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[Freeswitch-users] Not passing G729 calls


 
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pmhshz at gmail.com
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PostPosted: Sat Oct 18, 2008 8:45 am    Post subject: [Freeswitch-users] Not passing G729 calls Reply with quote

Hi all,

I am routing G729 calls from Gateway X to Gateway Z using FREESWITCH Y,

I have enabled bypass media for G729 passthrough.

my FreeSwitch Y accepts calls from X and when it route to Z it receives "183
Session Progress." from there.

Problem comes after that, FS is cancelling the call after getting 183
session progress.

Below is the ngrep trace of whole scenario i have mentioned above:

################################################################################
#
U xx.xx.xxx.xx:63263 -> yy.yy.yy.yyy:5060
INVITE sip:NUMBER@yy.yy.yy.yyy SIP/2.0.
Via: SIP/2.0/UDP xx.xx.xxx.xx:63263;branch=z9hG4bK5c350148;rport.
From: "1001" <sip:1001@xx.xx.xxx.xx>;tag=as237d0908.
To: <sip:NUMBER@yy.yy.yy.yyy>.
Contact: <sip:1001@xx.xx.xxx.xx>.
Call-ID: 416b336c14dc309e371459124f947790@xx.xx.xxx.xx.
CSeq: 102 INVITE.
User-Agent: Asterisk PBX.
Max-Forwards: 70.
Date: Sun, 19 Oct 2008 12:46:14 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Supported: replaces.
Content-Type: application/sdp.
Content-Length: 261.
.
v=0.
o=root 6936 6936 IN IP4 xx.xx.xxx.xx.
s=session.
c=IN IP4 xx.xx.xxx.xx.
t=0 0.
m=audio 63264 RTP/AVP 18 101.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.

#
U yy.yy.yy.yyy:5060 -> xx.xx.xxx.xx:63263
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP xx.xx.xxx.xx:63263;branch=z9hG4bK5c350148;rport=63263.
From: "1001" <sip:1001@xx.xx.xxx.xx>;tag=as237d0908.
To: <sip:NUMBER@yy.yy.yy.yyy>.
Call-ID: 416b336c14dc309e371459124f947790@xx.xx.xxx.xx.
CSeq: 102 INVITE.
User-Agent: FreeSWITCH-mod_sofia/1.0.1-exported.
Content-Length: 0.
.

#
U yy.yy.yy.yyy:5080 -> zz.zz.zz.zzz:5060
INVITE sip:NUMBER@zz.zz.zz.zzz SIP/2.0.
Via: SIP/2.0/UDP yy.yy.yy.yyy:5080;rport;branch=z9hG4bKN0NFB8QrS096F.
Max-Forwards: 69.
From: "1001" <sip:1001@yy.yy.yy.yyy>;tag=NUZ4UvUv8tKvK.
To: <sip:NUMBER@zz.zz.zz.zzz>.
Call-ID: eb8148d6-17ab-122c-93b5-003048911494.
CSeq: 106046574 INVITE.
Contact: <sip:mod_sofia@yy.yy.yy.yyy:5080>.
User-Agent: FreeSWITCH-mod_sofia/1.0.1-exported.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY,
REFER, UPDATE, REGISTER, INFO.
Supported: 100rel, timer, precondition, path, replaces.
Allow-Events: talk.
Min-SE: 120.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 279.
Remote-Party-ID: "1001" <sip:1001@yy.yy.yy.yyy>;screen=yes;privacy=off.
.
v=0.
o=FreeSWITCH 1666183455704603611 2165736585687934666 IN IP4 yy.yy.yy.yyy.
s=FreeSWITCH.
c=IN IP4 yy.yy.yy.yyy.
t=0 0.
a=sendrecv.
m=audio 20094 RTP/AVP 18 101.
a=rtpmap:18 G729/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.

#
U zz.zz.zz.zzz:5060 -> yy.yy.yy.yyy:5080
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP yy.yy.yy.yyy:5080;rport;branch=z9hG4bKN0NFB8QrS096F.
From: "1001" <sip:1001@yy.yy.yy.yyy>;tag=NUZ4UvUv8tKvK.
To: <sip:NUMBER@zz.zz.zz.zzz>.
Call-ID: eb8148d6-17ab-122c-93b5-003048911494.
CSeq: 106046574 INVITE.
Content-Length: 0.
.

#
U zz.zz.zz.zzz:5060 -> yy.yy.yy.yyy:5080
SIP/2.0 183 Session Progress.
Require: 100rel.
Via: SIP/2.0/UDP yy.yy.yy.yyy:5080;rport;branch=z9hG4bKN0NFB8QrS096F.
RSeq: 1.
To: <sip:NUMBER@zz.zz.zz.zzz>;tag=3433323190-141009.
From: "1001" <sip:1001@yy.yy.yy.yyy>;tag=NUZ4UvUv8tKvK.
Call-ID: eb8148d6-17ab-122c-93b5-003048911494.
CSeq: 106046574 INVITE.
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER,
SUBSCRIBE, PRACK, UPDATE.
Contact: <sip:NUMBER@zz.zz.zz.zzz:5060>.
Call-Info:
<sip:zz.zz.zz.zzz>;method="NOTIFY;Event=telephone-event;Duration=1000".
Allow-Events: telephone-event.
Content-Type: application/sdp.
Content-Length: 143.
.
v=0.
o=diamond-msc2 6683 705 IN IP4 zz.zz.zz.zzz.
s=sip call.
c=IN IP4 213.170.194.56.
t=0 0.
m=audio 43078 RTP/AVP 101.
a=rtpmap:101 /8000.

#
U yy.yy.yy.yyy:5080 -> zz.zz.zz.zzz:5060
CANCEL sip:NUMBER@zz.zz.zz.zzz SIP/2.0.
Via: SIP/2.0/UDP yy.yy.yy.yyy:5080;rport;branch=z9hG4bKN0NFB8QrS096F.
Max-Forwards: 69.
From: "1001" <sip:1001@yy.yy.yy.yyy>;tag=NUZ4UvUv8tKvK.
To: <sip:NUMBER@zz.zz.zz.zzz>.
Call-ID: eb8148d6-17ab-122c-93b5-003048911494.
CSeq: 106046574 CANCEL.
Content-Length: 0.
.

#
U zz.zz.zz.zzz:5060 -> yy.yy.yy.yyy:5080
SIP/2.0 200 OK.
Via: SIP/2.0/UDP yy.yy.yy.yyy:5080;rport;branch=z9hG4bKN0NFB8QrS096F.
To: <sip:NUMBER@zz.zz.zz.zzz>;tag=3433323190-141009.
From: "1001" <sip:1001@yy.yy.yy.yyy>;tag=NUZ4UvUv8tKvK.
Call-ID: eb8148d6-17ab-122c-93b5-003048911494.
CSeq: 106046574 CANCEL.
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER,
SUBSCRIBE, PRACK, UPDATE.
Contact: <sip:NUMBER@zz.zz.zz.zzz:5060>.
Content-Length: 0.
.

#
U zz.zz.zz.zzz:5060 -> yy.yy.yy.yyy:5080
SIP/2.0 487 Request Terminated.
Via: SIP/2.0/UDP yy.yy.yy.yyy:5080;rport;branch=z9hG4bKN0NFB8QrS096F.
To: <sip:NUMBER@zz.zz.zz.zzz>;tag=3433323190-141009.
From: "1001" <sip:1001@yy.yy.yy.yyy>;tag=NUZ4UvUv8tKvK.
Reason: Q.850;cause=16.
Call-ID: eb8148d6-17ab-122c-93b5-003048911494.
CSeq: 106046574 INVITE.
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER,
SUBSCRIBE, PRACK, UPDATE.
Contact: <sip:NUMBER@zz.zz.zz.zzz:5060>.
Call-Info:
<sip:zz.zz.zz.zzz>;method="NOTIFY;Event=telephone-event;Duration=1000".
Content-Length: 0.
.

#
U yy.yy.yy.yyy:5080 -> zz.zz.zz.zzz:5060
ACK sip:NUMBER@zz.zz.zz.zzz SIP/2.0.
Via: SIP/2.0/UDP yy.yy.yy.yyy:5080;rport;branch=z9hG4bKN0NFB8QrS096F.
Max-Forwards: 69.
From: "1001" <sip:1001@yy.yy.yy.yyy>;tag=NUZ4UvUv8tKvK.
To: <sip:NUMBER@zz.zz.zz.zzz>;tag=3433323190-141009.
Call-ID: eb8148d6-17ab-122c-93b5-003048911494.
CSeq: 106046574 ACK.
Content-Length: 0.

#####################################################################################

Waiting for your reply
Thanks

MSP
--
View this message in context: http://www.nabble.com/Not-passing-G729-calls-tp20047045p20047045.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.


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