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Кодеки, входящие и Asterisk

Проблемы Asterisk без вэб-оболочек и их решения

Модераторы: april22, Zavr2008

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Menog
Сообщения: 16
Зарегистрирован: 11 ноя 2013, 10:56

Кодеки, входящие и Asterisk

Сообщение Menog »

При звонке извне через SIP провайдера Zadarma на входящем телефоне Grandstream DP715 не слышу оппонента. При использовании X-Lite (skypemate телефона) все работает.
явно проблема с кодеками, но какая?
привожу настройки и логи:

sip.cfg

Код: Выделить всё

[general]
context=incoming  ; Default context for incoming calls
register => сип:пароль@sip.zadarma.com/сип
disallow=all ;by pfSense
;allow=g729
allow=gsm
allow=ulaw
allow=alaw

[zadarma]
type=friend
username=сип
secret=пароль
fromuser=сип
fromdomain=sip.zadarma.com
host=sipde.zadarma.com
host=siplv.zadarma.com
host=sip2.zadarma.com
nat=yes
dtmfmode=rfc2833
insecure=invite
context=incoming
canreinvite=no
disallow=all
allow=gsm ; GSM consumes far less bandwidth than ulaw
allow=ulaw
allow=alaw

[gs715](!) ; <== обратите внимание, восклицательный знак
; взят в круглые скобки. Это признак шаблона.
type=friend
context=phones
host=dynamic
disallow=all
allow=gsm ; GSM consumes far less bandwidth than ulaw
allow=ulaw
allow=alaw
dtmfmode=rfc2833
secret=qwerty
nat=no

[xlite](!)
; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
"Transmit Silence"=YES
type=friend
;regexten=1234                   ; When they register, create extension 1234
context=phones
host=dynamic                    ; This device needs to register
secret=qwerty
directmedia=no                  ; Typically set to NO if behind NAT
nat=no
disallow=all
allow=gsm ; GSM consumes far less bandwidth than ulaw
allow=ulaw
allow=alaw
call-limit=1
Debug на SkypeMate (слышимость везде ок)

Код: Выделить всё

router*CLI> sip set debug peer 310
SIP Debugging Enabled for IP: 192.168.1.87
    -- Executing [80257@incoming:1] Dial("SIP/5.9.108.25-00000c00", "SIP/310") in new stack
Audio is at 12870
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.1.87:21684:
INVITE sip:310@192.168.1.87:21684;rinstance=a40a3c6496eb4c25 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK0d1a943c
Max-Forwards: 70
From: "SIP2" <sip:60275@192.168.1.100>;tag=as2d0d2209
To: <sip:310@192.168.1.87:21684;rinstance=a40a3c6496eb4c25>
Contact: <sip:60275@192.168.1.100:5060>
Call-ID: 0c94a4040416f8df3324fdde1d4343bd@192.168.1.100:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.19.0
Date: Tue, 17 Dec 2013 06:37:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 313

v=0
o=root 2020075846 2020075846 IN IP4 192.168.1.100
s=Asterisk PBX 1.8.19.0
c=IN IP4 192.168.1.100
t=0 0
m=audio 12870 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called SIP/310

<--- SIP read from UDP:192.168.1.87:21684 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK0d1a943c
Contact: <sip:310@192.168.1.87:21684;rinstance=a40a3c6496eb4c25>
To: <sip:310@192.168.1.87:21684;rinstance=a40a3c6496eb4c25>;tag=6316066b
From: "SIP2"<sip:60275@192.168.1.100>;tag=as2d0d2209
Call-ID: 0c94a4040416f8df3324fdde1d4343bd@192.168.1.100:5060
CSeq: 102 INVITE
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
list_route: hop: <sip:310@192.168.1.87:21684;rinstance=a40a3c6496eb4c25>
    -- SIP/310-00000c01 is ringing

<--- SIP read from UDP:192.168.1.87:21684 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK0d1a943c
Contact: <sip:310@192.168.1.87:21684;rinstance=a40a3c6496eb4c25>
To: <sip:310@192.168.1.87:21684;rinstance=a40a3c6496eb4c25>;tag=6316066b
From: "SIP2"<sip:60275@192.168.1.100>;tag=as2d0d2209
Call-ID: 0c94a4040416f8df3324fdde1d4343bd@192.168.1.100:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 187

v=0
o=- 3 2 IN IP4 192.168.1.87
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.1.87
t=0 0
m=audio 30670 RTP/AVP 3 0 8 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv
<------------->
--- (11 headers 9 lines) ---
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.87:30670
list_route: hop: <sip:310@192.168.1.87:21684;rinstance=a40a3c6496eb4c25>
set_destination: Parsing <sip:310@192.168.1.87:21684;rinstance=a40a3c6496eb4c25> for address/port to send to
set_destination: set destination to 192.168.1.87:21684
Transmitting (no NAT) to 192.168.1.87:21684:
ACK sip:310@192.168.1.87:21684;rinstance=a40a3c6496eb4c25 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK1d3253e8
Max-Forwards: 70
From: "SIP2" <sip:60275@192.168.1.100>;tag=as2d0d2209
To: <sip:310@192.168.1.87:21684;rinstance=a40a3c6496eb4c25>;tag=6316066b
Contact: <sip:60275@192.168.1.100:5060>
Call-ID: 0c94a4040416f8df3324fdde1d4343bd@192.168.1.100:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.19.0
Content-Length: 0


---
    -- SIP/310-00000c01 answered SIP/5.9.108.25-00000c00
    -- Locally bridging SIP/5.9.108.25-00000c00 and SIP/310-00000c01
    -- Locally bridging SIP/5.9.108.25-00000c00 and SIP/310-00000c01
Scheduling destruction of SIP dialog '0c94a4040416f8df3324fdde1d4343bd@192.168.1.100:5060' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:310@192.168.1.87:21684;rinstance=a40a3c6496eb4c25> for address/port to send to
set_destination: set destination to 192.168.1.87:21684
Reliably Transmitting (no NAT) to 192.168.1.87:21684:
BYE sip:310@192.168.1.87:21684;rinstance=a40a3c6496eb4c25 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK56df6892
Max-Forwards: 70
From: "SIP2" <sip:60275@192.168.1.100>;tag=as2d0d2209
To: <sip:310@192.168.1.87:21684;rinstance=a40a3c6496eb4c25>;tag=6316066b
Call-ID: 0c94a4040416f8df3324fdde1d4343bd@192.168.1.100:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.8.19.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
  == Spawn extension (incoming, 80257, 1) exited non-zero on 'SIP/5.9.108.25-00000c00'

<--- SIP read from UDP:192.168.1.87:21684 --->
BYE sip:60275@192.168.1.100:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.87:21684;branch=z9hG4bK-d8754z-6372926b125fd11f-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:310@192.168.1.87:21684;rinstance=a40a3c6496eb4c25>
To: "SIP2"<sip:60275@192.168.1.100>;tag=as2d0d2209
From: <sip:310@192.168.1.87:21684;rinstance=a40a3c6496eb4c25>;tag=6316066b
Call-ID: 0c94a4040416f8df3324fdde1d4343bd@192.168.1.100:5060
CSeq: 2 BYE
User-Agent: X-Lite release 1104o stamp 56125
Reason: SIP;description="User Hung Up"
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 192.168.1.87:21684 (no NAT)
Scheduling destruction of SIP dialog '0c94a4040416f8df3324fdde1d4343bd@192.168.1.100:5060' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 192.168.1.87:21684 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.87:21684;branch=z9hG4bK-d8754z-6372926b125fd11f-1---d8754z-;received=192.168.1.87;rport=21684
From: <sip:310@192.168.1.87:21684;rinstance=a40a3c6496eb4c25>;tag=6316066b
To: "SIP2"<sip:60275@192.168.1.100>;tag=as2d0d2209
Call-ID: 0c94a4040416f8df3324fdde1d4343bd@192.168.1.100:5060
CSeq: 2 BYE
Server: Asterisk PBX 1.8.19.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


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

<--- SIP read from UDP:192.168.1.87:21684 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK56df6892
Contact: <sip:310@192.168.1.87:21684;rinstance=a40a3c6496eb4c25>
To: <sip:310@192.168.1.87:21684;rinstance=a40a3c6496eb4c25>;tag=6316066b
From: "SIP2"<sip:60275@192.168.1.100>;tag=as2d0d2209
Call-ID: 0c94a4040416f8df3324fdde1d4343bd@192.168.1.100:5060
CSeq: 103 BYE
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 0

Debug на Grandstream DP715 (нет звука на сабже)

Код: Выделить всё

router*CLI> sip set debug peer 110
SIP Debugging Enabled for IP: 192.168.1.204
    -- Executing [80257@incoming:1] Dial("SIP/5.9.108.25-00000c07", "SIP/110") in new stack
Audio is at 15278
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.1.204:5062:
INVITE sip:110@192.168.1.204:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK6a8e0b3d
Max-Forwards: 70
From: "SIP2" <sip:60275@192.168.1.100>;tag=as66dad136
To: <sip:110@192.168.1.204:5062>
Contact: <sip:60275@192.168.1.100:5060>
Call-ID: 7cf7e93602f044597670ba221abf4fd0@192.168.1.100:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.19.0
Date: Tue, 17 Dec 2013 06:40:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 313

v=0
o=root 1672227833 1672227833 IN IP4 192.168.1.100
s=Asterisk PBX 1.8.19.0
c=IN IP4 192.168.1.100
t=0 0
m=audio 15278 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called SIP/110

<--- SIP read from UDP:192.168.1.204:5062 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK6a8e0b3d
From: "SIP2" <sip:60275@192.168.1.100>;tag=as66dad136
To: <sip:110@192.168.1.204:5062>
Call-ID: 7cf7e93602f044597670ba221abf4fd0@192.168.1.100:5060
CSeq: 102 INVITE
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream DP715 1.0.0.23
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

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

<--- SIP read from UDP:192.168.1.204:5062 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK6a8e0b3d
From: "SIP2" <sip:60275@192.168.1.100>;tag=as66dad136
To: <sip:110@192.168.1.204:5062>;tag=20502056
Call-ID: 7cf7e93602f044597670ba221abf4fd0@192.168.1.100:5060
CSeq: 102 INVITE
Contact: <sip:110@192.168.1.204:5062>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream DP715 1.0.0.23
Allow-Events: talk, hold
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
list_route: hop: <sip:110@192.168.1.204:5062>
    -- SIP/110-00000c08 is ringing

<--- SIP read from UDP:192.168.1.204:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK6a8e0b3d
From: "SIP2" <sip:60275@192.168.1.100>;tag=as66dad136
To: <sip:110@192.168.1.204:5062>;tag=20502056
Call-ID: 7cf7e93602f044597670ba221abf4fd0@192.168.1.100:5060
CSeq: 102 INVITE
Contact: <sip:110@192.168.1.204:5062>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream DP715 1.0.0.23
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Type: application/sdp
Content-Length: 272

v=0
o=110 8002 8000 IN IP4 192.168.1.204
s=SIP Call
c=IN IP4 192.168.1.204
t=0 0
m=audio 5008 RTP/AVP 0 8 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32-36,54
a=silenceSupp:off - - - -
<------------->
--- (12 headers 13 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.204:5008
list_route: hop: <sip:110@192.168.1.204:5062>
set_destination: Parsing <sip:110@192.168.1.204:5062> for address/port to send to
set_destination: set destination to 192.168.1.204:5062
Transmitting (no NAT) to 192.168.1.204:5062:
ACK sip:110@192.168.1.204:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK0ad69583
Max-Forwards: 70
From: "SIP2" <sip:60275@192.168.1.100>;tag=as66dad136
To: <sip:110@192.168.1.204:5062>;tag=20502056
Contact: <sip:60275@192.168.1.100:5060>
Call-ID: 7cf7e93602f044597670ba221abf4fd0@192.168.1.100:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.19.0
Content-Length: 0


---
    -- SIP/110-00000c08 answered SIP/5.9.108.25-00000c07
    -- Remotely bridging SIP/5.9.108.25-00000c07 and SIP/110-00000c08
set_destination: Parsing <sip:110@192.168.1.204:5062> for address/port to send to
set_destination: set destination to 192.168.1.204:5062
Audio is at 15278
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.1.204:5062:
INVITE sip:110@192.168.1.204:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK614738d6
Max-Forwards: 70
From: "SIP2" <sip:60275@192.168.1.100>;tag=as66dad136
To: <sip:110@192.168.1.204:5062>;tag=20502056
Contact: <sip:60275@192.168.1.100:5060>
Call-ID: 7cf7e93602f044597670ba221abf4fd0@192.168.1.100:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.19.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 1672227833 1672227834 IN IP4 5.9.108.25
s=Asterisk PBX 1.8.19.0
c=IN IP4 5.9.108.25
t=0 0
m=audio 17140 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:192.168.1.204:5062 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK614738d6
From: "SIP2" <sip:60275@192.168.1.100>;tag=as66dad136
To: <sip:110@192.168.1.204:5062>;tag=20502056
Call-ID: 7cf7e93602f044597670ba221abf4fd0@192.168.1.100:5060
CSeq: 103 INVITE
Contact: <sip:110@192.168.1.204:5062>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream DP715 1.0.0.23
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

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

<--- SIP read from UDP:192.168.1.204:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK614738d6
From: "SIP2" <sip:60275@192.168.1.100>;tag=as66dad136
To: <sip:110@192.168.1.204:5062>;tag=20502056
Call-ID: 7cf7e93602f044597670ba221abf4fd0@192.168.1.100:5060
CSeq: 103 INVITE
Contact: <sip:110@192.168.1.204:5062>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream DP715 1.0.0.23
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Type: application/sdp
Content-Length: 272

v=0
o=110 8002 8001 IN IP4 192.168.1.204
s=SIP Call
c=IN IP4 192.168.1.204
t=0 0
m=audio 5008 RTP/AVP 0 8 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32-36,54
a=silenceSupp:off - - - -
<------------->
--- (12 headers 13 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.204:5008
set_destination: Parsing <sip:110@192.168.1.204:5062> for address/port to send to
set_destination: set destination to 192.168.1.204:5062
Transmitting (no NAT) to 192.168.1.204:5062:
ACK sip:110@192.168.1.204:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK7b0fb62f
Max-Forwards: 70
From: "SIP2" <sip:60275@192.168.1.100>;tag=as66dad136
To: <sip:110@192.168.1.204:5062>;tag=20502056
Contact: <sip:60275@192.168.1.100:5060>
Call-ID: 7cf7e93602f044597670ba221abf4fd0@192.168.1.100:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.8.19.0
Content-Length: 0


---
[Dec 17 10:40:03] NOTICE[37381]: chan_sip.c:25310 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 304

<--- SIP read from UDP:192.168.1.204:5062 --->
BYE sip:60275@192.168.1.100:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.204:5062;branch=z9hG4bK803576461;rport
From: <sip:110@192.168.1.204:5062>;tag=20502056
To: "SIP2" <sip:60275@192.168.1.100>;tag=as66dad136
Call-ID: 7cf7e93602f044597670ba221abf4fd0@192.168.1.100:5060
CSeq: 103 BYE
Contact: <sip:110@192.168.1.204:5062>
Max-Forwards: 70
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream DP715 1.0.0.23
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to 192.168.1.204:5062 (no NAT)
Scheduling destruction of SIP dialog '7cf7e93602f044597670ba221abf4fd0@192.168.1.100:5060' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 192.168.1.204:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.204:5062;branch=z9hG4bK803576461;received=192.168.1.204;rport=5062
From: <sip:110@192.168.1.204:5062>;tag=20502056
To: "SIP2" <sip:60275@192.168.1.100>;tag=as66dad136
Call-ID: 7cf7e93602f044597670ba221abf4fd0@192.168.1.100:5060
CSeq: 103 BYE
Server: Asterisk PBX 1.8.19.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


------------>
  == Spawn extension (incoming, 80257, 1) exited non-zero on 'SIP/5.9.108.25-00000c07'
router*CLI>

вообще хотелось бы узнать, как определить по какому кодеку идет связь в обеих случаях, и где это в логах определить?

PS. если звонить по внутренним номерам или изнутри наружу - то проблем нигде не наблюдается
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Vlad1983
Сообщения: 4251
Зарегистрирован: 09 авг 2011, 11:51

Re: Кодеки, входящие и Asterisk

Сообщение Vlad1983 »

[gs715](!)
...
directmedia=no
ЛС: @rostel
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Menog
Сообщения: 16
Зарегистрирован: 11 ноя 2013, 10:56

Re: Кодеки, входящие и Asterisk

Сообщение Menog »

Заработало, спасибо!

Но всё-равно повторю второй вопрос:
- как определить по какому кодеку идет связь в обеих случаях, и где это в логах определить?
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ded
Сообщения: 15628
Зарегистрирован: 26 авг 2010, 19:00

Re: Кодеки, входящие и Asterisk

Сообщение ded »

sip show channels
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Sfinx
Сообщения: 672
Зарегистрирован: 21 июн 2011, 23:40
Откуда: Odessa
Контактная информация:
Контактная информация пользователя Sfinx

Re: Кодеки, входящие и Asterisk

Сообщение Sfinx »

из консоли - core show channel xxx, там будет что-то типа:

...
NativeFormats: (gsm)
WriteFormat: gsm
ReadFormat: gsm
WriteTranscode: No
ReadTranscode: No
......

из диалплана - CHANNEL(audioreadformat)/CHANNEL(audiowriteformat)
Rus

-----------
SfinxSoft
http://sfinxsoft.com
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