKS
Сообщения: 80
Зарегистрирован: 12 мар 2014, 13:43

Re: Goip+Asterisk.Проблемы с распознаванием DTMF

Сообщение MIKS »

Сделайте передачу дтмф в SIP INFO и будет счастье.
RFC2833 - в RTP передает и тут может зависеть действительно даже от машинки на которой астер крутиться
Вернуться к началу
Roman_R
Сообщения: 18
Зарегистрирован: 18 мар 2015, 20:23

Re: Goip+Asterisk.Проблемы с распознаванием DTMF

Сообщение Roman_R »

Пробовал - не помогает. с SIP INFO не работает вообще никак :) Точно такой же дебаг.
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MIKS
Сообщения: 80
Зарегистрирован: 12 мар 2014, 13:43

Re: Goip+Asterisk.Проблемы с распознаванием DTMF

Сообщение MIKS »

Не может дебаг быть таким же(хотя китайцы...)
Там ДТМФ идет в сигнализации. Доказательство - поставьте sip set debug и уберите relax dtmf
Есть конечно сомнения как в GSM передача реализована но скорее всего явно вне полосы
Скриншоты тогда настройки GOIP желательны
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Roman_R
Сообщения: 18
Зарегистрирован: 18 мар 2015, 20:23

Re: Goip+Asterisk.Проблемы с распознаванием DTMF

Сообщение Roman_R »

Настройки на сервере:

Код: Выделить всё

[232]
        username=232
        type=friend
        secret=1234
        host=dynamic
        context=default
        nat=yes
        canreinvite=no
        disallow=all
        dtmfmode=info
        allow=ulaw
        allow=alaw
        callgroup=1
        pickupgroup=1
        deny=0.0.0.0/0.0.0.0
        permit=172.126.0.0/255.255.255.0
Звоним..

Код: Выделить всё

DSS*CLI> sip set debug peer 232
SIP Debugging Enabled for IP: 172.126.0.3

<--- SIP read from UDP:172.126.0.3:5060 --->

<------------->

<--- SIP read from UDP:172.126.0.3:5060 --->
REGISTER sip:172.126.0.100 SIP/2.0
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK1313704735
From: "GOIP" <sip:232@172.126.0.100>;tag=173694777
To: "GOIP" <sip:232@172.126.0.100>
Call-ID: 585551672@172.126.0.3
CSeq: 2492 REGISTER
Contact: <sip:232@172.126.0.3:5060>;expires=60
Authorization: Digest username="232", realm="asterisk", nonce="2eccaa49", uri="sip:172.126.0.100", response="ccd1febcd23ff9addbc7e8d378f84ef0", algorithm=MD5
Max-Forwards: 30
User-Agent: dble
Expires: 60
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to 172.126.0.3:5060 (NAT)
[Mar 25 14:23:39] NOTICE[13340]: chan_sip.c:15060 check_auth: Correct auth, but based on stale nonce received from '"GOIP" <sip:232@172.126.0.100>;tag=173694777'

<--- Transmitting (NAT) to 172.126.0.3:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK1313704735;received=172.126.0.3;rport=5060
From: "GOIP" <sip:232@172.126.0.100>;tag=173694777
To: "GOIP" <sip:232@172.126.0.100>;tag=as677354cb
Call-ID: 585551672@172.126.0.3
CSeq: 2492 REGISTER
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="286735b0", stale=true
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '585551672@172.126.0.3' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:172.126.0.3:5060 --->
REGISTER sip:172.126.0.100 SIP/2.0
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK255050095
From: "GOIP" <sip:232@172.126.0.100>;tag=173694777
To: "GOIP" <sip:232@172.126.0.100>
Call-ID: 585551672@172.126.0.3
CSeq: 2493 REGISTER
Contact: <sip:232@172.126.0.3:5060>;expires=60
Authorization: Digest username="232", realm="asterisk", nonce="286735b0", uri="sip:172.126.0.100", response="4601ab351001cdaf5ca5b632ec7111bf", algorithm=MD5
Max-Forwards: 30
User-Agent: dble
Expires: 60
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to 172.126.0.3:5060 (NAT)

<--- Transmitting (NAT) to 172.126.0.3:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK255050095;received=172.126.0.3;rport=5060
From: "GOIP" <sip:232@172.126.0.100>;tag=173694777
To: "GOIP" <sip:232@172.126.0.100>;tag=as677354cb
Call-ID: 585551672@172.126.0.3
CSeq: 2493 REGISTER
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 60
Contact: <sip:232@172.126.0.3:5060>;expires=60
Date: Wed, 25 Mar 2015 08:23:39 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '585551672@172.126.0.3' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:172.126.0.3:5060 --->
INVITE sip:200@172.126.0.100:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.126.0.3:5060;rport;branch=z9hG4bK876165734
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=60710202
To: <sip:200@172.126.0.100>
Call-ID: 401485700@172.126.0.3
CSeq: 222 INVITE
Contact: <sip:232@172.126.0.3:5060>
Max-Forwards: 30
User-Agent: dble
Remote-Party-ID: "77051842286" <sip:77051842286@172.126.0.100>;party=calling;screen=no;privacy=off
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REFER, REGISTER, MESSAGE, INFO, SUBSCRIBE
Content-Type: application/sdp
Content-Length: 174

v=0
o=dble 1427271789 1427271789 IN IP4 172.126.0.3
s=dble
c=IN IP4 172.126.0.3
t=0 0
m=audio 10000 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
<------------->
--- (13 headers 9 lines) ---
Sending to 172.126.0.3:5060 (NAT)
Using INVITE request as basis request - 401485700@172.126.0.3
Found peer '232' for '232' from 172.126.0.3:5060

<--- Reliably Transmitting (NAT) to 172.126.0.3:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK876165734;received=172.126.0.3;rport=5060
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=60710202
To: <sip:200@172.126.0.100>;tag=as1b16a958
Call-ID: 401485700@172.126.0.3
CSeq: 222 INVITE
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4499fe65"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '401485700@172.126.0.3' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:172.126.0.3:5060 --->
ACK sip:200@172.126.0.100:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.126.0.3:5060;rport;branch=z9hG4bK876165734
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=60710202
To: <sip:200@172.126.0.100>;tag=as1b16a958
Call-ID: 401485700@172.126.0.3
CSeq: 222 ACK
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:172.126.0.3:5060 --->
INVITE sip:200@172.126.0.100:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.126.0.3:5060;rport;branch=z9hG4bK1165155094
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=60710202
To: <sip:200@172.126.0.100>
Call-ID: 401485700@172.126.0.3
CSeq: 223 INVITE
Contact: <sip:232@172.126.0.3:5060>
Authorization: Digest username="232", realm="asterisk", nonce="4499fe65", uri="sip:200@172.126.0.100", response="d6a2988e1f7905087c94e8cb905a0928", algorithm=MD5
Max-Forwards: 30
User-Agent: dble
Remote-Party-ID: "77051842286" <sip:77051842286@172.126.0.100>;party=calling;screen=no;privacy=off
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REFER, REGISTER, MESSAGE, INFO, SUBSCRIBE
Content-Type: application/sdp
Content-Length: 174

v=0
o=dble 1427271789 1427271789 IN IP4 172.126.0.3
s=dble
c=IN IP4 172.126.0.3
t=0 0
m=audio 10000 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
<------------->
--- (14 headers 9 lines) ---
Sending to 172.126.0.3:5060 (NAT)
Using INVITE request as basis request - 401485700@172.126.0.3
Found peer '232' for '232' from 172.126.0.3:5060
  == Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 172.126.0.3:10000
Looking for 200 in default (domain 172.126.0.100)
list_route: hop: <sip:232@172.126.0.3:5060>

<--- Transmitting (NAT) to 172.126.0.3:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK1165155094;received=172.126.0.3;rport=5060
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=60710202
To: <sip:200@172.126.0.100>
Call-ID: 401485700@172.126.0.3
CSeq: 223 INVITE
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:200@172.126.0.100:5060>
Content-Length: 0


<------------>
    -- Executing [200@default:1] Goto("SIP/232-000000a9", "menu,s,1") in new stack
    -- Goto (menu,s,1)
    -- Executing [s@menu:1] Answer("SIP/232-000000a9", "") in new stack
Audio is at 13684
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP

<--- Reliably Transmitting (NAT) to 172.126.0.3:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK1165155094;received=172.126.0.3;rport=5060
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=60710202
To: <sip:200@172.126.0.100>;tag=as33b0fccc
Call-ID: 401485700@172.126.0.3
CSeq: 223 INVITE
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:200@172.126.0.100:5060>
ontent-Type: application/sdp
Content-Length: 234

v=0
o=root 1534590263 1534590263 IN IP4 172.126.0.100
s=Asterisk PBX 1.8.25.0
c=IN IP4 172.126.0.100
t=0 0
m=audio 13684 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
    -- Executing [s@menu:2] BackGround("SIP/232-000000a9", "beep") in new stack

<--- SIP read from UDP:172.126.0.3:5060 --->
ACK sip:200@172.126.0.100:5060 SIP/2.0
Via: SIP/2.0/UDP 172.126.0.3:5060;rport;branch=z9hG4bK1743015711
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=60710202
To: <sip:200@172.126.0.100>;tag=as33b0fccc
Call-ID: 401485700@172.126.0.3
CSeq: 223 ACK
Contact: <sip:232@172.126.0.3:5060>
Max-Forwards: 30
User-Agent: dble
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
    -- <SIP/232-000000a9> Playing 'beep.alaw' (language 'en')
    -- Executing [s@menu:3] WaitExten("SIP/232-000000a9", "5") in new stack

<--- SIP read from UDP:172.126.0.3:5060 --->
INFO sip:200@172.126.0.100:5060 SIP/2.0
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK270398163
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=60710202
To: <sip:200@172.126.0.100>;tag=as33b0fccc
Call-ID: 401485700@172.126.0.3
CSeq: 224 INFO
Contact: <sip:232@172.126.0.3:5060>
Max-Forwards: 30
User-Agent: dble
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=0
Duration=160
<------------->
--- (11 headers 2 lines) ---
Receiving INFO!
* DTMF-relay event received: 0

<--- Transmitting (NAT) to 172.126.0.3:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK270398163;received=172.126.0.3;rport=5060
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=60710202
To: <sip:200@172.126.0.100>;tag=as33b0fccc
Call-ID: 401485700@172.126.0.3
CSeq: 224 INFO
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[Mar 25 14:23:43] DTMF[29117]: channel.c:4066 __ast_read: DTMF end '0' received on SIP/232-000000a9, duration 160 ms
[Mar 25 14:23:43] DTMF[29117]: channel.c:4135 __ast_read: DTMF end passthrough '0' on SIP/232-000000a9
  == CDR updated on SIP/232-000000a9
    -- Executing [0@menu:1] Goto("SIP/232-000000a9", "dtmf-analys,s,1") in new stack
    -- Goto (dtmf-analys,s,1)
    -- Executing [s@dtmf-analys:1] SayDigits("SIP/232-000000a9", "0") in new stack
    -- <SIP/232-000000a9> Playing 'digits/0.alaw' (language 'en')
    -- Executing [s@dtmf-analys:2] Answer("SIP/232-000000a9", "") in new stack
    -- Executing [s@dtmf-analys:3] BackGround("SIP/232-000000a9", "beep") in new stack
    -- <SIP/232-000000a9> Playing 'beep.alaw' (language 'en')
    -- Executing [s@dtmf-analys:4] Wait("SIP/232-000000a9", "1") in new stack
    -- Executing [s@dtmf-analys:5] Read("SIP/232-000000a9", "digito,,9") in new stack
    -- Accepting a maximum of 9 digits.

<--- SIP read from UDP:172.126.0.3:5060 --->
INFO sip:200@172.126.0.100:5060 SIP/2.0
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK1543327375
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=60710202
To: <sip:200@172.126.0.100>;tag=as33b0fccc
Call-ID: 401485700@172.126.0.3
CSeq: 225 INFO
Contact: <sip:232@172.126.0.3:5060>
Max-Forwards: 30
User-Agent: dble
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=2
Duration=160
<------------->
--- (11 headers 2 lines) ---
Receiving INFO!
* DTMF-relay event received: 2

<--- Transmitting (NAT) to 172.126.0.3:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK1543327375;received=172.126.0.3;rport=5060
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=60710202
To: <sip:200@172.126.0.100>;tag=as33b0fccc
Call-ID: 401485700@172.126.0.3
CSeq: 225 INFO
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[Mar 25 14:23:47] DTMF[29117]: channel.c:4066 __ast_read: DTMF end '2' received on SIP/232-000000a9, duration 160 ms
[Mar 25 14:23:47] DTMF[29117]: channel.c:4135 __ast_read: DTMF end passthrough '2' on SIP/232-000000a9

<--- SIP read from UDP:172.126.0.3:5060 --->
INFO sip:200@172.126.0.100:5060 SIP/2.0
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK1600179442
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=60710202
To: <sip:200@172.126.0.100>;tag=as33b0fccc
Call-ID: 401485700@172.126.0.3
CSeq: 226 INFO
Contact: <sip:232@172.126.0.3:5060>
Max-Forwards: 30
User-Agent: dble
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=2
Duration=160
<------------->
--- (11 headers 2 lines) ---
Receiving INFO!
* DTMF-relay event received: 2

<--- Transmitting (NAT) to 172.126.0.3:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK1600179442;received=172.126.0.3;rport=5060
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=60710202
To: <sip:200@172.126.0.100>;tag=as33b0fccc
Call-ID: 401485700@172.126.0.3
CSeq: 226 INFO
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[Mar 25 14:23:47] DTMF[29117]: channel.c:4066 __ast_read: DTMF end '2' received on SIP/232-000000a9, duration 160 ms
[Mar 25 14:23:47] DTMF[29117]: channel.c:4135 __ast_read: DTMF end passthrough '2' on SIP/232-000000a9

<--- SIP read from UDP:172.126.0.3:5060 --->
INFO sip:200@172.126.0.100:5060 SIP/2.0
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK820912792
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=60710202
To: <sip:200@172.126.0.100>;tag=as33b0fccc
Call-ID: 401485700@172.126.0.3
CSeq: 227 INFO
Contact: <sip:232@172.126.0.3:5060>
Max-Forwards: 30
User-Agent: dble
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=3
Duration=160
<------------->
--- (11 headers 2 lines) ---
Receiving INFO!
* DTMF-relay event received: 3

<--- Transmitting (NAT) to 172.126.0.3:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK820912792;received=172.126.0.3;rport=5060
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=60710202
To: <sip:200@172.126.0.100>;tag=as33b0fccc
Call-ID: 401485700@172.126.0.3
CSeq: 227 INFO
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[Mar 25 14:23:49] DTMF[29117]: channel.c:4066 __ast_read: DTMF end '3' received on SIP/232-000000a9, duration 160 ms
[Mar 25 14:23:49] DTMF[29117]: channel.c:4135 __ast_read: DTMF end passthrough '3' on SIP/232-000000a9

<--- SIP read from UDP:172.126.0.3:5060 --->
INFO sip:200@172.126.0.100:5060 SIP/2.0
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK1515561016
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=60710202
To: <sip:200@172.126.0.100>;tag=as33b0fccc
Call-ID: 401485700@172.126.0.3
CSeq: 228 INFO
Contact: <sip:232@172.126.0.3:5060>
Max-Forwards: 30
User-Agent: dble
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=6
Duration=160
<------------->
--- (11 headers 2 lines) ---
Receiving INFO!
* DTMF-relay event received: 6

<--- Transmitting (NAT) to 172.126.0.3:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK1515561016;received=172.126.0.3;rport=5060
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=60710202
To: <sip:200@172.126.0.100>;tag=as33b0fccc
Call-ID: 401485700@172.126.0.3
CSeq: 228 INFO
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[Mar 25 14:23:52] DTMF[29117]: channel.c:4066 __ast_read: DTMF end '6' received on SIP/232-000000a9, duration 160 ms
[Mar 25 14:23:52] DTMF[29117]: channel.c:4135 __ast_read: DTMF end passthrough '6' on SIP/232-000000a9

<--- SIP read from UDP:172.126.0.3:5060 --->
INFO sip:200@172.126.0.100:5060 SIP/2.0
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK1588418680
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=60710202
To: <sip:200@172.126.0.100>;tag=as33b0fccc
Call-ID: 401485700@172.126.0.3
CSeq: 229 INFO
Contact: <sip:232@172.126.0.3:5060>
Max-Forwards: 30
User-Agent: dble
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=7
Duration=160
<------------->
--- (11 headers 2 lines) ---
Receiving INFO!
* DTMF-relay event received: 7

<--- Transmitting (NAT) to 172.126.0.3:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK1588418680;received=172.126.0.3;rport=5060
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=60710202
To: <sip:200@172.126.0.100>;tag=as33b0fccc
Call-ID: 401485700@172.126.0.3
CSeq: 229 INFO
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[Mar 25 14:23:53] DTMF[29117]: channel.c:4066 __ast_read: DTMF end '7' received on SIP/232-000000a9, duration 160 ms
[Mar 25 14:23:53] DTMF[29117]: channel.c:4135 __ast_read: DTMF end passthrough '7' on SIP/232-000000a9

<--- SIP read from UDP:172.126.0.3:5060 --->
INFO sip:200@172.126.0.100:5060 SIP/2.0
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK718654854
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=60710202
To: <sip:200@172.126.0.100>;tag=as33b0fccc
Call-ID: 401485700@172.126.0.3
CSeq: 230 INFO
Contact: <sip:232@172.126.0.3:5060>
Max-Forwards: 30
User-Agent: dble
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=8
Duration=160
<------------->
--- (11 headers 2 lines) ---
Receiving INFO!
* DTMF-relay event received: 8

<--- Transmitting (NAT) to 172.126.0.3:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK718654854;received=172.126.0.3;rport=5060
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=60710202
To: <sip:200@172.126.0.100>;tag=as33b0fccc
Call-ID: 401485700@172.126.0.3
CSeq: 230 INFO
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[Mar 25 14:23:55] DTMF[29117]: channel.c:4066 __ast_read: DTMF end '8' received on SIP/232-000000a9, duration 160 ms
[Mar 25 14:23:55] DTMF[29117]: channel.c:4135 __ast_read: DTMF end passthrough '8' on SIP/232-000000a9

<--- SIP read from UDP:172.126.0.3:5060 --->
INFO sip:200@172.126.0.100:5060 SIP/2.0
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK832550251
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=60710202
To: <sip:200@172.126.0.100>;tag=as33b0fccc
Call-ID: 401485700@172.126.0.3
CSeq: 231 INFO
Contact: <sip:232@172.126.0.3:5060>
Max-Forwards: 30
User-Agent: dble
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=9
Duration=160
<------------->
--- (11 headers 2 lines) ---
Receiving INFO!
* DTMF-relay event received: 9

<--- Transmitting (NAT) to 172.126.0.3:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK832550251;received=172.126.0.3;rport=5060
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=60710202
To: <sip:200@172.126.0.100>;tag=as33b0fccc
Call-ID: 401485700@172.126.0.3
CSeq: 231 INFO
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[Mar 25 14:23:56] DTMF[29117]: channel.c:4066 __ast_read: DTMF end '9' received on SIP/232-000000a9, duration 160 ms
[Mar 25 14:23:56] DTMF[29117]: channel.c:4135 __ast_read: DTMF end passthrough '9' on SIP/232-000000a9
    -- User entered '2236789'
    -- Executing [s@dtmf-analys:6] SayDigits("SIP/232-000000a9", "2236789") in new stack
    -- <SIP/232-000000a9> Playing 'digits/2.alaw' (language 'en')
    -- <SIP/232-000000a9> Playing 'digits/2.alaw' (language 'en')
    -- <SIP/232-000000a9> Playing 'digits/3.alaw' (language 'en')
    -- <SIP/232-000000a9> Playing 'digits/6.alaw' (language 'en')
    -- <SIP/232-000000a9> Playing 'digits/7.alaw' (language 'en')
    -- <SIP/232-000000a9> Playing 'digits/8.alaw' (language 'en')
    -- <SIP/232-000000a9> Playing 'digits/9.alaw' (language 'en')
    -- Executing [s@dtmf-analys:7] Goto("SIP/232-000000a9", "collect") in new stack
    -- Goto (dtmf-analys,s,5)
    -- Executing [s@dtmf-analys:5] Read("SIP/232-000000a9", "digito,,9") in new stack
    -- Accepting a maximum of 9 digits.

<--- SIP read from UDP:172.126.0.3:5060 --->
INFO sip:200@172.126.0.100:5060 SIP/2.0
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK922076461
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=60710202
To: <sip:200@172.126.0.100>;tag=as33b0fccc
Call-ID: 401485700@172.126.0.3
CSeq: 232 INFO
Contact: <sip:232@172.126.0.3:5060>
Max-Forwards: 30
User-Agent: dble
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=1
Duration=160
<------------->
--- (11 headers 2 lines) ---
Receiving INFO!
* DTMF-relay event received: 1

<--- Transmitting (NAT) to 172.126.0.3:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK922076461;received=172.126.0.3;rport=5060
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=60710202
To: <sip:200@172.126.0.100>;tag=as33b0fccc
Call-ID: 401485700@172.126.0.3
CSeq: 232 INFO
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[Mar 25 14:24:09] DTMF[29117]: channel.c:4066 __ast_read: DTMF end '1' received on SIP/232-000000a9, duration 160 ms
[Mar 25 14:24:09] DTMF[29117]: channel.c:4135 __ast_read: DTMF end passthrough '1' on SIP/232-000000a9

<--- SIP read from UDP:172.126.0.3:5060 --->
REGISTER sip:172.126.0.100 SIP/2.0
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK1986421456
From: "GOIP" <sip:232@172.126.0.100>;tag=841904062
To: "GOIP" <sip:232@172.126.0.100>
Call-ID: 585551672@172.126.0.3
CSeq: 2494 REGISTER
Contact: <sip:232@172.126.0.3:5060>;expires=60
Authorization: Digest username="232", realm="asterisk", nonce="286735b0", uri="sip:172.126.0.100", response="4601ab351001cdaf5ca5b632ec7111bf", algorithm=MD5
Max-Forwards: 30
User-Agent: dble
Expires: 60
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to 172.126.0.3:5060 (NAT)
[Mar 25 14:24:09] NOTICE[13340]: chan_sip.c:15060 check_auth: Correct auth, but based on stale nonce received from '"GOIP" <sip:232@172.126.0.100>;tag=841904062'

<--- Transmitting (NAT) to 172.126.0.3:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK1986421456;received=172.126.0.3;rport=5060
From: "GOIP" <sip:232@172.126.0.100>;tag=841904062
To: "GOIP" <sip:232@172.126.0.100>;tag=as677354cb
Call-ID: 585551672@172.126.0.3
CSeq: 2494 REGISTER
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="18cdd056", stale=true
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '585551672@172.126.0.3' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:172.126.0.3:5060 --->
REGISTER sip:172.126.0.100 SIP/2.0
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK81997278
From: "GOIP" <sip:232@172.126.0.100>;tag=841904062
To: "GOIP" <sip:232@172.126.0.100>
Call-ID: 585551672@172.126.0.3
CSeq: 2495 REGISTER
Contact: <sip:232@172.126.0.3:5060>;expires=60
Authorization: Digest username="232", realm="asterisk", nonce="18cdd056", uri="sip:172.126.0.100", response="d7b40832a8f7b412989aabbcd1003bfd", algorithm=MD5
Max-Forwards: 30
User-Agent: dble
Expires: 60
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to 172.126.0.3:5060 (NAT)

<--- Transmitting (NAT) to 172.126.0.3:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK81997278;received=172.126.0.3;rport=5060
From: "GOIP" <sip:232@172.126.0.100>;tag=841904062
To: "GOIP" <sip:232@172.126.0.100>;tag=as677354cb
Call-ID: 585551672@172.126.0.3
CSeq: 2495 REGISTER
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 60
Contact: <sip:232@172.126.0.3:5060>;expires=60
Date: Wed, 25 Mar 2015 08:24:09 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '585551672@172.126.0.3' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:172.126.0.3:5060 --->
INFO sip:200@172.126.0.100:5060 SIP/2.0
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK1064221101
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=60710202
To: <sip:200@172.126.0.100>;tag=as33b0fccc
Call-ID: 401485700@172.126.0.3
CSeq: 233 INFO
Contact: <sip:232@172.126.0.3:5060>
Max-Forwards: 30
User-Agent: dble
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=2
Duration=160
<------------->
--- (11 headers 2 lines) ---
Receiving INFO!
* DTMF-relay event received: 2

<--- Transmitting (NAT) to 172.126.0.3:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK1064221101;received=172.126.0.3;rport=5060
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=60710202
To: <sip:200@172.126.0.100>;tag=as33b0fccc
Call-ID: 401485700@172.126.0.3
CSeq: 233 INFO
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[Mar 25 14:24:10] DTMF[29117]: channel.c:4066 __ast_read: DTMF end '2' received on SIP/232-000000a9, duration 160 ms
[Mar 25 14:24:10] DTMF[29117]: channel.c:4135 __ast_read: DTMF end passthrough '2' on SIP/232-000000a9

<--- SIP read from UDP:172.126.0.3:5060 --->
INFO sip:200@172.126.0.100:5060 SIP/2.0
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK1823902517
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=60710202
To: <sip:200@172.126.0.100>;tag=as33b0fccc
Call-ID: 401485700@172.126.0.3
CSeq: 234 INFO
Contact: <sip:232@172.126.0.3:5060>
Max-Forwards: 30
User-Agent: dble
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=3
Duration=160
<------------->
--- (11 headers 2 lines) ---
Receiving INFO!
* DTMF-relay event received: 3

<--- Transmitting (NAT) to 172.126.0.3:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK1823902517;received=172.126.0.3;rport=5060
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=60710202
To: <sip:200@172.126.0.100>;tag=as33b0fccc
Call-ID: 401485700@172.126.0.3
CSeq: 234 INFO
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[Mar 25 14:24:12] DTMF[29117]: channel.c:4066 __ast_read: DTMF end '3' received on SIP/232-000000a9, duration 160 ms
[Mar 25 14:24:12] DTMF[29117]: channel.c:4135 __ast_read: DTMF end passthrough '3' on SIP/232-000000a9

<--- SIP read from UDP:172.126.0.3:5060 --->
INFO sip:200@172.126.0.100:5060 SIP/2.0
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK1028463445
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=60710202
To: <sip:200@172.126.0.100>;tag=as33b0fccc
Call-ID: 401485700@172.126.0.3
CSeq: 235 INFO
Contact: <sip:232@172.126.0.3:5060>
Max-Forwards: 30
User-Agent: dble
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=6
Duration=160
<------------->
--- (11 headers 2 lines) ---
Receiving INFO!
* DTMF-relay event received: 6

<--- Transmitting (NAT) to 172.126.0.3:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK1028463445;received=172.126.0.3;rport=5060
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=60710202
To: <sip:200@172.126.0.100>;tag=as33b0fccc
Call-ID: 401485700@172.126.0.3
CSeq: 235 INFO
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[Mar 25 14:24:17] DTMF[29117]: channel.c:4066 __ast_read: DTMF end '6' received on SIP/232-000000a9, duration 160 ms
[Mar 25 14:24:17] DTMF[29117]: channel.c:4135 __ast_read: DTMF end passthrough '6' on SIP/232-000000a9

<--- SIP read from UDP:172.126.0.3:5060 --->
INFO sip:200@172.126.0.100:5060 SIP/2.0
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK2097769386
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=60710202
To: <sip:200@172.126.0.100>;tag=as33b0fccc
Call-ID: 401485700@172.126.0.3
CSeq: 236 INFO
Contact: <sip:232@172.126.0.3:5060>
Max-Forwards: 30
User-Agent: dble
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=7
Duration=160
<------------->
--- (11 headers 2 lines) ---
Receiving INFO!
* DTMF-relay event received: 7

<--- Transmitting (NAT) to 172.126.0.3:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK2097769386;received=172.126.0.3;rport=5060
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=60710202
To: <sip:200@172.126.0.100>;tag=as33b0fccc
Call-ID: 401485700@172.126.0.3
CSeq: 236 INFO
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[Mar 25 14:24:18] DTMF[29117]: channel.c:4066 __ast_read: DTMF end '7' received on SIP/232-000000a9, duration 160 ms
[Mar 25 14:24:18] DTMF[29117]: channel.c:4135 __ast_read: DTMF end passthrough '7' on SIP/232-000000a9

<--- SIP read from UDP:172.126.0.3:5060 --->
INFO sip:200@172.126.0.100:5060 SIP/2.0
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK2039437795
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=60710202
To: <sip:200@172.126.0.100>;tag=as33b0fccc
Call-ID: 401485700@172.126.0.3
CSeq: 237 INFO
Contact: <sip:232@172.126.0.3:5060>
Max-Forwards: 30
User-Agent: dble
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=8
Duration=160
<------------->
--- (11 headers 2 lines) ---
Receiving INFO!
* DTMF-relay event received: 8

<--- Transmitting (NAT) to 172.126.0.3:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK2039437795;received=172.126.0.3;rport=5060
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=60710202
To: <sip:200@172.126.0.100>;tag=as33b0fccc
Call-ID: 401485700@172.126.0.3
CSeq: 237 INFO
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[Mar 25 14:24:19] DTMF[29117]: channel.c:4066 __ast_read: DTMF end '8' received on SIP/232-000000a9, duration 160 ms
[Mar 25 14:24:19] DTMF[29117]: channel.c:4135 __ast_read: DTMF end passthrough '8' on SIP/232-000000a9

<--- SIP read from UDP:172.126.0.3:5060 --->
INFO sip:200@172.126.0.100:5060 SIP/2.0
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK826376748
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=60710202
To: <sip:200@172.126.0.100>;tag=as33b0fccc
Call-ID: 401485700@172.126.0.3
CSeq: 238 INFO
Contact: <sip:232@172.126.0.3:5060>
Max-Forwards: 30
User-Agent: dble
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=9
Duration=160
<------------->
--- (11 headers 2 lines) ---
Receiving INFO!
* DTMF-relay event received: 9

<--- Transmitting (NAT) to 172.126.0.3:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK826376748;received=172.126.0.3;rport=5060
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=60710202
To: <sip:200@172.126.0.100>;tag=as33b0fccc
Call-ID: 401485700@172.126.0.3
CSeq: 238 INFO
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[Mar 25 14:24:21] DTMF[29117]: channel.c:4066 __ast_read: DTMF end '9' received on SIP/232-000000a9, duration 160 ms
[Mar 25 14:24:21] DTMF[29117]: channel.c:4135 __ast_read: DTMF end passthrough '9' on SIP/232-000000a9
[Mar 25 14:24:25] NOTICE[13340]: chan_sip.c:13654 sip_reregister:    -- Re-registration for  133331@sip.zadarma.com
       > doing dnsmgr_lookup for 'sip.zadarma.com'
       > ast_get_srv: SRV lookup for '_sip._udp.sip.zadarma.com' mapped to host sipfr.zadarma.com, port 5060
[Mar 25 14:24:25] NOTICE[13340]: chan_sip.c:13654 sip_reregister:    -- Re-registration for  133331@sip.zadarma.com
       > doing dnsmgr_lookup for 'sip.zadarma.com'
       > ast_get_srv: SRV lookup for '_sip._udp.sip.zadarma.com' mapped to host sipfr.zadarma.com, port 5060
       > doing dnsmgr_lookup for 'sip.zadarma.com'
       > ast_get_srv: SRV lookup for '_sip._udp.sip.zadarma.com' mapped to host sipfr.zadarma.com, port 5060
       > doing dnsmgr_lookup for 'sip.zadarma.com'
       > ast_get_srv: SRV lookup for '_sip._udp.sip.zadarma.com' mapped to host sipfr.zadarma.com, port 5060
[Mar 25 14:24:25] NOTICE[13340]: chan_sip.c:21548 handle_response_register: Outbound Registration: Expiry for sip.zadarma.com is 120 sec (Scheduling reregistration in 105 s)
[Mar 25 14:24:26] NOTICE[13340]: chan_sip.c:21548 handle_response_register: Outbound Registration: Expiry for sip.zadarma.com is 120 sec (Scheduling reregistration in 105 s)
    -- User entered '1236789'
    -- Executing [s@dtmf-analys:6] SayDigits("SIP/232-000000a9", "1236789") in new stack
    -- <SIP/232-000000a9> Playing 'digits/1.alaw' (language 'en')
    -- <SIP/232-000000a9> Playing 'digits/2.alaw' (language 'en')
    -- <SIP/232-000000a9> Playing 'digits/3.alaw' (language 'en')
    -- <SIP/232-000000a9> Playing 'digits/6.alaw' (language 'en')
    -- <SIP/232-000000a9> Playing 'digits/7.alaw' (language 'en')
    -- <SIP/232-000000a9> Playing 'digits/8.alaw' (language 'en')
    -- <SIP/232-000000a9> Playing 'digits/9.alaw' (language 'en')
    -- Executing [s@dtmf-analys:7] Goto("SIP/232-000000a9", "collect") in new stack
    -- Goto (dtmf-analys,s,5)
    -- Executing [s@dtmf-analys:5] Read("SIP/232-000000a9", "digito,,9") in new stack
    -- Accepting a maximum of 9 digits.

<--- SIP read from UDP:172.126.0.3:5060 --->
INFO sip:200@172.126.0.100:5060 SIP/2.0
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK1457898491
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=60710202
To: <sip:200@172.126.0.100>;tag=as33b0fccc
Call-ID: 401485700@172.126.0.3
CSeq: 239 INFO
Contact: <sip:232@172.126.0.3:5060>
Max-Forwards: 30
User-Agent: dble
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=1
Duration=160
<------------->
--- (11 headers 2 lines) ---
Receiving INFO!
* DTMF-relay event received: 1

<--- Transmitting (NAT) to 172.126.0.3:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK1457898491;received=172.126.0.3;rport=5060
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=60710202
To: <sip:200@172.126.0.100>;tag=as33b0fccc
Call-ID: 401485700@172.126.0.3
CSeq: 239 INFO
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[Mar 25 14:24:36] DTMF[29117]: channel.c:4066 __ast_read: DTMF end '1' received on SIP/232-000000a9, duration 160 ms
[Mar 25 14:24:36] DTMF[29117]: channel.c:4135 __ast_read: DTMF end passthrough '1' on SIP/232-000000a9

<--- SIP read from UDP:172.126.0.3:5060 --->
INFO sip:200@172.126.0.100:5060 SIP/2.0
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK566195022
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=60710202
To: <sip:200@172.126.0.100>;tag=as33b0fccc
Call-ID: 401485700@172.126.0.3
CSeq: 240 INFO
Contact: <sip:232@172.126.0.3:5060>
Max-Forwards: 30
User-Agent: dble
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=2
Duration=160
<------------->
--- (11 headers 2 lines) ---
Receiving INFO!
* DTMF-relay event received: 2

<--- Transmitting (NAT) to 172.126.0.3:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK566195022;received=172.126.0.3;rport=5060
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=60710202
To: <sip:200@172.126.0.100>;tag=as33b0fccc
Call-ID: 401485700@172.126.0.3
CSeq: 240 INFO
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[Mar 25 14:24:37] DTMF[29117]: channel.c:4066 __ast_read: DTMF end '2' received on SIP/232-000000a9, duration 160 ms
[Mar 25 14:24:37] DTMF[29117]: channel.c:4135 __ast_read: DTMF end passthrough '2' on SIP/232-000000a9

<--- SIP read from UDP:172.126.0.3:5060 --->
INFO sip:200@172.126.0.100:5060 SIP/2.0
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK1431069371
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=60710202
To: <sip:200@172.126.0.100>;tag=as33b0fccc
Call-ID: 401485700@172.126.0.3
CSeq: 241 INFO
Contact: <sip:232@172.126.0.3:5060>
Max-Forwards: 30
User-Agent: dble
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=3
Duration=160
<------------->
--- (11 headers 2 lines) ---
Receiving INFO!
* DTMF-relay event received: 3

<--- Transmitting (NAT) to 172.126.0.3:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK1431069371;received=172.126.0.3;rport=5060
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=60710202
To: <sip:200@172.126.0.100>;tag=as33b0fccc
Call-ID: 401485700@172.126.0.3
CSeq: 241 INFO
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[Mar 25 14:24:38] DTMF[29117]: channel.c:4066 __ast_read: DTMF end '3' received on SIP/232-000000a9, duration 160 ms
[Mar 25 14:24:38] DTMF[29117]: channel.c:4135 __ast_read: DTMF end passthrough '3' on SIP/232-000000a9

<--- SIP read from UDP:172.126.0.3:5060 --->
REGISTER sip:172.126.0.100 SIP/2.0
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK433445762
From: "GOIP" <sip:232@172.126.0.100>;tag=97001673
To: "GOIP" <sip:232@172.126.0.100>
Call-ID: 585551672@172.126.0.3
CSeq: 2496 REGISTER
Contact: <sip:232@172.126.0.3:5060>;expires=60
Authorization: Digest username="232", realm="asterisk", nonce="18cdd056", uri="sip:172.126.0.100", response="d7b40832a8f7b412989aabbcd1003bfd", algorithm=MD5
Max-Forwards: 30
User-Agent: dble
Expires: 60
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to 172.126.0.3:5060 (NAT)
[Mar 25 14:24:39] NOTICE[13340]: chan_sip.c:15060 check_auth: Correct auth, but based on stale nonce received from '"GOIP" <sip:232@172.126.0.100>;tag=97001673'

<--- Transmitting (NAT) to 172.126.0.3:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK433445762;received=172.126.0.3;rport=5060
From: "GOIP" <sip:232@172.126.0.100>;tag=97001673
To: "GOIP" <sip:232@172.126.0.100>;tag=as677354cb
Call-ID: 585551672@172.126.0.3
CSeq: 2496 REGISTER
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5fb7216a", stale=true
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '585551672@172.126.0.3' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:172.126.0.3:5060 --->
REGISTER sip:172.126.0.100 SIP/2.0
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK1604764149
From: "GOIP" <sip:232@172.126.0.100>;tag=97001673
To: "GOIP" <sip:232@172.126.0.100>
Call-ID: 585551672@172.126.0.3
CSeq: 2497 REGISTER
Contact: <sip:232@172.126.0.3:5060>;expires=60
Authorization: Digest username="232", realm="asterisk", nonce="5fb7216a", uri="sip:172.126.0.100", response="38d288219dbf9ea9c48878b6dae338ad", algorithm=MD5
Max-Forwards: 30
User-Agent: dble
Expires: 60
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to 172.126.0.3:5060 (NAT)

<--- Transmitting (NAT) to 172.126.0.3:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK1604764149;received=172.126.0.3;rport=5060
From: "GOIP" <sip:232@172.126.0.100>;tag=97001673
To: "GOIP" <sip:232@172.126.0.100>;tag=as677354cb
Call-ID: 585551672@172.126.0.3
CSeq: 2497 REGISTER
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 60
Contact: <sip:232@172.126.0.3:5060>;expires=60
Date: Wed, 25 Mar 2015 08:24:39 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '585551672@172.126.0.3' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:172.126.0.3:5060 --->
INFO sip:200@172.126.0.100:5060 SIP/2.0
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK1410706408
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=60710202
To: <sip:200@172.126.0.100>;tag=as33b0fccc
Call-ID: 401485700@172.126.0.3
CSeq: 242 INFO
Contact: <sip:232@172.126.0.3:5060>
Max-Forwards: 30
User-Agent: dble
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=6
Duration=160
<------------->
--- (11 headers 2 lines) ---
Receiving INFO!
* DTMF-relay event received: 6

<--- Transmitting (NAT) to 172.126.0.3:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK1410706408;received=172.126.0.3;rport=5060
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=60710202
To: <sip:200@172.126.0.100>;tag=as33b0fccc
Call-ID: 401485700@172.126.0.3
CSeq: 242 INFO
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[Mar 25 14:24:42] DTMF[29117]: channel.c:4066 __ast_read: DTMF end '6' received on SIP/232-000000a9, duration 160 ms
[Mar 25 14:24:42] DTMF[29117]: channel.c:4135 __ast_read: DTMF end passthrough '6' on SIP/232-000000a9

<--- SIP read from UDP:172.126.0.3:5060 --->
INFO sip:200@172.126.0.100:5060 SIP/2.0
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK688495857
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=60710202
To: <sip:200@172.126.0.100>;tag=as33b0fccc
Call-ID: 401485700@172.126.0.3
CSeq: 243 INFO
Contact: <sip:232@172.126.0.3:5060>
Max-Forwards: 30
User-Agent: dble
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=6
Duration=160
<------------->
--- (11 headers 2 lines) ---
Receiving INFO!
* DTMF-relay event received: 6

<--- Transmitting (NAT) to 172.126.0.3:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK688495857;received=172.126.0.3;rport=5060
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=60710202
To: <sip:200@172.126.0.100>;tag=as33b0fccc
Call-ID: 401485700@172.126.0.3
CSeq: 243 INFO
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[Mar 25 14:24:42] DTMF[29117]: channel.c:4066 __ast_read: DTMF end '6' received on SIP/232-000000a9, duration 160 ms
[Mar 25 14:24:42] DTMF[29117]: channel.c:4135 __ast_read: DTMF end passthrough '6' on SIP/232-000000a9

<--- SIP read from UDP:172.126.0.3:5060 --->
INFO sip:200@172.126.0.100:5060 SIP/2.0
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK1665474351
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=60710202
To: <sip:200@172.126.0.100>;tag=as33b0fccc
Call-ID: 401485700@172.126.0.3
CSeq: 244 INFO
Contact: <sip:232@172.126.0.3:5060>
Max-Forwards: 30
User-Agent: dble
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=7
Duration=160
<------------->
--- (11 headers 2 lines) ---
Receiving INFO!
* DTMF-relay event received: 7

<--- Transmitting (NAT) to 172.126.0.3:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK1665474351;received=172.126.0.3;rport=5060
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=60710202
To: <sip:200@172.126.0.100>;tag=as33b0fccc
Call-ID: 401485700@172.126.0.3
CSeq: 244 INFO
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[Mar 25 14:24:44] DTMF[29117]: channel.c:4066 __ast_read: DTMF end '7' received on SIP/232-000000a9, duration 160 ms
[Mar 25 14:24:44] DTMF[29117]: channel.c:4135 __ast_read: DTMF end passthrough '7' on SIP/232-000000a9

<--- SIP read from UDP:172.126.0.3:5060 --->
INFO sip:200@172.126.0.100:5060 SIP/2.0
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK1812192109
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=60710202
To: <sip:200@172.126.0.100>;tag=as33b0fccc
Call-ID: 401485700@172.126.0.3
CSeq: 245 INFO
Contact: <sip:232@172.126.0.3:5060>
Max-Forwards: 30
User-Agent: dble
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=8
Duration=160
<------------->
--- (11 headers 2 lines) ---
Receiving INFO!
* DTMF-relay event received: 8

<--- Transmitting (NAT) to 172.126.0.3:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK1812192109;received=172.126.0.3;rport=5060
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=60710202
To: <sip:200@172.126.0.100>;tag=as33b0fccc
Call-ID: 401485700@172.126.0.3
CSeq: 245 INFO
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[Mar 25 14:24:45] DTMF[29117]: channel.c:4066 __ast_read: DTMF end '8' received on SIP/232-000000a9, duration 160 ms
[Mar 25 14:24:45] DTMF[29117]: channel.c:4135 __ast_read: DTMF end passthrough '8' on SIP/232-000000a9

<--- SIP read from UDP:172.126.0.3:5060 --->
INFO sip:200@172.126.0.100:5060 SIP/2.0
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK1564661591
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=60710202
To: <sip:200@172.126.0.100>;tag=as33b0fccc
Call-ID: 401485700@172.126.0.3
CSeq: 246 INFO
Contact: <sip:232@172.126.0.3:5060>
Max-Forwards: 30
User-Agent: dble
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=9
Duration=160
<------------->
--- (11 headers 2 lines) ---
Receiving INFO!
* DTMF-relay event received: 9

<--- Transmitting (NAT) to 172.126.0.3:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK1564661591;received=172.126.0.3;rport=5060
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=60710202
To: <sip:200@172.126.0.100>;tag=as33b0fccc
Call-ID: 401485700@172.126.0.3
CSeq: 246 INFO
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[Mar 25 14:24:46] DTMF[29117]: channel.c:4066 __ast_read: DTMF end '9' received on SIP/232-000000a9, duration 160 ms
[Mar 25 14:24:46] DTMF[29117]: channel.c:4135 __ast_read: DTMF end passthrough '9' on SIP/232-000000a9
    -- User entered '12366789'
    -- Executing [s@dtmf-analys:6] SayDigits("SIP/232-000000a9", "12366789") in new stack
    -- <SIP/232-000000a9> Playing 'digits/1.alaw' (language 'en')
    -- <SIP/232-000000a9> Playing 'digits/2.alaw' (language 'en')
    -- <SIP/232-000000a9> Playing 'digits/3.alaw' (language 'en')
    -- <SIP/232-000000a9> Playing 'digits/6.alaw' (language 'en')
    -- <SIP/232-000000a9> Playing 'digits/6.alaw' (language 'en')
    -- <SIP/232-000000a9> Playing 'digits/7.alaw' (language 'en')
    -- <SIP/232-000000a9> Playing 'digits/8.alaw' (language 'en')
    -- <SIP/232-000000a9> Playing 'digits/9.alaw' (language 'en')
    -- Executing [s@dtmf-analys:7] Goto("SIP/232-000000a9", "collect") in new stack
    -- Goto (dtmf-analys,s,5)
    -- Executing [s@dtmf-analys:5] Read("SIP/232-000000a9", "digito,,9") in new stack
    -- Accepting a maximum of 9 digits.

<--- SIP read from UDP:172.126.0.3:5060 --->

<------------->
    -- User entered nothing.
Вложения
goip-4.JPG
Вернуться к началу
Roman_R
Сообщения: 18
Зарегистрирован: 18 мар 2015, 20:23

Re: Goip+Asterisk.Проблемы с распознаванием DTMF

Сообщение Roman_R »

Цифра 5 иногда определяется. Почему-то наблюдаюися сообщения о необходимости авторизации - хотя она выключена (видно в настройках). Цифра 4 не проходит вообще.
Вернуться к началу
Roman_R
Сообщения: 18
Зарегистрирован: 18 мар 2015, 20:23

Re: Goip+Asterisk.Проблемы с распознаванием DTMF

Сообщение Roman_R »

Вот тоже самое с другого телефона

Код: Выделить всё

DSS*CLI> sip set debug peer 232
SIP Debugging Enabled for IP: 172.126.0.3
Really destroying SIP dialog '1462287870@172.126.0.3' Method: BYE

<--- SIP read from UDP:172.126.0.3:5061 --->

<------------->

<--- SIP read from UDP:172.126.0.3:5061 --->
INVITE sip:200@172.126.0.100:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.126.0.3:5061;rport;branch=z9hG4bK994518083
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=711998119
To: <sip:200@172.126.0.100>
Call-ID: 1420088067@172.126.0.3
CSeq: 38 INVITE
Contact: <sip:232@172.126.0.3:5061>
Max-Forwards: 30
User-Agent: dble
Remote-Party-ID: "77779583773" <sip:77779583773@172.126.0.100>;party=calling;screen=no;privacy=off
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REFER, REGISTER, MESSAGE, INFO, SUBSCRIBE
Content-Type: application/sdp
Content-Length: 174

v=0
o=dble 1427272823 1427272823 IN IP4 172.126.0.3
s=dble
c=IN IP4 172.126.0.3
t=0 0
m=audio 10000 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
<------------->
--- (13 headers 9 lines) ---
Sending to 172.126.0.3:5061 (NAT)
Using INVITE request as basis request - 1420088067@172.126.0.3
Found peer '232' for '232' from 172.126.0.3:5061

<--- Reliably Transmitting (NAT) to 172.126.0.3:5061 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.126.0.3:5061;branch=z9hG4bK994518083;received=172.126.0.3;rport=5061
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=711998119
To: <sip:200@172.126.0.100>;tag=as2be8df4e
Call-ID: 1420088067@172.126.0.3
CSeq: 38 INVITE
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4e409bac"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '1420088067@172.126.0.3' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:172.126.0.3:5061 --->
ACK sip:200@172.126.0.100:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.126.0.3:5061;rport;branch=z9hG4bK994518083
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=711998119
To: <sip:200@172.126.0.100>;tag=as2be8df4e
Call-ID: 1420088067@172.126.0.3
CSeq: 38 ACK
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:172.126.0.3:5061 --->
INVITE sip:200@172.126.0.100:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.126.0.3:5061;rport;branch=z9hG4bK62356739
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=711998119
To: <sip:200@172.126.0.100>
Call-ID: 1420088067@172.126.0.3
CSeq: 39 INVITE
Contact: <sip:232@172.126.0.3:5061>
Authorization: Digest username="232", realm="asterisk", nonce="4e409bac", uri="sip:200@172.126.0.100", response="0da84ca1c34c1932179540498c2ed8f2", algorithm=MD5
Max-Forwards: 30
User-Agent: dble
Remote-Party-ID: "77779583773" <sip:77779583773@172.126.0.100>;party=calling;screen=no;privacy=off
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REFER, REGISTER, MESSAGE, INFO, SUBSCRIBE
Content-Type: application/sdp
Content-Length: 174

v=0
o=dble 1427272823 1427272823 IN IP4 172.126.0.3
s=dble
c=IN IP4 172.126.0.3
t=0 0
m=audio 10000 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
<------------->
--- (14 headers 9 lines) ---
Sending to 172.126.0.3:5061 (NAT)
Using INVITE request as basis request - 1420088067@172.126.0.3
Found peer '232' for '232' from 172.126.0.3:5061
  == Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 172.126.0.3:10000
Looking for 200 in default (domain 172.126.0.100)
list_route: hop: <sip:232@172.126.0.3:5061>

<--- Transmitting (NAT) to 172.126.0.3:5061 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.126.0.3:5061;branch=z9hG4bK62356739;received=172.126.0.3;rport=5061
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=711998119
To: <sip:200@172.126.0.100>
Call-ID: 1420088067@172.126.0.3
CSeq: 39 INVITE
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:200@172.126.0.100:5060>
Content-Length: 0


<------------>
    -- Executing [200@default:1] Goto("SIP/232-000000ad", "menu,s,1") in new stack
    -- Goto (menu,s,1)
    -- Executing [s@menu:1] Answer("SIP/232-000000ad", "") in new stack
Audio is at 10312
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP

<--- Reliably Transmitting (NAT) to 172.126.0.3:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.126.0.3:5061;branch=z9hG4bK62356739;received=172.126.0.3;rport=5061
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=711998119
To: <sip:200@172.126.0.100>;tag=as452493e2
Call-ID: 1420088067@172.126.0.3
CSeq: 39 INVITE
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:200@172.126.0.100:5060>
Content-Type: application/sdp
Content-Length: 232

v=0
o=root 147908857 147908857 IN IP4 172.126.0.100
s=Asterisk PBX 1.8.25.0
c=IN IP4 172.126.0.100
t=0 0
m=audio 10312 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.126.0.3:5061 --->
ACK sip:200@172.126.0.100:5060 SIP/2.0
Via: SIP/2.0/UDP 172.126.0.3:5061;rport;branch=z9hG4bK1389152635
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=711998119
To: <sip:200@172.126.0.100>;tag=as452493e2
Call-ID: 1420088067@172.126.0.3
CSeq: 39 ACK
Contact: <sip:232@172.126.0.3:5061>
Max-Forwards: 30
User-Agent: dble
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
    -- Executing [s@menu:2] BackGround("SIP/232-000000ad", "beep") in new stack
    -- <SIP/232-000000ad> Playing 'beep.alaw' (language 'en')
    -- Executing [s@menu:3] WaitExten("SIP/232-000000ad", "5") in new stack

<--- SIP read from UDP:172.126.0.3:5061 --->
INFO sip:200@172.126.0.100:5060 SIP/2.0
Via: SIP/2.0/UDP 172.126.0.3:5061;branch=z9hG4bK654202222
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=711998119
To: <sip:200@172.126.0.100>;tag=as452493e2
Call-ID: 1420088067@172.126.0.3
CSeq: 40 INFO
Contact: <sip:232@172.126.0.3:5061>
Max-Forwards: 30
User-Agent: dble
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=0
Duration=160
<------------->
--- (11 headers 2 lines) ---
Receiving INFO!
* DTMF-relay event received: 0

<--- Transmitting (NAT) to 172.126.0.3:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.126.0.3:5061;branch=z9hG4bK654202222;received=172.126.0.3;rport=5061
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=711998119
To: <sip:200@172.126.0.100>;tag=as452493e2
Call-ID: 1420088067@172.126.0.3
CSeq: 40 INFO
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[Mar 25 14:40:56] DTMF[29345]: channel.c:4066 __ast_read: DTMF end '0' received on SIP/232-000000ad, duration 160 ms
[Mar 25 14:40:56] DTMF[29345]: channel.c:4135 __ast_read: DTMF end passthrough '0' on SIP/232-000000ad
  == CDR updated on SIP/232-000000ad
    -- Executing [0@menu:1] Goto("SIP/232-000000ad", "dtmf-analys,s,1") in new stack
    -- Goto (dtmf-analys,s,1)
    -- Executing [s@dtmf-analys:1] SayDigits("SIP/232-000000ad", "0") in new stack
    -- <SIP/232-000000ad> Playing 'digits/0.alaw' (language 'en')
    -- Executing [s@dtmf-analys:2] Answer("SIP/232-000000ad", "") in new stack
    -- Executing [s@dtmf-analys:3] BackGround("SIP/232-000000ad", "beep") in new stack
    -- <SIP/232-000000ad> Playing 'beep.alaw' (language 'en')
    -- Executing [s@dtmf-analys:4] Wait("SIP/232-000000ad", "1") in new stack
    -- Executing [s@dtmf-analys:5] Read("SIP/232-000000ad", "digito,,9") in new stack
    -- Accepting a maximum of 9 digits.

<--- SIP read from UDP:172.126.0.3:5061 --->
INFO sip:200@172.126.0.100:5060 SIP/2.0
Via: SIP/2.0/UDP 172.126.0.3:5061;branch=z9hG4bK1262235633
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=711998119
To: <sip:200@172.126.0.100>;tag=as452493e2
Call-ID: 1420088067@172.126.0.3
CSeq: 41 INFO
Contact: <sip:232@172.126.0.3:5061>
Max-Forwards: 30
User-Agent: dble
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=1
Duration=160
<------------->
--- (11 headers 2 lines) ---
Receiving INFO!
* DTMF-relay event received: 1

<--- Transmitting (NAT) to 172.126.0.3:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.126.0.3:5061;branch=z9hG4bK1262235633;received=172.126.0.3;rport=5061
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=711998119
To: <sip:200@172.126.0.100>;tag=as452493e2
Call-ID: 1420088067@172.126.0.3
CSeq: 41 INFO
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[Mar 25 14:40:59] DTMF[29345]: channel.c:4066 __ast_read: DTMF end '1' received on SIP/232-000000ad, duration 160 ms
[Mar 25 14:40:59] DTMF[29345]: channel.c:4135 __ast_read: DTMF end passthrough '1' on SIP/232-000000ad

<--- SIP read from UDP:172.126.0.3:5061 --->
REGISTER sip:172.126.0.100 SIP/2.0
Via: SIP/2.0/UDP 172.126.0.3:5061;branch=z9hG4bK1314714824
From: "GOIP" <sip:232@172.126.0.100>;tag=459803596
To: "GOIP" <sip:232@172.126.0.100>
Call-ID: 1851179586@172.126.0.3
CSeq: 24 REGISTER
Contact: <sip:232@172.126.0.3:5061>;expires=60
Authorization: Digest username="232", realm="asterisk", nonce="25310b45", uri="sip:172.126.0.100", response="956b96083c5bf57cd004b206c0a91a82", algorithm=MD5
Max-Forwards: 30
User-Agent: dble
Expires: 60
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to 172.126.0.3:5061 (NAT)
[Mar 25 14:41:00] NOTICE[13340]: chan_sip.c:15060 check_auth: Correct auth, but based on stale nonce received from '"GOIP" <sip:232@172.126.0.100>;tag=459803596'

<--- Transmitting (NAT) to 172.126.0.3:5061 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.126.0.3:5061;branch=z9hG4bK1314714824;received=172.126.0.3;rport=5061
From: "GOIP" <sip:232@172.126.0.100>;tag=459803596
To: "GOIP" <sip:232@172.126.0.100>;tag=as4f47f8d6
Call-ID: 1851179586@172.126.0.3
CSeq: 24 REGISTER
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="26ec09c6", stale=true
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '1851179586@172.126.0.3' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:172.126.0.3:5061 --->
REGISTER sip:172.126.0.100 SIP/2.0
Via: SIP/2.0/UDP 172.126.0.3:5061;branch=z9hG4bK569562015
From: "GOIP" <sip:232@172.126.0.100>;tag=459803596
To: "GOIP" <sip:232@172.126.0.100>
Call-ID: 1851179586@172.126.0.3
CSeq: 25 REGISTER
Contact: <sip:232@172.126.0.3:5061>;expires=60
Authorization: Digest username="232", realm="asterisk", nonce="26ec09c6", uri="sip:172.126.0.100", response="5c112b388ff8efd0faffe36dc2a93abe", algorithm=MD5
Max-Forwards: 30
User-Agent: dble
Expires: 60
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to 172.126.0.3:5061 (NAT)

<--- Transmitting (NAT) to 172.126.0.3:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.126.0.3:5061;branch=z9hG4bK569562015;received=172.126.0.3;rport=5061
From: "GOIP" <sip:232@172.126.0.100>;tag=459803596
To: "GOIP" <sip:232@172.126.0.100>;tag=as4f47f8d6
Call-ID: 1851179586@172.126.0.3
CSeq: 25 REGISTER
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 60
Contact: <sip:232@172.126.0.3:5061>;expires=60
Date: Wed, 25 Mar 2015 08:41:00 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '1851179586@172.126.0.3' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:172.126.0.3:5061 --->
INFO sip:200@172.126.0.100:5060 SIP/2.0
Via: SIP/2.0/UDP 172.126.0.3:5061;branch=z9hG4bK1922091466
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=711998119
To: <sip:200@172.126.0.100>;tag=as452493e2
Call-ID: 1420088067@172.126.0.3
CSeq: 42 INFO
Contact: <sip:232@172.126.0.3:5061>
Max-Forwards: 30
User-Agent: dble
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=2
Duration=160
<------------->
--- (11 headers 2 lines) ---
Receiving INFO!
* DTMF-relay event received: 2

<--- Transmitting (NAT) to 172.126.0.3:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.126.0.3:5061;branch=z9hG4bK1922091466;received=172.126.0.3;rport=5061
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=711998119
To: <sip:200@172.126.0.100>;tag=as452493e2
Call-ID: 1420088067@172.126.0.3
CSeq: 42 INFO
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[Mar 25 14:41:01] DTMF[29345]: channel.c:4066 __ast_read: DTMF end '2' received on SIP/232-000000ad, duration 160 ms
[Mar 25 14:41:01] DTMF[29345]: channel.c:4135 __ast_read: DTMF end passthrough '2' on SIP/232-000000ad

<--- SIP read from UDP:172.126.0.3:5061 --->
INFO sip:200@172.126.0.100:5060 SIP/2.0
Via: SIP/2.0/UDP 172.126.0.3:5061;branch=z9hG4bK1976338592
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=711998119
To: <sip:200@172.126.0.100>;tag=as452493e2
Call-ID: 1420088067@172.126.0.3
CSeq: 43 INFO
Contact: <sip:232@172.126.0.3:5061>
Max-Forwards: 30
User-Agent: dble
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=3
Duration=160
<------------->
--- (11 headers 2 lines) ---
Receiving INFO!
* DTMF-relay event received: 3

<--- Transmitting (NAT) to 172.126.0.3:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.126.0.3:5061;branch=z9hG4bK1976338592;received=172.126.0.3;rport=5061
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=711998119
To: <sip:200@172.126.0.100>;tag=as452493e2
Call-ID: 1420088067@172.126.0.3
CSeq: 43 INFO
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[Mar 25 14:41:02] DTMF[29345]: channel.c:4066 __ast_read: DTMF end '3' received on SIP/232-000000ad, duration 160 ms
[Mar 25 14:41:02] DTMF[29345]: channel.c:4135 __ast_read: DTMF end passthrough '3' on SIP/232-000000ad

<--- SIP read from UDP:172.126.0.3:5061 --->
INFO sip:200@172.126.0.100:5060 SIP/2.0
Via: SIP/2.0/UDP 172.126.0.3:5061;branch=z9hG4bK1682247637
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=711998119
To: <sip:200@172.126.0.100>;tag=as452493e2
Call-ID: 1420088067@172.126.0.3
CSeq: 44 INFO
Contact: <sip:232@172.126.0.3:5061>
Max-Forwards: 30
User-Agent: dble
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=4
Duration=160
<------------->
--- (11 headers 2 lines) ---
Receiving INFO!
* DTMF-relay event received: 4

<--- Transmitting (NAT) to 172.126.0.3:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.126.0.3:5061;branch=z9hG4bK1682247637;received=172.126.0.3;rport=5061
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=711998119
To: <sip:200@172.126.0.100>;tag=as452493e2
Call-ID: 1420088067@172.126.0.3
CSeq: 44 INFO
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[Mar 25 14:41:03] DTMF[29345]: channel.c:4066 __ast_read: DTMF end '4' received on SIP/232-000000ad, duration 160 ms
[Mar 25 14:41:03] DTMF[29345]: channel.c:4135 __ast_read: DTMF end passthrough '4' on SIP/232-000000ad

<--- SIP read from UDP:172.126.0.3:5061 --->
INFO sip:200@172.126.0.100:5060 SIP/2.0
Via: SIP/2.0/UDP 172.126.0.3:5061;branch=z9hG4bK36914158
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=711998119
To: <sip:200@172.126.0.100>;tag=as452493e2
Call-ID: 1420088067@172.126.0.3
CSeq: 45 INFO
Contact: <sip:232@172.126.0.3:5061>
Max-Forwards: 30
User-Agent: dble
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=5
Duration=160
<------------->
--- (11 headers 2 lines) ---
Receiving INFO!
* DTMF-relay event received: 5

<--- Transmitting (NAT) to 172.126.0.3:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.126.0.3:5061;branch=z9hG4bK36914158;received=172.126.0.3;rport=5061
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=711998119
To: <sip:200@172.126.0.100>;tag=as452493e2
Call-ID: 1420088067@172.126.0.3
CSeq: 45 INFO
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[Mar 25 14:41:04] DTMF[29345]: channel.c:4066 __ast_read: DTMF end '5' received on SIP/232-000000ad, duration 160 ms
[Mar 25 14:41:04] DTMF[29345]: channel.c:4135 __ast_read: DTMF end passthrough '5' on SIP/232-000000ad

<--- SIP read from UDP:172.126.0.3:5061 --->
INFO sip:200@172.126.0.100:5060 SIP/2.0
Via: SIP/2.0/UDP 172.126.0.3:5061;branch=z9hG4bK195285816
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=711998119
To: <sip:200@172.126.0.100>;tag=as452493e2
Call-ID: 1420088067@172.126.0.3
CSeq: 46 INFO
Contact: <sip:232@172.126.0.3:5061>
Max-Forwards: 30
User-Agent: dble
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=6
Duration=160
<------------->
--- (11 headers 2 lines) ---
Receiving INFO!
* DTMF-relay event received: 6

<--- Transmitting (NAT) to 172.126.0.3:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.126.0.3:5061;branch=z9hG4bK195285816;received=172.126.0.3;rport=5061
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=711998119
To: <sip:200@172.126.0.100>;tag=as452493e2
Call-ID: 1420088067@172.126.0.3
CSeq: 46 INFO
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[Mar 25 14:41:05] DTMF[29345]: channel.c:4066 __ast_read: DTMF end '6' received on SIP/232-000000ad, duration 160 ms
[Mar 25 14:41:05] DTMF[29345]: channel.c:4135 __ast_read: DTMF end passthrough '6' on SIP/232-000000ad

<--- SIP read from UDP:172.126.0.3:5061 --->
INFO sip:200@172.126.0.100:5060 SIP/2.0
Via: SIP/2.0/UDP 172.126.0.3:5061;branch=z9hG4bK1732976612
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=711998119
To: <sip:200@172.126.0.100>;tag=as452493e2
Call-ID: 1420088067@172.126.0.3
CSeq: 47 INFO
Contact: <sip:232@172.126.0.3:5061>
Max-Forwards: 30
User-Agent: dble
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=7
Duration=160
<------------->
--- (11 headers 2 lines) ---
Receiving INFO!
* DTMF-relay event received: 7

<--- Transmitting (NAT) to 172.126.0.3:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.126.0.3:5061;branch=z9hG4bK1732976612;received=172.126.0.3;rport=5061
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=711998119
To: <sip:200@172.126.0.100>;tag=as452493e2
Call-ID: 1420088067@172.126.0.3
CSeq: 47 INFO
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[Mar 25 14:41:06] DTMF[29345]: channel.c:4066 __ast_read: DTMF end '7' received on SIP/232-000000ad, duration 160 ms
[Mar 25 14:41:06] DTMF[29345]: channel.c:4135 __ast_read: DTMF end passthrough '7' on SIP/232-000000ad

<--- SIP read from UDP:172.126.0.3:5061 --->
INFO sip:200@172.126.0.100:5060 SIP/2.0
Via: SIP/2.0/UDP 172.126.0.3:5061;branch=z9hG4bK760671503
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=711998119
To: <sip:200@172.126.0.100>;tag=as452493e2
Call-ID: 1420088067@172.126.0.3
CSeq: 48 INFO
Contact: <sip:232@172.126.0.3:5061>
Max-Forwards: 30
User-Agent: dble
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=8
Duration=160
<------------->
--- (11 headers 2 lines) ---
Receiving INFO!
* DTMF-relay event received: 8

<--- Transmitting (NAT) to 172.126.0.3:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.126.0.3:5061;branch=z9hG4bK760671503;received=172.126.0.3;rport=5061
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=711998119
To: <sip:200@172.126.0.100>;tag=as452493e2
Call-ID: 1420088067@172.126.0.3
CSeq: 48 INFO
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[Mar 25 14:41:08] DTMF[29345]: channel.c:4066 __ast_read: DTMF end '8' received on SIP/232-000000ad, duration 160 ms
[Mar 25 14:41:08] DTMF[29345]: channel.c:4135 __ast_read: DTMF end passthrough '8' on SIP/232-000000ad

<--- SIP read from UDP:172.126.0.3:5061 --->
INFO sip:200@172.126.0.100:5060 SIP/2.0
Via: SIP/2.0/UDP 172.126.0.3:5061;branch=z9hG4bK1046215628
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=711998119
To: <sip:200@172.126.0.100>;tag=as452493e2
Call-ID: 1420088067@172.126.0.3
CSeq: 49 INFO
Contact: <sip:232@172.126.0.3:5061>
Max-Forwards: 30
User-Agent: dble
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=9
Duration=160
<------------->
--- (11 headers 2 lines) ---
Receiving INFO!
* DTMF-relay event received: 9

<--- Transmitting (NAT) to 172.126.0.3:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.126.0.3:5061;branch=z9hG4bK1046215628;received=172.126.0.3;rport=5061
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=711998119
To: <sip:200@172.126.0.100>;tag=as452493e2
Call-ID: 1420088067@172.126.0.3
CSeq: 49 INFO
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[Mar 25 14:41:09] DTMF[29345]: channel.c:4066 __ast_read: DTMF end '9' received on SIP/232-000000ad, duration 160 ms
[Mar 25 14:41:09] DTMF[29345]: channel.c:4135 __ast_read: DTMF end passthrough '9' on SIP/232-000000ad
    -- User entered '123456789'
    -- Executing [s@dtmf-analys:6] SayDigits("SIP/232-000000ad", "123456789") in new stack
    -- <SIP/232-000000ad> Playing 'digits/1.alaw' (language 'en')
    -- <SIP/232-000000ad> Playing 'digits/2.alaw' (language 'en')
    -- <SIP/232-000000ad> Playing 'digits/3.alaw' (language 'en')
    -- <SIP/232-000000ad> Playing 'digits/4.alaw' (language 'en')
    -- <SIP/232-000000ad> Playing 'digits/5.alaw' (language 'en')
    -- <SIP/232-000000ad> Playing 'digits/6.alaw' (language 'en')
    -- <SIP/232-000000ad> Playing 'digits/7.alaw' (language 'en')
    -- <SIP/232-000000ad> Playing 'digits/8.alaw' (language 'en')
    -- <SIP/232-000000ad> Playing 'digits/9.alaw' (language 'en')
    -- Executing [s@dtmf-analys:7] Goto("SIP/232-000000ad", "collect") in new stack
    -- Goto (dtmf-analys,s,5)
    -- Executing [s@dtmf-analys:5] Read("SIP/232-000000ad", "digito,,9") in new stack
    -- Accepting a maximum of 9 digits.

<--- SIP read from UDP:172.126.0.3:5061 --->
BYE sip:200@172.126.0.100:5060 SIP/2.0
Via: SIP/2.0/UDP 172.126.0.3:5061;branch=z9hG4bK553872004
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=711998119
To: <sip:200@172.126.0.100>;tag=as452493e2
Call-ID: 1420088067@172.126.0.3
CSeq: 50 BYE
Contact: <sip:232@172.126.0.3:5061>
Authorization: Digest username="232", realm="asterisk", nonce="4e409bac", uri="sip:200@172.126.0.100", response="0da84ca1c34c1932179540498c2ed8f2", algorithm=MD5
Max-Forwards: 30
User-Agent: dble
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 172.126.0.3:5061 (NAT)
    -- User disconnected
Scheduling destruction of SIP dialog '1420088067@172.126.0.3' in 32000 ms (Method: BYE)

<--- Transmitting (NAT) to 172.126.0.3:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.126.0.3:5061;branch=z9hG4bK553872004;received=172.126.0.3;rport=5061
From: "GOIP" <sip:232@172.126.0.100:5060;user=phone>;tag=711998119
To: <sip:200@172.126.0.100>;tag=as452493e2
Call-ID: 1420088067@172.126.0.3
CSeq: 50 BYE
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Really destroying SIP dialog '1420088067@172.126.0.3' Method: BYE
DSS*CLI> sip set debug off
SIP Debugging Disabled
DSS*CLI>
Без разницы SIP INFO или RFC2833 :!:
Вернуться к началу
ded
Сообщения: 15618
Зарегистрирован: 26 авг 2010, 19:00

Re: Goip+Asterisk.Проблемы с распознаванием DTMF

Сообщение ded »

Почему то при этом ваш GoIP авторизуется, не знаю, где Вы там отключили авторизацию в настройках
<--- SIP read from UDP:172.126.0.3:5060 --->
REGISTER sip:172.126.0.100 SIP/2.0
Via: SIP/2.0/UDP 172.126.0.3:5060;branch=z9hG4bK1604764149
From: "GOIP" <sip:232@172.126.0.100>;tag=97001673
To: "GOIP" <sip:232@172.126.0.100>
Call-ID: 585551672@172.126.0.3
CSeq: 2497 REGISTER
Contact: <sip:232@172.126.0.3:5060>;expires=60
Authorization: Digest username="232", realm="asterisk", nonce="5fb7216a", uri="sip:172.126.0.100", response="38d288219dbf9ea9c48878b6dae338ad", algorithm=MD5
Max-Forwards: 30
User-Agent: dble
Expires: 60
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