[asterisk-users] G.729 codec in pass-thru mode
Kamlesh Kumar
kamlesh_kmr at hotmail.com
Mon Jun 3 23:54:29 CDT 2013
Matthew,
SIP.conf
[100]
username=100
secret=password
type=friend
host=dynamic
nat=yes
canreinvite=no
insecure=port
disallow=all
allow=ulaw
allow=alaw
allow=g729
context=asterisk
qualify=no
dialplan
[asterisk]
exten => _X.,1,AGI(call.php)
exten => h,1,AGI(hangup.php)
SIP Trace:
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=7c0c4b22
Via: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-181300058-1--d87543-;rport
Call-ID: 6601fe453f41d566
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: 228
v=0
o=- 7847157 7847631 IN IP4 201.xxx.xxx.xxx
s=eyeBeam
c=IN IP4 201.xxx.xxx.xxx
t=0 0
m=audio 8614 RTP/AVP 0 101
a=alt:1 1 : 05A48429 0000007F 201.xxx.xxx.xxx 8614
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv
<------------->
[Jun 3 13:11:27] --- (12 headers 10 lines) ---
[Jun 3 13:11:27] == Using SIP RTP CoS mark 5
[Jun 3 13:11:27] Sending to 201.xxx.xxx.xxx : 5060 (no NAT)
[Jun 3 13:11:27] Using INVITE request as basis request - 6601fe453f41d566
[Jun 3 13:11:27] Found peer '100' for '100' from 201.xxx.xxx.xxx:5060
[Jun 3 13:11:27]
<--- Reliably Transmitting (NAT) to 201.xxx.xxx.xxx:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-181300058-1--d87543-;received=201.xxx.xxx.xxx;rport=5060
From: 100<sip:100 at xxx.xxx.xxx.xxx>;tag=7c0c4b22
To: <sip:12127773456 at xxx.xxx.xxx.xxx>;tag=as1999a2fb
Call-ID: 6601fe453f41d566
CSeq: 1 INVITE
Server: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0c9deee3"
Content-Length: 0
<------------>
[Jun 3 13:11:27] Scheduling destruction of SIP dialog '6601fe453f41d566' in 32000 ms (Method: INVITE)
[Jun 3 13:11:27]
<--- 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=7c0c4b22
Via: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-254704504-1--d87543-;rport
Call-ID: 6601fe453f41d566
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="0c9deee3",uri="sip:12127773456 at xxx.xxx.xxx.xxx",response="e84d9090cfc8e60f94768583218ae0ad",algorithm=MD5
Content-Length: 228
v=0
o=- 7847157 7847631 IN IP4 201.xxx.xxx.xxx
s=eyeBeam
c=IN IP4 201.xxx.xxx.xxx
t=0 0
m=audio 8614 RTP/AVP 0 101
a=alt:1 1 : 05A48429 0000007F 201.xxx.xxx.xxx 8614
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv
<------------->
[Jun 3 13:11:27] --- (13 headers 10 lines) ---
[Jun 3 13:11:27] Sending to 201.xxx.xxx.xxx : 5060 (NAT)
[Jun 3 13:11:27] Using INVITE request as basis request - 6601fe453f41d566
[Jun 3 13:11:27] Found peer '100' for '100' from 201.xxx.xxx.xxx:5060
[Jun 3 13:11:27] Found RTP audio format 0
[Jun 3 13:11:27] Found RTP audio format 101
[Jun 3 13:11:27] Found audio description format telephone-event for ID 101
[Jun 3 13:11:27] Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
[Jun 3 13:11:27] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Jun 3 13:11:27] Peer audio RTP is at port 201.xxx.xxx.xxx:8614
[Jun 3 13:11:27] Looking for 12127773456 in asterisk (domain xxx.xxx.xxx.xxx)
[Jun 3 13:11:27] list_route: hop: <sip:100 at 201.xxx.xxx.xxx:5060>
[Jun 3 13:11:27]
<--- Transmitting (NAT) to 201.xxx.xxx.xxx:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-254704504-1--d87543-;received=201.xxx.xxx.xxx;rport=5060
From: 100<sip:100 at xxx.xxx.xxx.xxx>;tag=7c0c4b22
To: <sip:12127773456 at xxx.xxx.xxx.xxx>
Call-ID: 6601fe453f41d566
CSeq: 2 INVITE
Server: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:12127773456 at xxx.xxx.xxx.xxx>
Content-Length: 0
<------------>
[Jun 3 13:11:27] -- Executing [12127773456 at asterisk:1] AGI("SIP/100-000034d8", "call.php") in new stack
[Jun 3 13:11:27] -- Launched AGI Script /var/lib/asterisk/agi-bin/call.php
[Jun 3 13:11:28] -- AGI Script Executing Application: (Dial) Options: (SIP/yyy.yyy.yyy.yyy/12127773456)
[Jun 3 13:11:28] == Using SIP RTP CoS mark 5
[Jun 3 13:11:28] Audio is at xxx.xxx.xxx.xxx port 56248
[Jun 3 13:11:28] Adding codec 0x4 (ulaw) to SDP
[Jun 3 13:11:28] Adding non-codec 0x1 (telephone-event) to SDP
[Jun 3 13:11:28] Reliably Transmitting (NAT) to yyy.yyy.yyy.yyy:5060:
INVITE sip:12127773456 at yyy.yyy.yyy.yyy SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK15380659;rport
Max-Forwards: 70
From: "100" <sip:100 at xxx.xxx.xxx.xxx>;tag=as643c20b1
To: <sip:12127773456 at yyy.yyy.yyy.yyy>
Contact: <sip:100 at xxx.xxx.xxx.xxx>
Call-ID: 07714ae4593feb5c3e42b3a01cf4aa20 at xxx.xxx.xxx.xxx
CSeq: 102 INVITE
User-Agent: PBX
Date: Mon, 03 Jun 2013 13:11:28 IST
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 248
v=0
o=WP 1365830908 1365830908 IN IP4 xxx.xxx.xxx.xxx
s=PBX
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 56248 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
[Jun 3 13:11:28] -- Called yyy.yyy.yyy.yyy/12127773456
[Jun 3 13:11:28]
<--- SIP read from UDP:yyy.yyy.yyy.yyy:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK15380659;rport=5060
From: "100" <sip:100 at xxx.xxx.xxx.xxx>;tag=as643c20b1
To: <sip:12127773456 at yyy.yyy.yyy.yyy>;tag=gK029aaa8c
Call-ID: 07714ae4593feb5c3e42b3a01cf4aa20 at xxx.xxx.xxx.xxx
CSeq: 102 INVITE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
<--- SIP read from UDP:yyy.yyy.yyy.yyy:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK15380659;rport=5060
From: "100" <sip:100 at xxx.xxx.xxx.xxx>;tag=as643c20b1
To: <sip:12127773456 at yyy.yyy.yyy.yyy>;tag=gK029aaa8c
Call-ID: 07714ae4593feb5c3e42b3a01cf4aa20 at xxx.xxx.xxx.xxx
CSeq: 102 INVITE
Contact: <sip:12127773456 at yyy.yyy.yyy.yyy:5060>
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS
Content-Length: 234
Content-Disposition: session; handling=required
Content-Type: application/sdp
v=0
o=Sonus_UAC 24592 17457 IN IP4 yyy.yyy.yyy.yyy
s=SIP Media Capabilities
c=IN IP4 zzz.zzz.zzz.zzz
t=0 0
m=audio 21996 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:20
<------------->
[Jun 3 13:11:31] --- (11 headers 11 lines) ---
[Jun 3 13:11:31] Found RTP audio format 0
[Jun 3 13:11:31] Found RTP audio format 101
[Jun 3 13:11:31] Found audio description format PCMU for ID 0
[Jun 3 13:11:31] Found audio description format telephone-event for ID 101
[Jun 3 13:11:31] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
[Jun 3 13:11:31] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Jun 3 13:11:31] Peer audio RTP is at port zzz.zzz.zzz.zzz:21996
[Jun 3 13:11:31] -- SIP/yyy.yyy.yyy.yyy-000034d9 is making progress passing it to SIP/100-000034d8
[Jun 3 13:11:31] Audio is at xxx.xxx.xxx.xxx port 26042
[Jun 3 13:11:31] Adding codec 0x4 (ulaw) to SDP
[Jun 3 13:11:31] Adding non-codec 0x1 (telephone-event) to SDP
[Jun 3 13:11:31]
<--- Transmitting (NAT) to 201.xxx.xxx.xxx:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-254704504-1--d87543-;received=201.xxx.xxx.xxx;rport=5060
From: 100<sip:100 at xxx.xxx.xxx.xxx>;tag=7c0c4b22
To: <sip:12127773456 at xxx.xxx.xxx.xxx>;tag=as654371b0
Call-ID: 6601fe453f41d566
CSeq: 2 INVITE
Server: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:12127773456 at xxx.xxx.xxx.xxx>
Content-Type: application/sdp
Content-Length: 248
v=0
o=WP 1745765504 1745765504 IN IP4 xxx.xxx.xxx.xxx
s=PBX
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 26042 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
[0K[Jun 3 13:11:32]
<--- SIP read from UDP:yyy.yyy.yyy.yyy:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK15380659;rport=5060
From: "100" <sip:100 at xxx.xxx.xxx.xxx>;tag=as643c20b1
To: <sip:12127773456 at yyy.yyy.yyy.yyy>;tag=gK029aaa8c
Call-ID: 07714ae4593feb5c3e42b3a01cf4aa20 at xxx.xxx.xxx.xxx
CSeq: 102 INVITE
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Contact: <sip:12127773456 at yyy.yyy.yyy.yyy:5060>
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS
Require: timer
Supported: timer,replaces
Session-Expires: 1800;refresher=uac
Content-Length: 234
Content-Disposition: session; handling=required
Content-Type: application/sdp
v=0
o=Sonus_UAC 24592 17457 IN IP4 yyy.yyy.yyy.yyy
s=SIP Media Capabilities
c=IN IP4 zzz.zzz.zzz.zzz
t=0 0
m=audio 21996 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:20
<------------->
[Jun 3 13:11:32] --- (15 headers 11 lines) ---
[Jun 3 13:11:32] list_route: hop: <sip:12127773456 at yyy.yyy.yyy.yyy:5060>
[Jun 3 13:11:32] set_destination: Parsing <sip:12127773456 at yyy.yyy.yyy.yyy:5060> for address/port to send to
[Jun 3 13:11:32] set_destination: set destination to yyy.yyy.yyy.yyy, port 5060
[Jun 3 13:11:32] Transmitting (NAT) to yyy.yyy.yyy.yyy:5060:
ACK sip:12127773456 at yyy.yyy.yyy.yyy:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK570164a3;rport
Max-Forwards: 70
From: "100" <sip:100 at xxx.xxx.xxx.xxx>;tag=as643c20b1
To: <sip:12127773456 at yyy.yyy.yyy.yyy>;tag=gK029aaa8c
Contact: <sip:100 at xxx.xxx.xxx.xxx>
Call-ID: 07714ae4593feb5c3e42b3a01cf4aa20 at xxx.xxx.xxx.xxx
CSeq: 102 ACK
User-Agent: PBX
Content-Length: 0
[Jun 3 13:11:32] -- SIP/yyy.yyy.yyy.yyy-000034d9 answered SIP/100-000034d8
[Jun 3 13:11:32] Audio is at xxx.xxx.xxx.xxx port 26042
[Jun 3 13:11:32] Adding codec 0x4 (ulaw) to SDP
[Jun 3 13:11:32] Adding non-codec 0x1 (telephone-event) to SDP
[Jun 3 13:11:32]
<--- Reliably Transmitting (NAT) to 201.xxx.xxx.xxx:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-254704504-1--d87543-;received=201.xxx.xxx.xxx;rport=5060
From: 100<sip:100 at xxx.xxx.xxx.xxx>;tag=7c0c4b22
To: <sip:12127773456 at xxx.xxx.xxx.xxx>;tag=as654371b0
Call-ID: 6601fe453f41d566
CSeq: 2 INVITE
Server: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:12127773456 at xxx.xxx.xxx.xxx>
Content-Type: application/sdp
Content-Length: 248
v=0
o=WP 1745765504 1745765505 IN IP4 xxx.xxx.xxx.xxx
s=PBX
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 26042 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
[Jun 3 13:11:32] -- Packet2Packet bridging SIP/100-000034d8 and SIP/yyy.yyy.yyy.yyy-000034d9
[Jun 3 13:11:32]
<--- 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=as654371b0
From: 100<sip:100 at xxx.xxx.xxx.xxx>;tag=7c0c4b22
Via: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-558309054-1--d87543-;rport
Call-ID: 6601fe453f41d566
CSeq: 2 ACK
Contact: <sip:100 at 201.xxx.xxx.xxx:5060>
Max-Forwards: 70
User-Agent: eyeBeam release 3007n stamp 17816
Authorization: Digest username="100",realm="asterisk",nonce="0c9deee3",uri="sip:12127773456 at xxx.xxx.xxx.xxx",response="e84d9090cfc8e60f94768583218ae0ad",algorithm=MD5
Content-Length: 0
<------------->
[Jun 3 13:11:32] --- (11 headers 0 lines) ---
[Jun 3 13:11:32] Really destroying SIP dialog '2d60af2732e232cf3dec876f601c38d9 at 127.0.0.1' Method: REGISTER
[Jun 3 13:11:34]
<--- SIP read from UDP:201.xxx.xxx.xxx:5060 --->
BYE sip:12127773456 at xxx.xxx.xxx.xxx SIP/2.0
To: <sip:12127773456 at xxx.xxx.xxx.xxx>;tag=as654371b0
From: 100<sip:100 at xxx.xxx.xxx.xxx>;tag=7c0c4b22
Via: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-845749512-1--d87543-;rport
Call-ID: 6601fe453f41d566
CSeq: 3 BYE
Contact: <sip:100 at 201.xxx.xxx.xxx:5060>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: eyeBeam release 3007n stamp 17816
Authorization: Digest username="100",realm="asterisk",nonce="0c9deee3",uri="sip:12127773456 at xxx.xxx.xxx.xxx",response="ae82a7f181072735f72c23f2c63020e7",algorithm=MD5
Content-Length: 0
[Jun 3 13:11:35] --- (12 headers 0 lines) ---
[Jun 3 13:11:35] Sending to 201.xxx.xxx.xxx : 5060 (NAT)
[Jun 3 13:11:35]
<--- Transmitting (NAT) to 201.xxx.xxx.xxx:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-845749512-1--d87543-;received=201.xxx.xxx.xxx;rport=5060
From: 100<sip:100 at xxx.xxx.xxx.xxx>;tag=7c0c4b22
To: <sip:12127773456 at xxx.xxx.xxx.xxx>;tag=as654371b0
Call-ID: 6601fe453f41d566
CSeq: 3 BYE
Server: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
[Jun 3 13:11:35] -- Executing [h at asterisk:1] AGI("SIP/100-000034d8", "hangup.php") in new stack
[Jun 3 13:11:35] -- Launched AGI Script /var/lib/asterisk/agi-bin/hangup.php
[Jun 3 13:11:35] -- <SIP/100-000034d8>AGI Script hangup.php completed, returning 0
[Jun 3 13:11:35] Scheduling destruction of SIP dialog '07714ae4593feb5c3e42b3a01cf4aa20 at xxx.xxx.xxx.xxx' in 32000 ms (Method: INVITE)
[Jun 3 13:11:35] set_destination: Parsing <sip:12127773456 at yyy.yyy.yyy.yyy:5060> for address/port to send to
[Jun 3 13:11:35] set_destination: set destination to yyy.yyy.yyy.yyy, port 5060
[Jun 3 13:11:35] Reliably Transmitting (NAT) to yyy.yyy.yyy.yyy:5060:
BYE sip:12127773456 at yyy.yyy.yyy.yyy:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK78ec518c;rport
Max-Forwards: 70
From: "100" <sip:100 at xxx.xxx.xxx.xxx>;tag=as643c20b1
To: <sip:12127773456 at yyy.yyy.yyy.yyy>;tag=gK029aaa8c
Call-ID: 07714ae4593feb5c3e42b3a01cf4aa20 at xxx.xxx.xxx.xxx
CSeq: 103 BYE
User-Agent: PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
[Jun 3 13:11:35] -- <SIP/100-000034d8>AGI Script call.php completed, returning -1
[Jun 3 13:11:35]
<--- SIP read from UDP:yyy.yyy.yyy.yyy:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK78ec518c;rport=5060
From: "100" <sip:100 at xxx.xxx.xxx.xxx>;tag=as643c20b1
To: <sip:12127773456 at yyy.yyy.yyy.yyy>;tag=gK029aaa8c
Call-ID: 07714ae4593feb5c3e42b3a01cf4aa20 at xxx.xxx.xxx.xxx
CSeq: 103 BYE
Content-Length: 0
Regards,
Kamlesh
> Date: Fri, 31 May 2013 08:50:38 -0500
> From: mroth at imminc.com
> To: asterisk-users at lists.digium.com
> Subject: Re: [asterisk-users] G.729 codec in pass-thru mode
>
> Kamlesh Kumar wrote:
> >
> > Yes that's correct, when I use u-law call works fine.
> >
> > In case of g729, I enabled sip debug with 'sip set debug on' and captured all
> > the sip traces and got whatever I posted in last email. There was no other
> > call on the system when I captured sip trace. Please suggest further
> > proceedings.
>
>
> Kamlesh,
>
> Please provide a SIP trace (sip set debug on) of a successful u-law call. I'm
> especially interested in the dialog between the Asterisk server and the ITSP in
> this scenario.
>
> Also include the relevant sections of sip.conf and the dialplan.
>
> Regards,
>
> Matthew Roth
> InterMedia Marketing Solutions
> Software Engineer and Systems Developer
>
> --
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