IPLDK + мультифон

defiso · Сообщение **defiso** » 25 авг 2011, 16:40

Здравствуйте.

Проблема при звонке с аналоговой АТС LG, подключенной к астериску через плату IPLDK по SIP, на внешний номер через Мегафоновский мультифон (тоже по SIP).

Гудки не идут, вылезает варинг:

Код: Выделить всё

-- Executing [89222221243@office:1] Dial("SIP/ipldk-000000db", "SIP/multifon/89222221243,80,tT") in new stack
== Using SIP RTP CoS mark 5
    -- Called multifon/89222221243
[Aug 25 18:21:56] WARNING[5985]: chan_sip.c:18480 handle_response_invite: Re-invite to non-existing call leg on other UA. SIP dialog '4a53b5386b0560a7035d4770402b0eeb@multifon.ru'. Giving up.

Причем с обычного SIP-телефона, подключенного к астериску - звонки через мультифон идут. И через SIP-аккаунт IPLDK звонки во вне, например через h323-транк тоже идут. Проблема возникает при их связке.

В sip.conf прописаны:

Аккаунт для Мегафон-мультифон:

Код: Выделить всё

[multifon]
type=peer
host=sbc.megafon.ru
username=<name>
secret=<pass>
insecure=invite,port
fromdomain=multifon.ru
fromuser=<name>
disallow=all
allow=ulaw
qualify=yes
nat=no
dtmfmode=inband
context=incom
canreinvite=no

Пользователь для LG IPLDK:

Код: Выделить всё

[ipldk]
type=peer
host=dynamic
username=<name>
secret=<pass>
context=office
canreinvite=no
nat=no
allow=ulaw
allow=alaw
qualify=yes

ded · Сообщение **ded** » 25 авг 2011, 17:03

sip set debug peer multifon
поможет увидеть причину.

defiso · Сообщение **defiso** » 26 авг 2011, 12:52

В спойлере простыня с дебагом.

В ней: melk2 - SIP аккаунт IPLDK
m79221533510 - SIP аккаунт мультифона
89222221243 - набираемый номер с IPLDK

Мне не нравится строчка:

Код: Выделить всё

<--- SIP read from UDP:193.201.229.35:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 10.0.0.45:5060;received=195.58.7.132;branch=z9hG4bK0681b28c;rport=5060
From: "asterisk" <sip:asterisk@10.0.0.45:5060>;tag=as605f982b
To: <sip:sbc.megafon.ru>;tag=aprqngfrt-2tb6h500000c6
Call-ID: 1ef3542d2e03ace548edbd9877162ac8@10.0.0.45
CSeq: 102 OPTIONS
Reason: Q.850;cause=55;text="Call Terminated"

Не проходит авторизация на мультифоне? Почему?

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

-- Executing [89222221243@office:1] Dial("SIP/melk2-00000080", "SIP/m79221533510/89222221243,80,tT") in new stack
== Using SIP RTP CoS mark 5
Audio is at 10.0.0.45 port 13332
Adding codec 0x4 (ulaw) to SDP
Reliably Transmitting (no NAT) to 193.201.229.35:5060:
INVITE sip:89222221243@sbc.megafon.ru SIP/2.0
Via: SIP/2.0/UDP 10.0.0.45:5060;branch=z9hG4bK69303f55;rport
Max-Forwards: 70
From: "�� " <sip:79221533510@multifon.ru>;tag=as7bb67aea
To: <sip:89222221243@sbc.megafon.ru>
Contact: <sip:79221533510@10.0.0.45>
Call-ID: 73aaf0dd4f3c20fe75c564f654fb53b7@multifon.ru
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.20
Date: Fri, 26 Aug 2011 08:37:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 200

v=0
o=root 577024726 577024726 IN IP4 10.0.0.45
s=Asterisk PBX 1.6.2.20
c=IN IP4 10.0.0.45
t=0 0
m=audio 13332 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
-- Called m79221533510/89222221243

<--- SIP read from UDP:193.201.229.35:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.0.45:5060;received=195.58.7.132;branch=z9hG4bK69303f55;rport=5060
From: "�� " <sip:79221533510@multifon.ru>;tag=as7bb67aea
To: <sip:89222221243@sbc.megafon.ru>
Call-ID: 73aaf0dd4f3c20fe75c564f654fb53b7@multifon.ru
CSeq: 102 INVITE

<------------->
--- (6 headers 0 lines) ---
Reliably Transmitting (NAT) to 193.201.229.35:5060:
OPTIONS sip:sbc.megafon.ru SIP/2.0
Via: SIP/2.0/UDP 10.0.0.45:5060;branch=z9hG4bK0681b28c;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.0.0.45>;tag=as605f982b
To: <sip:sbc.megafon.ru>
Contact: <sip:asterisk@10.0.0.45>
Call-ID: 1ef3542d2e03ace548edbd9877162ac8@10.0.0.45
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.20
Date: Fri, 26 Aug 2011 08:37:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

---

<--- SIP read from UDP:193.201.229.35:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 10.0.0.45:5060;received=195.58.7.132;branch=z9hG4bK0681b28c;rport=5060
From: "asterisk" <sip:asterisk@10.0.0.45:5060>;tag=as605f982b
To: <sip:sbc.megafon.ru>;tag=aprqngfrt-2tb6h500000c6
Call-ID: 1ef3542d2e03ace548edbd9877162ac8@10.0.0.45
CSeq: 102 OPTIONS
Reason: Q.850;cause=55;text="Call Terminated"

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '1ef3542d2e03ace548edbd9877162ac8@10.0.0.45' Method: OPTIONS
Reliably Transmitting (NAT) to 193.201.229.35:5060:
OPTIONS sip:sbc.megafon.ru SIP/2.0
Via: SIP/2.0/UDP 10.0.0.45:5060;branch=z9hG4bK68415c65;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.0.0.45>;tag=as63172b11
To: <sip:sbc.megafon.ru>
Contact: <sip:asterisk@10.0.0.45>
Call-ID: 23eddab4311342f01493bc466d7f96d1@10.0.0.45
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.20
Date: Fri, 26 Aug 2011 08:37:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

---
Reliably Transmitting (no NAT) to 193.201.229.35:5060:
OPTIONS sip:sbc.megafon.ru SIP/2.0
Via: SIP/2.0/UDP 10.0.0.45:5060;branch=z9hG4bK45f39752;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.0.0.45>;tag=as2d9bbce8
To: <sip:sbc.megafon.ru>
Contact: <sip:asterisk@10.0.0.45>
Call-ID: 5b4994192c1131bb45effc2f5802125b@10.0.0.45
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.20
Date: Fri, 26 Aug 2011 08:37:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

---

<--- SIP read from UDP:193.201.229.35:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 10.0.0.45:5060;received=195.58.7.132;branch=z9hG4bK68415c65;rport=5060
From: "asterisk" <sip:asterisk@10.0.0.45:5060>;tag=as63172b11
To: <sip:sbc.megafon.ru>;tag=aprqngfrt-8qed4s30000c6
Call-ID: 23eddab4311342f01493bc466d7f96d1@10.0.0.45
CSeq: 102 OPTIONS
Reason: Q.850;cause=55;text="Call Terminated"

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '23eddab4311342f01493bc466d7f96d1@10.0.0.45' Method: OPTIONS

<--- SIP read from UDP:193.201.229.35:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 10.0.0.45:5060;received=195.58.7.132;branch=z9hG4bK45f39752;rport=5060
From: "asterisk" <sip:asterisk@10.0.0.45:5060>;tag=as2d9bbce8
To: <sip:sbc.megafon.ru>;tag=aprqngfrt-eqqf0f20000c6
Call-ID: 5b4994192c1131bb45effc2f5802125b@10.0.0.45
CSeq: 102 OPTIONS
Reason: Q.850;cause=55;text="Call Terminated"

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '5b4994192c1131bb45effc2f5802125b@10.0.0.45' Method: OPTIONS
Reliably Transmitting (NAT) to 193.201.229.35:5060:
OPTIONS sip:sbc.megafon.ru SIP/2.0
Via: SIP/2.0/UDP 10.0.0.45:5060;branch=z9hG4bK03abf6fe;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.0.0.45>;tag=as2de46646
To: <sip:sbc.megafon.ru>
Contact: <sip:asterisk@10.0.0.45>
Call-ID: 10f0e03301ec1918365a681033dced3f@10.0.0.45
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.20
Date: Fri, 26 Aug 2011 08:37:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

---

<--- SIP read from UDP:193.201.229.35:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 10.0.0.45:5060;received=195.58.7.132;branch=z9hG4bK03abf6fe;rport=5060
From: "asterisk" <sip:asterisk@10.0.0.45:5060>;tag=as2de46646
To: <sip:sbc.megafon.ru>;tag=aprqngfrt-3io41620000c6
Call-ID: 10f0e03301ec1918365a681033dced3f@10.0.0.45
CSeq: 102 OPTIONS
Reason: Q.850;cause=55;text="Call Terminated"

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '10f0e03301ec1918365a681033dced3f@10.0.0.45' Method: OPTIONS
[2011-08-26 14:37:51] NOTICE[1572]: chan_sip.c:12063 sip_reregister: -- Re-registration for 79221533510@multifon.ru@sbc.megafon.ru
> doing dnsmgr_lookup for 'sbc.megafon.ru'
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 193.201.229.35:5060:
REGISTER sip:multifon.ru SIP/2.0
Via: SIP/2.0/UDP 10.0.0.45:5060;branch=z9hG4bK61214afa;rport
Max-Forwards: 70
From: <sip:79221533510@multifon.ru>;tag=as58b68d46
To: <sip:79221533510@multifon.ru>
Call-ID: 4f8d814668cee71b2686de294cf867bd@10.0.0.45
CSeq: 216 REGISTER
User-Agent: Asterisk PBX 1.6.2.20
Authorization: Digest username="79221533510", realm="BREDBAND", algorithm=MD5, uri="sip:multifon.ru", nonce="MTMxNDMzNjE4OTo+n64FgAy9zaTA38qgdJue", response="eb973b6e08b14e211c4c8d3490825ea7", opaque="MTMxNDMzNjE4OTo+n64FgAy9zaTA38qgdJue", qop=auth, cnonce="377075bb", nc=00000071
Expires: 120
Contact: <sip:79221533510@10.0.0.45>
Content-Length: 0

---

<--- SIP read from UDP:193.201.229.35:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.45:5060;received=195.58.7.132;branch=z9hG4bK61214afa;rport=5060
From: <sip:79221533510@multifon.ru>;tag=as58b68d46
To: <sip:79221533510@multifon.ru>;tag=41E9324631353641B8C0D700
Call-ID: 4f8d814668cee71b2686de294cf867bd@10.0.0.45
CSeq: 216 REGISTER
Contact: <sip:79221533510@10.0.0.45:5060>;expires=180
Content-Length: 0
Service-Route: <sip:79221533510@193.201.229.35:5060;transport=udp;lr>

<------------->
--- (9 headers 0 lines) ---
Scheduling destruction of SIP dialog '4f8d814668cee71b2686de294cf867bd@10.0.0.45' in 32000 ms (Method: REGISTER)
[2011-08-26 14:37:51] NOTICE[1572]: chan_sip.c:18868 handle_response_register: Outbound Registration: Expiry for sbc.megafon.ru is 120 sec (Scheduling reregistration in 105 s)
[2011-08-26 14:37:51] NOTICE[1572]: chan_sip.c:12063 sip_reregister: -- Re-registration for 79221533511@multifon.ru@sbc.megafon.ru
> doing dnsmgr_lookup for 'sbc.megafon.ru'
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 193.201.229.35:5060:
REGISTER sip:multifon.ru SIP/2.0
Via: SIP/2.0/UDP 10.0.0.45:5060;branch=z9hG4bK70adea37;rport
Max-Forwards: 70
From: <sip:79221533511@multifon.ru>;tag=as486dc994
To: <sip:79221533511@multifon.ru>
Call-ID: 71ca018175d84e415c6a6e6a69db1696@10.0.0.45
CSeq: 216 REGISTER
User-Agent: Asterisk PBX 1.6.2.20
Authorization: Digest username="79221533511", realm="BREDBAND", algorithm=MD5, uri="sip:multifon.ru", nonce="MTMxNDMzNjE4OTo+n64FgAy9zaTA38qgdJue", response="03c20232037f8549ec5e5487f37a4b4b", opaque="MTMxNDMzNjE4OTo+n64FgAy9zaTA38qgdJue", qop=auth, cnonce="5f455dfc", nc=00000071
Expires: 120
Contact: <sip:79221533511@10.0.0.45>
Content-Length: 0

---

<--- SIP read from UDP:193.201.229.35:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.45:5060;received=195.58.7.132;branch=z9hG4bK70adea37;rport=5060
From: <sip:79221533511@multifon.ru>;tag=as486dc994
To: <sip:79221533511@multifon.ru>;tag=453D324631353641C4C0D700
Call-ID: 71ca018175d84e415c6a6e6a69db1696@10.0.0.45
CSeq: 216 REGISTER
Contact: <sip:79221533511@10.0.0.45:5060>;expires=180
Content-Length: 0
Service-Route: <sip:79221533511@193.201.229.35:5060;transport=udp;lr>

<------------->
--- (9 headers 0 lines) ---
Scheduling destruction of SIP dialog '71ca018175d84e415c6a6e6a69db1696@10.0.0.45' in 32000 ms (Method: REGISTER)
[2011-08-26 14:37:51] NOTICE[1572]: chan_sip.c:18868 handle_response_register: Outbound Registration: Expiry for sbc.megafon.ru is 120 sec (Scheduling reregistration in 105 s)
[2011-08-26 14:37:52] NOTICE[1572]: chan_sip.c:12063 sip_reregister: -- Re-registration for 79221533512@multifon.ru@sbc.megafon.ru
> doing dnsmgr_lookup for 'sbc.megafon.ru'
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 193.201.229.35:5060:
REGISTER sip:multifon.ru SIP/2.0
Via: SIP/2.0/UDP 10.0.0.45:5060;branch=z9hG4bK704aa659;rport
Max-Forwards: 70
From: <sip:79221533512@multifon.ru>;tag=as38bcb5ef
To: <sip:79221533512@multifon.ru>
Call-ID: 063554943ecc102835d8de745b4fbd83@10.0.0.45
CSeq: 216 REGISTER
User-Agent: Asterisk PBX 1.6.2.20
Authorization: Digest username="79221533512", realm="BREDBAND", algorithm=MD5, uri="sip:multifon.ru", nonce="MTMxNDMzNjE4OTo+n64FgAy9zaTA38qgdJue", response="bda18fd85c70414239fec2d8e04889de", opaque="MTMxNDMzNjE4OTo+n64FgAy9zaTA38qgdJue", qop=auth, cnonce="3966ad3a", nc=00000071
Expires: 120
Contact: <sip:79221533512@10.0.0.45>
Content-Length: 0

---

<--- SIP read from UDP:193.201.229.35:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.45:5060;received=195.58.7.132;branch=z9hG4bK704aa659;rport=5060
From: <sip:79221533512@multifon.ru>;tag=as38bcb5ef
To: <sip:79221533512@multifon.ru>;tag=A97B32463135364135C1D700
Call-ID: 063554943ecc102835d8de745b4fbd83@10.0.0.45
CSeq: 216 REGISTER
Contact: <sip:79221533512@10.0.0.45:5060>;expires=176
Content-Length: 0
Service-Route: <sip:79221533512@193.201.229.35:5060;transport=udp;lr>

<------------->
--- (9 headers 0 lines) ---
Scheduling destruction of SIP dialog '063554943ecc102835d8de745b4fbd83@10.0.0.45' in 32000 ms (Method: REGISTER)
[2011-08-26 14:37:53] NOTICE[1572]: chan_sip.c:18868 handle_response_register: Outbound Registration: Expiry for sbc.megafon.ru is 120 sec (Scheduling reregistration in 105 s)
[2011-08-26 14:37:54] NOTICE[1572]: chan_sip.c:12063 sip_reregister: -- Re-registration for 79221533514@multifon.ru@sbc.megafon.ru
> doing dnsmgr_lookup for 'sbc.megafon.ru'
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 193.201.229.35:5060:
REGISTER sip:multifon.ru SIP/2.0
Via: SIP/2.0/UDP 10.0.0.45:5060;branch=z9hG4bK4926f4c7;rport
Max-Forwards: 70
From: <sip:79221533514@multifon.ru>;tag=as1addf625
To: <sip:79221533514@multifon.ru>
Call-ID: 16139d1578737dad45bd9ac67412c71e@10.0.0.45
CSeq: 216 REGISTER
User-Agent: Asterisk PBX 1.6.2.20
Authorization: Digest username="79221533514", realm="BREDBAND", algorithm=MD5, uri="sip:multifon.ru", nonce="MTMxNDMzNjE4OTo+n64FgAy9zaTA38qgdJue", response="7af43a4db0713bbaa88a689e380e63e1", opaque="MTMxNDMzNjE4OTo+n64FgAy9zaTA38qgdJue", qop=auth, cnonce="18474499", nc=00000071
Expires: 120
Contact: <sip:79221533514@10.0.0.45>
Content-Length: 0

---

<--- SIP read from UDP:193.201.229.35:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.45:5060;received=195.58.7.132;branch=z9hG4bK4926f4c7;rport=5060
From: <sip:79221533514@multifon.ru>;tag=as1addf625
To: <sip:79221533514@multifon.ru>;tag=AF48324631353641C2C1D700
Call-ID: 16139d1578737dad45bd9ac67412c71e@10.0.0.45
CSeq: 216 REGISTER
Contact: <sip:79221533514@10.0.0.45:5060>;expires=180
Content-Length: 0
Service-Route: <sip:79221533514@193.201.229.35:5060;transport=udp;lr>

<------------->
--- (9 headers 0 lines) ---
Scheduling destruction of SIP dialog '16139d1578737dad45bd9ac67412c71e@10.0.0.45' in 32000 ms (Method: REGISTER)
[2011-08-26 14:37:54] NOTICE[1572]: chan_sip.c:18868 handle_response_register: Outbound Registration: Expiry for sbc.megafon.ru is 120 sec (Scheduling reregistration in 105 s)

<--- SIP read from UDP:193.201.229.35:5060 --->
SIP/2.0 408 Request Timeout
Via: SIP/2.0/UDP 10.0.0.45:5060;received=195.58.7.132;branch=z9hG4bK69303f55;rport=5060
From: "�� " <sip:79221533510@multifon.ru>;tag=as7bb67aea
To: <sip:89222221243@sbc.megafon.ru>;tag=aprqngfrt-acb1eh30000c6
Call-ID: 73aaf0dd4f3c20fe75c564f654fb53b7@multifon.ru
CSeq: 102 INVITE
Reason: Q.850;cause=18;text="Call Terminated"

<------------->
--- (7 headers 0 lines) ---
[2011-08-26 14:37:59] WARNING[1572]: chan_sip.c:18480 handle_response_invite: Re-invite to non-existing call leg on other UA. SIP dialog '73aaf0dd4f3c20fe75c564f654fb53b7@multifon.ru'. Giving up.
Transmitting (no NAT) to 193.201.229.35:5060:
ACK sip:89222221243@sbc.megafon.ru SIP/2.0
Via: SIP/2.0/UDP 10.0.0.45:5060;branch=z9hG4bK69303f55;rport
Max-Forwards: 70
From: "�� " <sip:79221533510@multifon.ru>;tag=as7bb67aea
To: <sip:89222221243@sbc.megafon.ru>;tag=aprqngfrt-acb1eh30000c6
Contact: <sip:79221533510@10.0.0.45>
Call-ID: 73aaf0dd4f3c20fe75c564f654fb53b7@multifon.ru
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.20
Content-Length: 0

Vlad1983 · Сообщение **Vlad1983** » 26 авг 2011, 13:56

SIP/2.0 403 Forbidden

это ответ на OPTIONS
долго рассказывать, но это нормально

приведи дебаг нормального звонка

defiso · Сообщение **defiso** » 26 авг 2011, 14:04

Нормальный звонок с софтового клиента:

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

-- Executing [89024457140@office:1] Dial("SIP/9998-00000091", "SIP/m79221533510/89024457140,80,tT") in new stack
== Using SIP RTP CoS mark 5
Audio is at 10.0.0.45 port 19708
Adding codec 0x4 (ulaw) to SDP
Reliably Transmitting (no NAT) to 193.201.229.35:5060:
INVITE sip:89024457140@sbc.megafon.ru SIP/2.0
Via: SIP/2.0/UDP 10.0.0.45:5060;branch=z9hG4bK334e4d42;rport
Max-Forwards: 70
From: "9998" <sip:79221533510@multifon.ru>;tag=as3055c390
To: <sip:89024457140@sbc.megafon.ru>
Contact: <sip:79221533510@10.0.0.45>
Call-ID: 67f0d88941779746256990c822130bb4@multifon.ru
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.20
Date: Fri, 26 Aug 2011 08:54:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 202

v=0
o=root 1034240275 1034240275 IN IP4 10.0.0.45
s=Asterisk PBX 1.6.2.20
c=IN IP4 10.0.0.45
t=0 0
m=audio 19708 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
-- Called m79221533510/89024457140

<--- SIP read from UDP:193.201.229.35:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.0.45:5060;received=195.58.7.132;branch=z9hG4bK334e4d42;rport=5060
From: "9998" <sip:79221533510@multifon.ru>;tag=as3055c390
To: <sip:89024457140@sbc.megafon.ru>
Call-ID: 67f0d88941779746256990c822130bb4@multifon.ru
CSeq: 102 INVITE

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

<--- SIP read from UDP:193.201.229.35:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.0.0.45:5060;received=195.58.7.132;branch=z9hG4bK334e4d42;rport=5060
From: "9998" <sip:79221533510@multifon.ru>;tag=as3055c390
To: <sip:89024457140@sbc.megafon.ru>;tag=E6C6324631353641343DD900
Call-ID: 67f0d88941779746256990c822130bb4@multifon.ru
CSeq: 102 INVITE
Proxy-Authenticate: Digest nonce="MTMxNDM0ODg0NToBlvvvvarZEGAlZCWZfRYX",opaque="MTMxNDM0ODg0NToBlvvvvarZEGAlZCWZfRYX",algorithm=md5,realm="BREDBAND",qop="auth"
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Transmitting (no NAT) to 193.201.229.35:5060:
ACK sip:89024457140@sbc.megafon.ru SIP/2.0
Via: SIP/2.0/UDP 10.0.0.45:5060;branch=z9hG4bK334e4d42;rport
Max-Forwards: 70
From: "9998" <sip:79221533510@multifon.ru>;tag=as3055c390
To: <sip:89024457140@sbc.megafon.ru>;tag=E6C6324631353641343DD900
Contact: <sip:79221533510@10.0.0.45>
Call-ID: 67f0d88941779746256990c822130bb4@multifon.ru
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.20
Content-Length: 0

---
Audio is at 10.0.0.45 port 19708
Adding codec 0x4 (ulaw) to SDP
Reliably Transmitting (no NAT) to 193.201.229.35:5060:
INVITE sip:89024457140@sbc.megafon.ru SIP/2.0
Via: SIP/2.0/UDP 10.0.0.45:5060;branch=z9hG4bK40b2ef88;rport
Max-Forwards: 70
From: "9998" <sip:79221533510@multifon.ru>;tag=as3055c390
To: <sip:89024457140@sbc.megafon.ru>
Contact: <sip:79221533510@10.0.0.45>
Call-ID: 67f0d88941779746256990c822130bb4@multifon.ru
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.2.20
Proxy-Authorization: Digest username="79221533510", realm="BREDBAND", algorithm=MD5, uri="sip:89024457140@sbc.megafon.ru", nonce="MTMxNDM0ODg0NToBlvvvvarZEGAlZCWZfRYX", response="4cdf7c1033baa4c902d90982803f3394", opaque="MTMxNDM0ODg0NToBlvvvvarZEGAlZCWZfRYX", qop=auth, cnonce="67cd38d9", nc=00000001
Date: Fri, 26 Aug 2011 08:54:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 202

v=0
o=root 1034240275 1034240276 IN IP4 10.0.0.45
s=Asterisk PBX 1.6.2.20
c=IN IP4 10.0.0.45
t=0 0
m=audio 19708 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:193.201.229.35:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.0.45:5060;received=195.58.7.132;branch=z9hG4bK40b2ef88;rport=5060
From: "9998" <sip:79221533510@multifon.ru>;tag=as3055c390
To: <sip:89024457140@sbc.megafon.ru>
Call-ID: 67f0d88941779746256990c822130bb4@multifon.ru
CSeq: 103 INVITE

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

<--- SIP read from UDP:193.201.229.35:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.0.0.45:5060;received=195.58.7.132;branch=z9hG4bK40b2ef88;rport=5060
From: "9998" <sip:79221533510@multifon.ru>;tag=as3055c390
To: <sip:89024457140@sbc.megafon.ru>;tag=7AFF324631353641393DD900
Call-ID: 67f0d88941779746256990c822130bb4@multifon.ru
CSeq: 103 INVITE
Allow: OPTIONS,CANCEL,UPDATE
Server: Dialogic-SIP/10.5.3.203 IMG1 0
Content-Length: 188
Content-Type: application/sdp
Contact: <sip:89024457140@193.201.229.35:5060;transport=udp>

v=0
o=Dialogic_SDP 13957027 0 IN IP4 193.201.229.35
s=Dialogic-SIP
c=IN IP4 193.201.229.35
t=0 0
m=audio 56522 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
a=ptime:20

<------------->
--- (11 headers 9 lines) ---
Found RTP audio format 0
Found audio description format PCMU for ID 0
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 193.201.229.35:56522
-- SIP/m79221533510-00000092 is making progress passing it to SIP/9998-00000091
[2011-08-26 14:54:09] NOTICE[2382]: rtp.c:1809 ast_rtp_read: Unknown RTP codec 126 received from '87.224.234.108'
[2011-08-26 14:54:09] NOTICE[2382]: rtp.c:1809 ast_rtp_read: Unknown RTP codec 126 received from '87.224.234.108'
[2011-08-26 14:54:09] NOTICE[2382]: rtp.c:1809 ast_rtp_read: Unknown RTP codec 126 received from '87.224.234.108'
Really destroying SIP dialog '4f8d814668cee71b2686de294cf867bd@10.0.0.45' Method: REGISTER
Really destroying SIP dialog '71ca018175d84e415c6a6e6a69db1696@10.0.0.45' Method: REGISTER
Really destroying SIP dialog '063554943ecc102835d8de745b4fbd83@10.0.0.45' Method: REGISTER

<--- SIP read from UDP:193.201.229.35:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.0.0.45:5060;received=195.58.7.132;branch=z9hG4bK40b2ef88;rport=5060
From: "9998" <sip:79221533510@multifon.ru>;tag=as3055c390
To: <sip:89024457140@sbc.megafon.ru>;tag=7AFF324631353641393DD900
Call-ID: 67f0d88941779746256990c822130bb4@multifon.ru
CSeq: 103 INVITE
Allow: OPTIONS,CANCEL,UPDATE
Server: Dialogic-SIP/10.5.3.203 IMG1 0
Content-Length: 0
Contact: <sip:89024457140@193.201.229.35:5060;transport=udp>

<------------->
--- (10 headers 0 lines) ---
-- SIP/m79221533510-00000092 is ringing
Scheduling destruction of SIP dialog '67f0d88941779746256990c822130bb4@multifon.ru' in 6400 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 193.201.229.35:5060:
CANCEL sip:89024457140@sbc.megafon.ru SIP/2.0
Via: SIP/2.0/UDP 10.0.0.45:5060;branch=z9hG4bK40b2ef88;rport
Max-Forwards: 70
From: "9998" <sip:79221533510@multifon.ru>;tag=as3055c390
To: <sip:89024457140@sbc.megafon.ru>
Call-ID: 67f0d88941779746256990c822130bb4@multifon.ru
CSeq: 103 CANCEL
User-Agent: Asterisk PBX 1.6.2.20
Content-Length: 0

---
Scheduling destruction of SIP dialog '67f0d88941779746256990c822130bb4@multifon.ru' in 6400 ms (Method: INVITE)
== Spawn extension (office, 89024457140, 1) exited non-zero on 'SIP/9998-00000091'

<--- SIP read from UDP:193.201.229.35:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.45:5060;received=195.58.7.132;branch=z9hG4bK40b2ef88;rport=5060
From: "9998" <sip:79221533510@multifon.ru>;tag=as3055c390
To: <sip:89024457140@sbc.megafon.ru>;tag=7AFF324631353641393DD900
Call-ID: 67f0d88941779746256990c822130bb4@multifon.ru
CSeq: 103 CANCEL

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

<--- SIP read from UDP:193.201.229.35:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.0.0.45:5060;received=195.58.7.132;branch=z9hG4bK40b2ef88;rport=5060
From: "9998" <sip:79221533510@multifon.ru>;tag=as3055c390
To: <sip:89024457140@sbc.megafon.ru>;tag=7AFF324631353641393DD900
Call-ID: 67f0d88941779746256990c822130bb4@multifon.ru
CSeq: 103 INVITE
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Transmitting (no NAT) to 193.201.229.35:5060:
ACK sip:89024457140@sbc.megafon.ru SIP/2.0
Via: SIP/2.0/UDP 10.0.0.45:5060;branch=z9hG4bK40b2ef88;rport
Max-Forwards: 70
From: "9998" <sip:79221533510@multifon.ru>;tag=as3055c390
To: <sip:89024457140@sbc.megafon.ru>;tag=7AFF324631353641393DD900
Contact: <sip:79221533510@10.0.0.45>
Call-ID: 67f0d88941779746256990c822130bb4@multifon.ru
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.6.2.20
Content-Length: 0

Vlad1983 · Сообщение **Vlad1983** » 26 авг 2011, 14:14

заметил только то, что идет кириллический CALLERID(name)

From: "????????? ?" <sip:79221533510@multifon.ru>;tag=as7bb67aea

попробуй поставить перед Dial на мультифон

Код: Выделить всё

Exten => ...,...,Set(CALLERID(name)=${CALLERID(number)})

IPLDK + мультифон

IPLDK + мультифон

Re: IPLDK + мультифон

Re: IPLDK + мультифон

Re: IPLDK + мультифон

Re: IPLDK + мультифон

Re: IPLDK + мультифон