Audiocodes m1k with asterisk: no audio

Asterisk Asterisk Support
1

We are using m1k (firmware 5.0) as pstn(T1 CAS) gateway.

No audio (playfile or musiconhold) is happend.

The m1k error log shows below:
1d:6h:17m:49s ( lgr_flow)(1603 ) ---- Outgoing SIP Message to 192.168.1.102:5060 ----

1d:6h:17m:49s INVITE sip:8003@192.168.1.102;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.103;branch=z9hG4bKac120928249 Max-Forwards: 70 From: <sip:109@Mediant 1000>;tag=1c120921112 To: <sip:8003@192.168.1.102;user=phone> Call-ID: 12092043711200061749@192.168.1.103 CSeq: 1 INVITE Contact: <sip:109@192.168.1.103> Supported: em,100rel,timer,replaces,path,early-session,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-Mediant 1000/v.5.00A.035.003 Content-Type: application/sdp Content-Disposition: session Content-Length: 269 v=0 o=AudiocodesGW 120900778 120900489 IN IP4 192.168.1.103 s=Phone-Call c=IN IP4 192.168.1.103 t=0 0 m=audio 60390 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=rtcp:60391 IN IP4 192.168.1.103

1d:6h:17m:49s ( sip_stack)(1605 ) UdpRtxMngr::Transmit 1 INVITE Rtx Left: 6 Dest: c0a80166:5060

1d:6h:17m:49s ( sip_stack)(1606 ) SIPTransaction::ResendLastMessage - Resending last message

1d:6h:17m:49s ( lgr_flow)(1607 ) ---- Incoming SIP Message from 192.168.1.102:5060 ----

1d:6h:17m:49s SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.103;branch=z9hG4bKac120928249;received=192.168.1.103 From: <sip:109@Mediant 1000>;tag=1c120921112 To: <sip:8003@192.168.1.102;user=phone> Call-ID: 12092043711200061749@192.168.1.103 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:8003@192.168.1.102> Content-Length: 0

1d:6h:17m:49s ( sip_stack)(1609 ) AcSIPParser: Problem in SIP Message Headers

1d:6h:17m:49s ( sip_stack)(1610 ) !! [ERROR] AcSIPParser: Parse Error. Unexpected symbol ’ '. Expected ‘>’

1d:6h:17m:49s ( sip_stack)(1611 ) !! [ERROR] Message type: 100 Trying

1d:6h:17m:49s ( sip_stack)(1612 ) !! [ERROR] Source header:

1d:6h:17m:49s ( sip_stack)(1613 ) !! [ERROR] Line: 3. Column: 23

1d:6h:17m:49s SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.103;branch=z9hG4bKac120928249;received=192.168.1.103 From: <sip:109@Mediant 1000>;tag=1c120921112 To: <sip:8003@192.168.1.102;user=phone> Call-ID: 12092043711200061749@192.168.1.103 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:8003@192.168.1.102> Content-Length: 0

*** There is no obvious error in Asterisk sip debug log.

Pls advise.

2

The problem seems * sends an incorrect sip message, as indicated by this line:

1d:6h:17m:49s ( sip_stack)(1610 ) !! [ERROR] AcSIPParser: Parse Error. Unexpected symbol ’ '. Expected ‘>’

Which * version are you using ?
Did you try activate the sip debug in * (cli command “sip set debug”) to see all the sip messags exchanged ?

Bye.

3

version 1.4.0

sip debug log:
210.245.163.62/900.txt

4

I have used audiocodes mp118(fxo, firmware 4.8) and (fxs firmware 5.0) without problem

5

I read the log, seems to me the problem is the Audiocodes send the INVITE for the call, Asterisk replies to the INVITE but then the Audiocodes sends again an INVITE as if it hasn’t receveid the response for the first INVITE, so the sip dialog gets established in a wrong manner and the audio doesn’t flow.

I’m sorry but I don’t know how to solve the problem, it happened to a customer of mine, an Asterisk box connected to an ITSP, the behaviour is similar, see this post:
forums.digium.com/viewtopic.php?t=15372.

Please, if someone more skilled then me could read the sip trace above and write what he thinks about, we would appreciate a lot.

Bye.