[asterisk-users] G.729 codec in pass-thru mode
Kamlesh Kumar
kamlesh_kmr at hotmail.com
Tue May 28 04:13:42 CDT 2013
hello,
201.xxx.xxx.xxx = SIP Softphone which originates the call
xxx.xxx.xxx.xxx = Asterisk server
yyy.yyy.yyy.yyy = ITSP
<--- SIP read from UDP:201.xxx.xxx.xxx:5060 --->
INVITE sip:12127773456 at xxx.xxx.xxx.xxx SIP/2.0
To: <sip:12127773456 at xxx.xxx.xxx.xxx>
From: 100<sip:100 at xxx.xxx.xxx.xxx>;tag=c4446262
Via: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-422974952-1--d87543-;rport
Call-ID: 052fcf17df558f7b
CSeq: 1 INVITE
Contact: <sip:100 at 201.xxx.xxx.xxx:5060>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: eyeBeam release 3007n stamp 17816
Content-Length: 233
v=0
o=- 872302269 872302706 IN IP4 201.xxx.xxx.xxx
s=eyeBeam
c=IN IP4 201.xxx.xxx.xxx
t=0 0
m=audio 8612 RTP/AVP 18 101
a=alt:1 1 : 88385B47 00000038 201.xxx.xxx.xxx 8612
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv
<------------->
[May 28 11:51:34] --- (12 headers 10 lines) ---
[May 28 11:51:34] == Using SIP RTP CoS mark 5
[May 28 11:51:34] Sending to 201.xxx.xxx.xxx : 5060 (no NAT)
[May 28 11:51:34] Using INVITE request as basis request - 052fcf17df558f7b
[May 28 11:51:34] Found peer '100' for '100' from 201.xxx.xxx.xxx:5060
[0K[May 28 11:51:34]
<--- Reliably Transmitting (NAT) to 201.xxx.xxx.xxx:5060 --->
SIP/2.0 401 Unauthorized
v: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-422974952-1--d87543-;received=201.xxx.xxx.xxx;rport=5060
f: 100<sip:100 at xxx.xxx.xxx.xxx>;tag=c4446262
t: <sip:12127773456 at xxx.xxx.xxx.xxx>;tag=as22c91f20
i: 052fcf17df558f7b
CSeq: 1 INVITE
Server: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
k: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="75cff649"
l: 0
<------------>
[May 28 11:51:34] Scheduling destruction of SIP dialog '052fcf17df558f7b' in 32000 ms (Method: INVITE)
[May 28 11:51:34]
<--- SIP read from UDP:201.xxx.xxx.xxx:5060 --->
ACK sip:12127773456 at xxx.xxx.xxx.xxx SIP/2.0
To: <sip:12127773456 at xxx.xxx.xxx.xxx>;tag=as22c91f20
From: 100<sip:100 at xxx.xxx.xxx.xxx>;tag=c4446262
Via: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-422974952-1--d87543-;rport
Call-ID: 052fcf17df558f7b
CSeq: 1 ACK
Content-Length: 0
<------------->
[May 28 11:51:34] --- (7 headers 0 lines) ---
[May 28 11:51:34]
<--- SIP read from UDP:201.xxx.xxx.xxx:5060 --->
INVITE sip:12127773456 at xxx.xxx.xxx.xxx SIP/2.0
To: <sip:12127773456 at xxx.xxx.xxx.xxx>
From: 100<sip:100 at xxx.xxx.xxx.xxx>;tag=c4446262
Via: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-61806499-1--d87543-;rport
Call-ID: 052fcf17df558f7b
CSeq: 2 INVITE
Contact: <sip:100 at 201.xxx.xxx.xxx:5060>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: eyeBeam release 3007n stamp 17816
Authorization: Digest username="100",realm="asterisk",nonce="75cff649",uri="sip:12127773456 at xxx.xxx.xxx.xxx",response="ffeaf69c547dd3d1252e1bc7ab614fea",algorithm=MD5
Content-Length: 233
v=0
o=- 872302269 872302706 IN IP4 201.xxx.xxx.xxx
s=eyeBeam
c=IN IP4 201.xxx.xxx.xxx
t=0 0
m=audio 8612 RTP/AVP 18 101
a=alt:1 1 : 88385B47 00000038 201.xxx.xxx.xxx 8612
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv
<------------->
[May 28 11:51:34] --- (13 headers 10 lines) ---
[May 28 11:51:34] Sending to 201.xxx.xxx.xxx : 5060 (NAT)
[May 28 11:51:34] Using INVITE request as basis request - 052fcf17df558f7b
[May 28 11:51:34] Found peer '100' for '100' from 201.xxx.xxx.xxx:5060
[May 28 11:51:34] Found RTP audio format 18
[May 28 11:51:34] Found RTP audio format 101
[May 28 11:51:34] Found audio description format telephone-event for ID 101
[May 28 11:51:34] Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729)
[May 28 11:51:34] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[May 28 11:51:34] Peer audio RTP is at port 201.xxx.xxx.xxx:8612
[May 28 11:51:34] Looking for 12127773456 in asterisk (domain xxx.xxx.xxx.xxx)
[May 28 11:51:34] list_route: hop: <sip:100 at 201.xxx.xxx.xxx:5060>
[May 28 11:51:34]
<--- Transmitting (NAT) to 201.xxx.xxx.xxx:5060 --->
SIP/2.0 100 Trying
v: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-61806499-1--d87543-;received=201.xxx.xxx.xxx;rport=5060
f: 100<sip:100 at xxx.xxx.xxx.xxx>;tag=c4446262
t: <sip:12127773456 at xxx.xxx.xxx.xxx>
i: 052fcf17df558f7b
CSeq: 2 INVITE
Server: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
k: replaces, timer
m: <sip:12127773456 at xxx.xxx.xxx.xxx>
l: 0
<------------>
[May 28 11:51:34] -- Executing AGI("SIP/100-0000115f", "call.php")
[May 28 11:51:34] -- Launched AGI Script /var/lib/asterisk/agi-bin/call.php
[May 28 11:51:34] -- AGI Script Executing Application: (Dial) Options: (SIP/yyy.yyy.yyy.yyy/12127773456)
[May 28 11:51:34] == Using SIP RTP CoS mark 5
[May 28 11:51:34] -- Couldn't call yyy.yyy.yyy.yyy/12127773456
[May 28 11:51:34] Scheduling destruction of SIP dialog '142182ef20750fda512f8d2b0b071ad6 at xxx.xxx.xxx.xxx' in 32000 ms (Method: INVITE)
[May 28 11:51:34] == Everyone is busy/congested at this time (0:0/0/0)
[May 28 11:51:34] -- <SIP/100-0000115f>AGI Script call.php completed, returning 0
[May 28 11:51:34] -- Auto fallthrough, channel 'SIP/100-0000115f' status is 'CHANUNAVAIL'
[May 28 11:51:34]
<--- Reliably Transmitting (NAT) to 201.xxx.xxx.xxx:5060 --->
SIP/2.0 503 Service Unavailable
v: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-61806499-1--d87543-;received=201.xxx.xxx.xxx;rport=5060
f: 100<sip:100 at xxx.xxx.xxx.xxx>;tag=c4446262
t: <sip:12127773456 at xxx.xxx.xxx.xxx>;tag=as4e329d09
i: 052fcf17df558f7b
CSeq: 2 INVITE
Server: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
k: replaces, timer
l: 0
<------------>
[May 28 11:51:34]
<--- SIP read from UDP:201.xxx.xxx.xxx:5060 --->
ACK sip:12127773456 at xxx.xxx.xxx.xxx SIP/2.0
To: <sip:12127773456 at xxx.xxx.xxx.xxx>;tag=as4e329d09
From: 100<sip:100 at xxx.xxx.xxx.xxx>;tag=c4446262
Via: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-61806499-1--d87543-;rport
Call-ID: 052fcf17df558f7b
CSeq: 2 ACK
Content-Length: 0
<------------->
[May 28 11:51:34] --- (7 headers 0 lines) ---
[May 28 11:51:34] -- Executing AGI("SIP/100-0000115f", "hangup.php")
[May 28 11:51:34] -- Launched AGI Script /var/lib/asterisk/agi-bin/hangup.php
[May 28 11:51:34] -- <SIP/100-0000115f>AGI Script hangup.php completed, returning 0
Thanks,
Kamlesh
> From: EWieling at nyigc.com
> To: asterisk-users at lists.digium.com
> Date: Mon, 27 May 2013 11:51:53 -0400
> Subject: Re: [asterisk-users] G.729 codec in pass-thru mode
>
> Show us the sip debug for a failed call.
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Kamlesh Kumar
> Sent: Monday, May 27, 2013 2:20 AM
> To: asterisk-users at lists.digium.com
> Subject: [asterisk-users] G.729 codec in pass-thru mode
>
> Hello,
> Trying to use g729 in pass-thru mode.
> Call flow:
> SIP IP Phone (G.729)-->Asterisk(1.6.2.9)--->SIP Trunk to ITSP(G.729) When using G.729, call is not getting connected. Below is the extract from CLI.
> == Using SIP RTP CoS mark 5
> -- Executing [12127773456 at default:1] AGI("SIP/100-00000000", "call.php") in new stack
> -- Launched AGI Script /var/lib/asterisk/agi-bin/call.php
> -- AGI Script Executing Application: (Dial) Options: (SIP/xxx.xxx.xxx.xxx/12127773456)
> -- Couldn't call xxx.xxx.xxx.xxx/12127773456 == Everyone is busy/congested at this time (0:0/0/0)
> -- <SIP/100-00000000>AGI Script call.php completed, returning 0
> -- Auto fallthrough, channel 'SIP/100-00000000' status is 'CHANUNAVAIL'
>
> If I use, ulaw, call works fine.
>
> Thanks,
> Kamlesh
>
>
> --
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