Freebsd + Asterisk 1.6.2.18 + g729 =?

[StaLeWaR]

Добрый День!
с * на оно т.к занимаюсь меньше недели, кодек собрал сам, загрузил, с софтофона с поддержкой g729 делаю звонок на такой же софотофон в логах

Код: Выделить всё

    -- Executing [202@office:1] Dial("SIP/201-00000007", "SIP/202,40") in new stack
    -- Called 202
    -- SIP/202-00000008 is ringing
[May  2 11:29:12] WARNING[924]: chan_sip.c:9101 process_sdp_o: SDP syntax error in o= line version
    -- SIP/202-00000008 answered SIP/201-00000007
  == Spawn extension (office, 202, 1) exited non-zero on 'SIP/201-00000007'

куда копать ума не приложу?

[ded]

sip set debug peer 202
и изучать как выглядит SDP offer. Будет что-то типа

INVITE sip:destino@example.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.33:5060;branch=z9hG4bKjyofoqmp
Max-Forwards: 70
To: <sip:destino@example.com>
From: "Alice" <sip:alice@example.com>;tag=nrrrx
Call-ID: xetazdjyktlpsfo@192.168.1.33
CSeq: 800 INVITE
Contact: <sip:alice@192.168.1.33:5060>
Content-Type: application/sdp
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
Supported: replaces,norefersub,100rel
User-Agent: Twinkle/1.1
Content-Length: 312

v=0
o=alice 2890844526 2890844526 IN IP4 host.anywhere.com
s=
c=IN IP4 host.anywhere.com
t=0 0
m=audio 62986 RTP/AVP 0 4 18
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=inactive

Или сходу указать в секции [general] файла sip.conf
ignoresdpversion=no

[StaLeWaR]

Выполнил:

Код: Выделить всё

SIP Debugging Enabled for IP: 192.168.0.105:5060
    -- Executing [202@office:1] Dial("SIP/201-00000006", "SIP/202,40") in new stack
Audio is at 192.168.0.110 port 11232
Adding codec 0x100 (g729) to SDP
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 (NAT) to 192.168.0.105:5060:
INVITE sip:202@0.0.0.0:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.110:5060;branch=z9hG4bK1be58ccb;rport
Max-Forwards: 70
From: "user1" <sip:201@192.168.0.110>;tag=as64bb48e1
To: <sip:202@0.0.0.0:5060>
Contact: <sip:201@192.168.0.110>
Call-ID: 0889c9d22eb743c768de29722668f655@192.168.0.110
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.18
Date: Mon, 02 May 2011 14:46:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
upported: replaces, timer
Content-Type: application/sdp
Content-Length: 333

v=0
o=root 1503763652 1503763652 IN IP4 192.168.0.110
s=Asterisk PBX 1.6.2.18
c=IN IP4 192.168.0.110
t=0 0
m=audio 11232 RTP/AVP 18 3 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
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=ptime:20
a=sendrecv

---
    -- Called 202

<--- SIP read from UDP:192.168.0.105:5060 --->
SIP/2.0 180 Ringing
To: <sip:202@0.0.0.0:5060>;tag=c5167e18
From: "user1"<sip:201@192.168.0.110>;tag=as64bb48e1
Via: SIP/2.0/UDP 192.168.0.110:5060;branch=z9hG4bK1be58ccb;rport=5060;received=192.168.0.110
Call-ID: 0889c9d22eb743c768de29722668f655@192.168.0.110
CSeq: 102 INVITE
Contact: <sip:0.0.0.0:5060>
User-Agent: Firefly 2.0
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
    -- SIP/202-00000007 is ringing

<--- SIP read from UDP:192.168.0.105:5060 --->
SIP/2.0 200 OK
To: <sip:202@0.0.0.0:5060>;tag=c5167e18
From: "user1"<sip:201@192.168.0.110>;tag=as64bb48e1
Via: SIP/2.0/UDP 192.168.0.110:5060;branch=z9hG4bK1be58ccb;rport=5060;received=192.168.0.110
Call-ID: 0889c9d22eb743c768de29722668f655@192.168.0.110
CSeq: 102 INVITE
Contact: <sip:0.0.0.0:5060>
Content-Type: application/sdp
User-Agent: Firefly 2.0
Content-Length: 180

v=0
o= 0 0 IN IP4 192.168.0.105
s=
c=IN IP4 192.168.0.105
t=0 0
m=audio 5000 RTP/AVP 18 101
a=rtpmap:18 G729/8000/1/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

<------------->
--- (10 headers 9 lines) ---
[May  2 14:46:54] WARNING[923]: chan_sip.c:9101 process_sdp_o: SDP syntax error in o= line version
list_route: hop: <sip:0.0.0.0:5060>
set_destination: Parsing <sip:0.0.0.0:5060> for address/port to send to
set_destination: set destination to 0.0.0.0, port 5060
Transmitting (NAT) to 192.168.0.105:5060:
ACK sip:0.0.0.0:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.110:5060;branch=z9hG4bK51604365;rport
Max-Forwards: 70
From: "user1" <sip:201@192.168.0.110>;tag=as64bb48e1
To: <sip:202@0.0.0.0:5060>;tag=c5167e18
Contact: <sip:201@192.168.0.110>
Call-ID: 0889c9d22eb743c768de29722668f655@192.168.0.110
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.18
Content-Length: 0


---
set_destination: Parsing <sip:0.0.0.0:5060> for address/port to send to
set_destination: set destination to 0.0.0.0, port 5060
Reliably Transmitting (NAT) to 192.168.0.105:5060:
BYE sip:0.0.0.0:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.110:5060;branch=z9hG4bK7b5a7d0b;rport
Max-Forwards: 70
From: "user1" <sip:201@192.168.0.110>;tag=as64bb48e1
To: <sip:202@0.0.0.0:5060>;tag=c5167e18
Call-ID: 0889c9d22eb743c768de29722668f655@192.168.0.110
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.6.2.18
X-Asterisk-HangupCause: Unknown
X-Asterisk-HangupCauseCode: 0
Content-Length: 0


--
Scheduling destruction of SIP dialog '0889c9d22eb743c768de29722668f655@192.168.0.110' in 32000 ms (Method: INVITE)
    -- SIP/202-00000007 answered SIP/201-00000006
Scheduling destruction of SIP dialog '0889c9d22eb743c768de29722668f655@192.168.0.110' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:0.0.0.0:5060> for address/port to send to
set_destination: set destination to 0.0.0.0, port 5060
Reliably Transmitting (NAT) to 192.168.0.105:5060:
BYE sip:0.0.0.0:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.110:5060;branch=z9hG4bK5c2e04f5;rport
Max-Forwards: 70
From: "user1" <sip:201@192.168.0.110>;tag=as64bb48e1
To: <sip:202@0.0.0.0:5060>;tag=c5167e18
Call-ID: 0889c9d22eb743c768de29722668f655@192.168.0.110
CSeq: 104 BYE
User-Agent: Asterisk PBX 1.6.2.18
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
  == Spawn extension (office, 202, 1) exited non-zero on 'SIP/201-00000006'

<--- SIP read from UDP:192.168.0.105:5060 --->
SIP/2.0 200 OK
To: <sip:202@0.0.0.0:5060>;tag=c5167e18
From: "user1" <sip:201@192.168.0.110>;tag=as64bb48e1
Via: SIP/2.0/UDP 192.168.0.110:5060;branch=z9hG4bK7b5a7d0b;rport=5060;received=192.168.0.110
Call-ID: 0889c9d22eb743c768de29722668f655@192.168.0.110
CSeq: 103 BYE
Contact: <sip:0.0.0.0:5060>
User-Agent: Firefly 2.0
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
Retransmitting #1 (NAT) to 192.168.0.105:5060:
BYE sip:0.0.0.0:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.110:5060;branch=z9hG4bK5c2e04f5;rport
Max-Forwards: 70
From: "user1" <sip:201@192.168.0.110>;tag=as64bb48e1
To: <sip:202@0.0.0.0:5060>;tag=c5167e18
Call-ID: 0889c9d22eb743c768de29722668f655@192.168.0.110
CSeq: 104 BYE
User-Agent: Asterisk PBX 1.6.2.18
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
Retransmitting #2 (NAT) to 192.168.0.105:5060:
BYE sip:0.0.0.0:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.110:5060;branch=z9hG4bK5c2e04f5;rport
Max-Forwards: 70
From: "user1" <sip:201@192.168.0.110>;tag=as64bb48e1
To: <sip:202@0.0.0.0:5060>;tag=c5167e18
Call-ID: 0889c9d22eb743c768de29722668f655@192.168.0.110
CSeq: 104 BYE
User-Agent: Asterisk PBX 1.6.2.18
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
Retransmitting #3 (NAT) to 192.168.0.105:5060:
BYE sip:0.0.0.0:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.110:5060;branch=z9hG4bK5c2e04f5;rport
Max-Forwards: 70
From: "user1" <sip:201@192.168.0.110>;tag=as64bb48e1
To: <sip:202@0.0.0.0:5060>;tag=c5167e18
Call-ID: 0889c9d22eb743c768de29722668f655@192.168.0.110
CSeq: 104 BYE
User-Agent: Asterisk PBX 1.6.2.18
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
Retransmitting #4 (NAT) to 192.168.0.105:5060:
BYE sip:0.0.0.0:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.110:5060;branch=z9hG4bK5c2e04f5;rport
Max-Forwards: 70
From: "user1" <sip:201@192.168.0.110>;tag=as64bb48e1
To: <sip:202@0.0.0.0:5060>;tag=c5167e18
Call-ID: 0889c9d22eb743c768de29722668f655@192.168.0.110
CSeq: 104 BYE
User-Agent: Asterisk PBX 1.6.2.18
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
Retransmitting #5 (NAT) to 192.168.0.105:5060:
BYE sip:0.0.0.0:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.110:5060;branch=z9hG4bK5c2e04f5;rport
Max-Forwards: 70
From: "user1" <sip:201@192.168.0.110>;tag=as64bb48e1
To: <sip:202@0.0.0.0:5060>;tag=c5167e18
Call-ID: 0889c9d22eb743c768de29722668f655@192.168.0.110
CSeq: 104 BYE
User-Agent: Asterisk PBX 1.6.2.18
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
Retransmitting #6 (NAT) to 192.168.0.105:5060:
BYE sip:0.0.0.0:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.110:5060;branch=z9hG4bK5c2e04f5;rport
Max-Forwards: 70
From: "user1" <sip:201@192.168.0.110>;tag=as64bb48e1
To: <sip:202@0.0.0.0:5060>;tag=c5167e18
Call-ID: 0889c9d22eb743c768de29722668f655@192.168.0.110
CSeq: 104 BYE
User-Agent: Asterisk PBX 1.6.2.18
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
[May  2 14:47:14] WARNING[923]: chan_sip.c:3968 retrans_pkt: Maximum retries exceeded on transmission 0889c9d22eb743c768de29722668f655@192.168.0.110 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Really destroying SIP dialog '0889c9d22eb743c768de29722668f655@192.168.0.110' Method: INVITE

здесь вроде
o=root 1503763652 1503763652 IN IP4 192.168.0.110
понять не могу в чем проблема, вроде все как и должно быть
объясните пожалуйста неоптныму

[ded]

Не, так не всё должно быть.

здесь вроде
o=root 1503763652 1503763652 IN IP4 192.168.0.110

а чуть ниже
o= 0 0 IN IP4 192.168.0.105

Получается какая-то ерунда -

set_destination: Parsing <sip:0.0.0.0:5060> for address/port to send to
set_destination: set destination to 0.0.0.0, port 5060
Transmitting (NAT) to 192.168.0.105:5060:
ACK sip:0.0.0.0:5060 SIP/2.0

которая ретрансмитится (перепосылается без ответа) несколько раз:
Retransmitting #4 (NAT) to 192.168.0.105:5060:
BYE sip:0.0.0.0:5060 SIP/2.0

[StaLeWaR]

хм...
а что это значит?

[ded]

С первой прикидки вот что значит:
* направо пойдёшь - в избу-читальню придёшь http://asterisk.ru/knowledgebase/books знанья обретёшь;
* налево пойдёшь - в платный суппорт попадёшь.

[StaLeWaR]

ок, пасиба почитаю =)
посоветуйте софтофон с подержкой g729 на windows

[ded]

По определению - на винду, да ещё с 729-м - только платные, ввиду платности самого кодека.
Ну да есть вроде - Firefly-g729.exe

Я вот не понял, если звонить по alaw | ulaw - всё работает? Сомневаюсь я...
Я думаю, что можно покрутить опцию ignoresdpversion= всё таки.

[StaLeWaR]

судя по всему * говарить по g729 умеет, когда проговаривает мне сообщение о том что номер недоступен

Код: Выделить всё

[May  2 16:25:14] NOTICE[923]: channel.c:3162 __ast_read: Dropping incompatible voice frame on SIP/201-0000001d of format gsm since our native format has changed to 0x100 (g729)

да при такой конфигурации мне звонки проходят, а вот после снятия трубки в софтофоне - сразу заръеденение.

[ded]

У вас возмсожно не объявлен доступным сам g729 в sip.conf
Проверить -
sip show settings

судя по Dropping incompatible voice frame on SIP/201-0000001d of format gsm - у вас там стоит gsm & ulaw & alaw
Кроме того сам звонок пытается повязаться именно по gsm.