[asterisk-bugs] [Asterisk 0014244]: No Audio on Call Transfer (Invite not being forwarded to Provider via Asterisk)
Asterisk Bug Tracker
noreply at bugs.digium.com
Tue May 19 09:47:48 CDT 2009
A NOTE has been added to this issue.
======================================================================
https://issues.asterisk.org/view.php?id=14244
======================================================================
Reported By: mbnwa
Assigned To: file
======================================================================
Project: Asterisk
Issue ID: 14244
Category: Channels/chan_sip/Transfers
Reproducibility: always
Severity: major
Priority: normal
Status: assigned
Target Version: 1.4.26
Asterisk Version: 1.4.18
Regression: No
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
======================================================================
Date Submitted: 2009-01-14 18:13 CST
Last Modified: 2009-05-19 09:47 CDT
======================================================================
Summary: No Audio on Call Transfer (Invite not being
forwarded to Provider via Asterisk)
Description:
Notes:
Asterisk 1.4.18 & Asterisk 1.6.x effected
Running directrtpsetup=yes
OS Debian
Kernel Version 2.6.26-1-amd64
Called number 13605551212
Caller's number 13605551211
Extension to get transfer: 13605551210
Caller z.z.z.z
Asterisk Server x.x.x.x
Carrier y.y.y.y
Call Flow
13605551211 calls 13605551212 makes the transfer to 13605551210 at this
point call is direct between 13605551212 and 13605551210 but no audio
Issue:
Phone 1 makes an outbound call then transfers to another extension invite
is sent from phone 1 to asterisk, Asterisk ack's however it never sends
the invite to the carrier to update the audio path resulting in no audio
SIP Trace
INVITE sip:13605551210 at x.x.x.x SIP/2.0
Via: SIP/2.0/UDP z.z.z.z:5060;branch=z9hG4bK0b2ec42a
From: "13605551211" <sip:13605551211 at z.z.z.z>;tag=as660257ef
To: <sip:13605551210 at x.x.x.x>
Contact: <sip:13605551211 at z.z.z.z>
Call-ID: 2623f3bb4a5b79b67a7d559b69d3ac7f at z.z.z.z
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 14 Jan 2009 23:07:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 386
v=0
o=root 3032 3032 IN IP4 z.z.z.z
s=session
c=IN IP4 z.z.z.z
b=CT:384
t=0 0
m=audio 12052 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
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
m=video 16626 RTP/AVP 31 34
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000
a=sendrecv
SIP/2.0 100 Trying
Via: SIP/2.0/UDP z.z.z.z:5060;branch=z9hG4bK0b2ec42a;received=z.z.z.z
From: "13605551211" <sip:13605551211 at z.z.z.z>;tag=as660257ef
To: <sip:13605551210 at x.x.x.x>
Call-ID: 2623f3bb4a5b79b67a7d559b69d3ac7f at z.z.z.z
CSeq: 102 INVITE
Server: ASTERISK
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:13605551210 at x.x.x.x>
Content-Length: 0
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP z.z.z.z:5060;branch=z9hG4bK0b2ec42a;received=z.z.z.z
From: "13605551211" <sip:13605551211 at z.z.z.z>;tag=as660257ef
To: <sip:13605551210 at x.x.x.x>;tag=as3caea504
Call-ID: 2623f3bb4a5b79b67a7d559b69d3ac7f at z.z.z.z
CSeq: 102 INVITE
Server: ASTERISK
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:13605551210 at x.x.x.x>
Content-Type: application/sdp
Content-Length: 385
v=0
o=ASTERISK 2096545457 2096545457 IN IP4 y.y.y.y
s=ASTERISK
c=IN IP4 y.y.y.y
b=CT:384
t=0 0
m=audio 11600 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
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
m=video 10810 RTP/AVP 34
a=rtpmap:34 H263/90000
a=sendrecv
SIP/2.0 200 OK
Via: SIP/2.0/UDP z.z.z.z:5060;branch=z9hG4bK0b2ec42a;received=z.z.z.z
From: "13605551211" <sip:13605551211 at z.z.z.z>;tag=as660257ef
To: <sip:13605551210 at x.x.x.x>;tag=as3caea504
Call-ID: 2623f3bb4a5b79b67a7d559b69d3ac7f at z.z.z.z
CSeq: 102 INVITE
Server: ASTERISK
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:13605551210 at x.x.x.x>
Content-Type: application/sdp
Content-Length: 385
v=0
o=ASTERISK 2096545457 2096545458 IN IP4 y.y.y.y
s=ASTERISK
c=IN IP4 y.y.y.y
b=CT:384
t=0 0
m=audio 11600 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
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
m=video 10810 RTP/AVP 34
a=rtpmap:34 H263/90000
a=sendrecv
ACK sip:13605551210 at x.x.x.x SIP/2.0
Via: SIP/2.0/UDP z.z.z.z:5060;branch=z9hG4bK56e7ac1e
From: "13605551211" <sip:13605551211 at z.z.z.z>;tag=as660257ef
To: <sip:13605551210 at x.x.x.x>;tag=as3caea504
Contact: <sip:13605551211 at z.z.z.z>
Call-ID: 2623f3bb4a5b79b67a7d559b69d3ac7f at z.z.z.z
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
INVITE sip:13605551211 at z.z.z.z SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK00000000;rport
Max-Forwards: 70
From: <sip:13605551210 at x.x.x.x>;tag=as3caea504
To: "13605551211" <sip:13605551211 at z.z.z.z>;tag=as660257ef
Contact: <sip:13605551210 at x.x.x.x>
Call-ID: 2623f3bb4a5b79b67a7d559b69d3ac7f at z.z.z.z
CSeq: 102 INVITE
User-Agent: ASTERISK
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 385
v=0
o=ASTERISK 2096545457 2096545459 IN IP4 y.y.y.y
s=ASTERISK
c=IN IP4 y.y.y.y
b=CT:384
t=0 0
m=audio 11600 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
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
m=video 16961 RTP/AVP 34
a=rtpmap:34 H263/90000
a=sendrecv
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
x.x.x.x:5060;branch=z9hG4bK00000000;rport;received=x.x.x.x
From: <sip:13605551210 at x.x.x.x>;tag=as3caea504
To: "13605551211" <sip:13605551211 at z.z.z.z>;tag=as660257ef
Call-ID: 2623f3bb4a5b79b67a7d559b69d3ac7f at z.z.z.z
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:13605551211 at z.z.z.z>
Content-Length: 0
SIP/2.0 200 OK
Via: SIP/2.0/UDP
x.x.x.x:5060;branch=z9hG4bK00000000;rport;received=x.x.x.x
From: <sip:13605551210 at x.x.x.x>;tag=as3caea504
To: "13605551211" <sip:13605551211 at z.z.z.z>;tag=as660257ef
Call-ID: 2623f3bb4a5b79b67a7d559b69d3ac7f at z.z.z.z
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:13605551211 at z.z.z.z>
Content-Type: application/sdp
Content-Length: 359
v=0
o=root 3032 3033 IN IP4 z.z.z.z
s=session
c=IN IP4 z.z.z.z
b=CT:384
t=0 0
m=audio 12052 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
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
m=video 16626 RTP/AVP 34
a=rtpmap:34 H263/90000
a=sendrecv
ACK sip:13605551211 at z.z.z.z SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK1cc19329;rport
Max-Forwards: 70
From: <sip:13605551210 at x.x.x.x>;tag=as3caea504
To: "13605551211" <sip:13605551211 at z.z.z.z>;tag=as660257ef
Contact: <sip:13605551210 at x.x.x.x>
Call-ID: 2623f3bb4a5b79b67a7d559b69d3ac7f at z.z.z.z
CSeq: 102 ACK
User-Agent: ASTERISK
Content-Length: 0
INVITE sip:13605551212 at x.x.x.x SIP/2.0
Via: SIP/2.0/UDP z.z.z.z:5060;branch=z9hG4bK7b098e16
From: "13605551211" <sip:13605551211 at z.z.z.z>;tag=as3d7e03cf
To: <sip:13605551212 at x.x.x.x>
Contact: <sip:13605551211 at z.z.z.z>
Call-ID: 55a3b4f81ba6febf010fc5f44862be2c at z.z.z.z
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 14 Jan 2009 23:08:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 386
v=0
o=root 3032 3032 IN IP4 z.z.z.z
s=session
c=IN IP4 z.z.z.z
b=CT:384
t=0 0
m=audio 10568 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
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
m=video 16642 RTP/AVP 31 34
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000
a=sendrecv
SIP/2.0 100 Trying
Via: SIP/2.0/UDP z.z.z.z:5060;branch=z9hG4bK7b098e16;received=z.z.z.z
From: "13605551211" <sip:13605551211 at z.z.z.z>;tag=as3d7e03cf
To: <sip:13605551212 at x.x.x.x>
Call-ID: 55a3b4f81ba6febf010fc5f44862be2c at z.z.z.z
CSeq: 102 INVITE
Server: ASTERISK
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:13605551212 at x.x.x.x>
Content-Length: 0
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP z.z.z.z:5060;branch=z9hG4bK7b098e16;received=z.z.z.z
From: "13605551211" <sip:13605551211 at z.z.z.z>;tag=as3d7e03cf
To: <sip:13605551212 at x.x.x.x>;tag=as10598c8a
Call-ID: 55a3b4f81ba6febf010fc5f44862be2c at z.z.z.z
CSeq: 102 INVITE
Server: ASTERISK
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:13605551212 at x.x.x.x>
Content-Type: application/sdp
Content-Length: 385
v=0
o=ASTERISK 1597817427 1597817427 IN IP4 y.y.y.y
s=ASTERISK
c=IN IP4 y.y.y.y
b=CT:384
t=0 0
m=audio 27614 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
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
m=video 34128 RTP/AVP 34
a=rtpmap:34 H263/90000
a=sendrecv
INVITE sip:13605551210 at x.x.x.x SIP/2.0
Via: SIP/2.0/UDP z.z.z.z:5060;branch=z9hG4bK4b46cd67
From: "13605551211" <sip:13605551211 at z.z.z.z>;tag=as660257ef
To: <sip:13605551210 at x.x.x.x>;tag=as3caea504
Contact: <sip:13605551211 at z.z.z.z>
Call-ID: 2623f3bb4a5b79b67a7d559b69d3ac7f at z.z.z.z
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 359
v=0
o=root 3032 3034 IN IP4 y.y.y.y
s=session
c=IN IP4 y.y.y.y
b=CT:384
t=0 0
m=audio 27614 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
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
m=video 34128 RTP/AVP 34
a=rtpmap:34 H263/90000
a=sendrecv
SIP/2.0 100 Trying
Via: SIP/2.0/UDP z.z.z.z:5060;branch=z9hG4bK4b46cd67;received=z.z.z.z
From: "13605551211" <sip:13605551211 at z.z.z.z>;tag=as660257ef
To: <sip:13605551210 at x.x.x.x>;tag=as3caea504
Call-ID: 2623f3bb4a5b79b67a7d559b69d3ac7f at z.z.z.z
CSeq: 103 INVITE
Server: ASTERISK
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:13605551210 at x.x.x.x>
Content-Length: 0
SIP/2.0 200 OK
Via: SIP/2.0/UDP z.z.z.z:5060;branch=z9hG4bK4b46cd67;received=z.z.z.z
From: "13605551211" <sip:13605551211 at z.z.z.z>;tag=as660257ef
To: <sip:13605551210 at x.x.x.x>;tag=as3caea504
Call-ID: 2623f3bb4a5b79b67a7d559b69d3ac7f at z.z.z.z
CSeq: 103 INVITE
Server: ASTERISK
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:13605551210 at x.x.x.x>
Content-Type: application/sdp
Content-Length: 385
v=0
o=ASTERISK 2096545457 2096545460 IN IP4 y.y.y.y
s=ASTERISK
c=IN IP4 y.y.y.y
b=CT:384
t=0 0
m=audio 11600 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
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
m=video 16961 RTP/AVP 34
a=rtpmap:34 H263/90000
a=sendrecv
ACK sip:13605551210 at x.x.x.x SIP/2.0
Via: SIP/2.0/UDP z.z.z.z:5060;branch=z9hG4bK21611593
From: "13605551211" <sip:13605551211 at z.z.z.z>;tag=as660257ef
To: <sip:13605551210 at x.x.x.x>;tag=as3caea504
Contact: <sip:13605551211 at z.z.z.z>
Call-ID: 2623f3bb4a5b79b67a7d559b69d3ac7f at z.z.z.z
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
SIP/2.0 200 OK
Via: SIP/2.0/UDP z.z.z.z:5060;branch=z9hG4bK7b098e16;received=z.z.z.z
From: "13605551211" <sip:13605551211 at z.z.z.z>;tag=as3d7e03cf
To: <sip:13605551212 at x.x.x.x>;tag=as10598c8a
Call-ID: 55a3b4f81ba6febf010fc5f44862be2c at z.z.z.z
CSeq: 102 INVITE
Server: ASTERISK
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:13605551212 at x.x.x.x>
Content-Type: application/sdp
Content-Length: 385
v=0
o=ASTERISK 1597817427 1597817428 IN IP4 y.y.y.y
s=ASTERISK
c=IN IP4 y.y.y.y
b=CT:384
t=0 0
m=audio 27614 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
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
m=video 34128 RTP/AVP 34
a=rtpmap:34 H263/90000
a=sendrecv
ACK sip:13605551212 at x.x.x.x SIP/2.0
Via: SIP/2.0/UDP z.z.z.z:5060;branch=z9hG4bK25117140
From: "13605551211" <sip:13605551211 at z.z.z.z>;tag=as3d7e03cf
To: <sip:13605551212 at x.x.x.x>;tag=as10598c8a
Contact: <sip:13605551211 at z.z.z.z>
Call-ID: 55a3b4f81ba6febf010fc5f44862be2c at z.z.z.z
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
INVITE sip:13605551212 at x.x.x.x SIP/2.0
Via: SIP/2.0/UDP z.z.z.z:5060;branch=z9hG4bK3d7a16a5
From: "13605551211" <sip:13605551211 at z.z.z.z>;tag=as3d7e03cf
To: <sip:13605551212 at x.x.x.x>;tag=as10598c8a
Contact: <sip:13605551211 at z.z.z.z>
Call-ID: 55a3b4f81ba6febf010fc5f44862be2c at z.z.z.z
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 359
v=0
o=root 3032 3033 IN IP4 y.y.y.y
s=session
c=IN IP4 y.y.y.y
b=CT:384
t=0 0
m=audio 11600 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
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
m=video 16961 RTP/AVP 34
a=rtpmap:34 H263/90000
a=sendrecv
SIP/2.0 100 Trying
Via: SIP/2.0/UDP z.z.z.z:5060;branch=z9hG4bK3d7a16a5;received=z.z.z.z
From: "13605551211" <sip:13605551211 at z.z.z.z>;tag=as3d7e03cf
To: <sip:13605551212 at x.x.x.x>;tag=as10598c8a
Call-ID: 55a3b4f81ba6febf010fc5f44862be2c at z.z.z.z
CSeq: 103 INVITE
Server: ASTERISK
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:13605551212 at x.x.x.x>
Content-Length: 0
SIP/2.0 200 OK
Via: SIP/2.0/UDP z.z.z.z:5060;branch=z9hG4bK3d7a16a5;received=z.z.z.z
From: "13605551211" <sip:13605551211 at z.z.z.z>;tag=as3d7e03cf
To: <sip:13605551212 at x.x.x.x>;tag=as10598c8a
Call-ID: 55a3b4f81ba6febf010fc5f44862be2c at z.z.z.z
CSeq: 103 INVITE
Server: ASTERISK
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:13605551212 at x.x.x.x>
Content-Type: application/sdp
Content-Length: 385
v=0
o=ASTERISK 1597817427 1597817429 IN IP4 y.y.y.y
s=ASTERISK
c=IN IP4 y.y.y.y
b=CT:384
t=0 0
m=audio 27614 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
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
m=video 34128 RTP/AVP 34
a=rtpmap:34 H263/90000
a=sendrecv
ACK sip:13605551212 at x.x.x.x SIP/2.0
Via: SIP/2.0/UDP z.z.z.z:5060;branch=z9hG4bK6a03a563
From: "13605551211" <sip:13605551211 at z.z.z.z>;tag=as3d7e03cf
To: <sip:13605551212 at x.x.x.x>;tag=as10598c8a
Contact: <sip:13605551211 at z.z.z.z>
Call-ID: 55a3b4f81ba6febf010fc5f44862be2c at z.z.z.z
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
INVITE sip:13605551211 at z.z.z.z SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK00000000;rport
Max-Forwards: 70
From: <sip:13605551212 at x.x.x.x>;tag=as10598c8a
To: "13605551211" <sip:13605551211 at z.z.z.z>;tag=as3d7e03cf
Contact: <sip:13605551212 at x.x.x.x>
Call-ID: 55a3b4f81ba6febf010fc5f44862be2c at z.z.z.z
CSeq: 102 INVITE
User-Agent: ASTERISK
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 385
v=0
o=ASTERISK 1597817427 1597817430 IN IP4 y.y.y.y
s=ASTERISK
c=IN IP4 y.y.y.y
b=CT:384
t=0 0
m=audio 27614 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
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
m=video 26304 RTP/AVP 34
a=rtpmap:34 H263/90000
a=sendrecv
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
x.x.x.x:5060;branch=z9hG4bK00000000;rport;received=x.x.x.x
From: <sip:13605551212 at x.x.x.x>;tag=as10598c8a
To: "13605551211" <sip:13605551211 at z.z.z.z>;tag=as3d7e03cf
Call-ID: 55a3b4f81ba6febf010fc5f44862be2c at z.z.z.z
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:13605551211 at z.z.z.z>
Content-Length: 0
SIP/2.0 200 OK
Via: SIP/2.0/UDP
x.x.x.x:5060;branch=z9hG4bK00000000;rport;received=x.x.x.x
From: <sip:13605551212 at x.x.x.x>;tag=as10598c8a
To: "13605551211" <sip:13605551211 at z.z.z.z>;tag=as3d7e03cf
Call-ID: 55a3b4f81ba6febf010fc5f44862be2c at z.z.z.z
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:13605551211 at z.z.z.z>
Content-Type: application/sdp
Content-Length: 359
v=0
o=root 3032 3034 IN IP4 y.y.y.y
s=session
c=IN IP4 y.y.y.y
b=CT:384
t=0 0
m=audio 11600 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
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
m=video 16961 RTP/AVP 34
a=rtpmap:34 H263/90000
a=sendrecv
ACK sip:13605551211 at z.z.z.z SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK281ef63b;rport
Max-Forwards: 70
From: <sip:13605551212 at x.x.x.x>;tag=as10598c8a
To: "13605551211" <sip:13605551211 at z.z.z.z>;tag=as3d7e03cf
Contact: <sip:13605551212 at x.x.x.x>
Call-ID: 55a3b4f81ba6febf010fc5f44862be2c at z.z.z.z
CSeq: 102 ACK
User-Agent: ASTERISK
Content-Length: 0
INVITE sip:13605551210 at x.x.x.x SIP/2.0
Via: SIP/2.0/UDP z.z.z.z:5060;branch=z9hG4bK53629ddb
From: "13605551211" <sip:13605551211 at z.z.z.z>;tag=as660257ef
To: <sip:13605551210 at x.x.x.x>;tag=as3caea504
Contact: <sip:13605551211 at z.z.z.z>
Call-ID: 2623f3bb4a5b79b67a7d559b69d3ac7f at z.z.z.z
CSeq: 104 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 359
v=0
o=root 3032 3035 IN IP4 y.y.y.y
s=session
c=IN IP4 y.y.y.y
b=CT:384
t=0 0
m=audio 27614 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
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
m=video 26304 RTP/AVP 34
a=rtpmap:34 H263/90000
a=sendrecv
SIP/2.0 100 Trying
Via: SIP/2.0/UDP z.z.z.z:5060;branch=z9hG4bK53629ddb;received=z.z.z.z
From: "13605551211" <sip:13605551211 at z.z.z.z>;tag=as660257ef
To: <sip:13605551210 at x.x.x.x>;tag=as3caea504
Call-ID: 2623f3bb4a5b79b67a7d559b69d3ac7f at z.z.z.z
CSeq: 104 INVITE
Server: ASTERISK
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:13605551210 at x.x.x.x>
Content-Length: 0
SIP/2.0 200 OK
Via: SIP/2.0/UDP z.z.z.z:5060;branch=z9hG4bK53629ddb;received=z.z.z.z
From: "13605551211" <sip:13605551211 at z.z.z.z>;tag=as660257ef
To: <sip:13605551210 at x.x.x.x>;tag=as3caea504
Call-ID: 2623f3bb4a5b79b67a7d559b69d3ac7f at z.z.z.z
CSeq: 104 INVITE
Server: ASTERISK
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:13605551210 at x.x.x.x>
Content-Type: application/sdp
Content-Length: 385
v=0
o=ASTERISK 2096545457 2096545461 IN IP4 y.y.y.y
s=ASTERISK
c=IN IP4 y.y.y.y
b=CT:384
t=0 0
m=audio 11600 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
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
m=video 16961 RTP/AVP 34
a=rtpmap:34 H263/90000
a=sendrecv
ACK sip:13605551210 at x.x.x.x SIP/2.0
Via: SIP/2.0/UDP z.z.z.z:5060;branch=z9hG4bK01f3ba2d
From: "13605551211" <sip:13605551211 at z.z.z.z>;tag=as660257ef
To: <sip:13605551210 at x.x.x.x>;tag=as3caea504
Contact: <sip:13605551211 at z.z.z.z>
Call-ID: 2623f3bb4a5b79b67a7d559b69d3ac7f at z.z.z.z
CSeq: 104 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
INVITE sip:13605551211 at z.z.z.z SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK00000000;rport
Max-Forwards: 70
From: <sip:13605551212 at x.x.x.x>;tag=as10598c8a
To: "13605551211" <sip:13605551211 at z.z.z.z>;tag=as3d7e03cf
Contact: <sip:13605551212 at x.x.x.x>
Call-ID: 55a3b4f81ba6febf010fc5f44862be2c at z.z.z.z
CSeq: 103 INVITE
User-Agent: ASTERISK
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 385
v=0
o=ASTERISK 1597817427 1597817431 IN IP4 y.y.y.y
s=ASTERISK
c=IN IP4 y.y.y.y
b=CT:384
t=0 0
m=audio 27614 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
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
m=video 16961 RTP/AVP 34
a=rtpmap:34 H263/90000
a=sendrecv
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
x.x.x.x:5060;branch=z9hG4bK00000000;rport;received=x.x.x.x
From: <sip:13605551212 at x.x.x.x>;tag=as10598c8a
To: "13605551211" <sip:13605551211 at z.z.z.z>;tag=as3d7e03cf
Call-ID: 55a3b4f81ba6febf010fc5f44862be2c at z.z.z.z
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:13605551211 at z.z.z.z>
Content-Length: 0
SIP/2.0 200 OK
Via: SIP/2.0/UDP
x.x.x.x:5060;branch=z9hG4bK00000000;rport;received=x.x.x.x
From: <sip:13605551212 at x.x.x.x>;tag=as10598c8a
To: "13605551211" <sip:13605551211 at z.z.z.z>;tag=as3d7e03cf
Call-ID: 55a3b4f81ba6febf010fc5f44862be2c at z.z.z.z
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:13605551211 at z.z.z.z>
Content-Type: application/sdp
Content-Length: 359
v=0
o=root 3032 3035 IN IP4 y.y.y.y
s=session
c=IN IP4 y.y.y.y
b=CT:384
t=0 0
m=audio 11600 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
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
m=video 16961 RTP/AVP 34
a=rtpmap:34 H263/90000
a=sendrecv
ACK sip:13605551211 at z.z.z.z SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK2158005c;rport
Max-Forwards: 70
From: <sip:13605551212 at x.x.x.x>;tag=as10598c8a
To: "13605551211" <sip:13605551211 at z.z.z.z>;tag=as3d7e03cf
Contact: <sip:13605551212 at x.x.x.x>
Call-ID: 55a3b4f81ba6febf010fc5f44862be2c at z.z.z.z
CSeq: 103 ACK
User-Agent: ASTERISK
Content-Length: 0
INVITE sip:13605551210 at x.x.x.x SIP/2.0
Via: SIP/2.0/UDP z.z.z.z:5060;branch=z9hG4bK4a36f34b
From: "13605551211" <sip:13605551211 at z.z.z.z>;tag=as660257ef
To: <sip:13605551210 at x.x.x.x>;tag=as3caea504
Contact: <sip:13605551211 at z.z.z.z>
Call-ID: 2623f3bb4a5b79b67a7d559b69d3ac7f at z.z.z.z
CSeq: 105 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 359
v=0
o=root 3032 3036 IN IP4 y.y.y.y
s=session
c=IN IP4 y.y.y.y
b=CT:384
t=0 0
m=audio 27614 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
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
m=video 16961 RTP/AVP 34
a=rtpmap:34 H263/90000
a=sendrecv
SIP/2.0 100 Trying
Via: SIP/2.0/UDP z.z.z.z:5060;branch=z9hG4bK4a36f34b;received=z.z.z.z
From: "13605551211" <sip:13605551211 at z.z.z.z>;tag=as660257ef
To: <sip:13605551210 at x.x.x.x>;tag=as3caea504
Call-ID: 2623f3bb4a5b79b67a7d559b69d3ac7f at z.z.z.z
CSeq: 105 INVITE
Server: ASTERISK
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:13605551210 at x.x.x.x>
Content-Length: 0
SIP/2.0 200 OK
Via: SIP/2.0/UDP z.z.z.z:5060;branch=z9hG4bK4a36f34b;received=z.z.z.z
From: "13605551211" <sip:13605551211 at z.z.z.z>;tag=as660257ef
To: <sip:13605551210 at x.x.x.x>;tag=as3caea504
Call-ID: 2623f3bb4a5b79b67a7d559b69d3ac7f at z.z.z.z
CSeq: 105 INVITE
Server: ASTERISK
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:13605551210 at x.x.x.x>
Content-Type: application/sdp
Content-Length: 385
v=0
o=ASTERISK 2096545457 2096545462 IN IP4 y.y.y.y
s=ASTERISK
c=IN IP4 y.y.y.y
b=CT:384
t=0 0
m=audio 11600 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
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
m=video 16961 RTP/AVP 34
a=rtpmap:34 H263/90000
a=sendrecv
ACK sip:13605551210 at x.x.x.x SIP/2.0
Via: SIP/2.0/UDP z.z.z.z:5060;branch=z9hG4bK14f0c3cb
From: "13605551211" <sip:13605551211 at z.z.z.z>;tag=as660257ef
To: <sip:13605551210 at x.x.x.x>;tag=as3caea504
Contact: <sip:13605551211 at z.z.z.z>
Call-ID: 2623f3bb4a5b79b67a7d559b69d3ac7f at z.z.z.z
CSeq: 105 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
INVITE sip:13605551211 at z.z.z.z SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK00000000;rport
Max-Forwards: 70
From: <sip:13605551210 at x.x.x.x>;tag=as3caea504
To: "13605551211" <sip:13605551211 at z.z.z.z>;tag=as660257ef
Contact: <sip:13605551210 at x.x.x.x>
Call-ID: 2623f3bb4a5b79b67a7d559b69d3ac7f at z.z.z.z
CSeq: 103 INVITE
User-Agent: ASTERISK
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 385
v=0
o=ASTERISK 2096545457 2096545463 IN IP4 x.x.x.x
s=ASTERISK
c=IN IP4 x.x.x.x
b=CT:384
t=0 0
m=audio 42782 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
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
m=video 10810 RTP/AVP 34
a=rtpmap:34 H263/90000
a=sendrecv
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
x.x.x.x:5060;branch=z9hG4bK00000000;rport;received=x.x.x.x
From: <sip:13605551210 at x.x.x.x>;tag=as3caea504
To: "13605551211" <sip:13605551211 at z.z.z.z>;tag=as660257ef
Call-ID: 2623f3bb4a5b79b67a7d559b69d3ac7f at z.z.z.z
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:13605551211 at z.z.z.z>
Content-Length: 0
SIP/2.0 200 OK
Via: SIP/2.0/UDP
x.x.x.x:5060;branch=z9hG4bK00000000;rport;received=x.x.x.x
From: <sip:13605551210 at x.x.x.x>;tag=as3caea504
To: "13605551211" <sip:13605551211 at z.z.z.z>;tag=as660257ef
Call-ID: 2623f3bb4a5b79b67a7d559b69d3ac7f at z.z.z.z
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:13605551211 at z.z.z.z>
Content-Type: application/sdp
Content-Length: 359
v=0
o=root 3032 3037 IN IP4 y.y.y.y
s=session
c=IN IP4 y.y.y.y
b=CT:384
t=0 0
m=audio 27614 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
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
m=video 16961 RTP/AVP 34
a=rtpmap:34 H263/90000
a=sendrecv
SIP/2.0 503 Server error
Via: SIP/2.0/UDP
x.x.x.x:5060;branch=z9hG4bK00000000;received=x.x.x.x;received=z.z.z.z;rport=5060
From: <sip:13605551210 at x.x.x.x>;tag=as3caea504
To: "13605551211" <sip:13605551211 at z.z.z.z>;tag=as660257ef
Call-ID: 2623f3bb4a5b79b67a7d559b69d3ac7f at z.z.z.z
CSeq: 103 INVITE
Server: ASTERISK
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:13605551210 at x.x.x.x>
Content-Length: 0
ACK sip:13605551211 at z.z.z.z SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK57ff9c9a;rport
Max-Forwards: 70
From: <sip:13605551210 at x.x.x.x>;tag=as3caea504
To: "13605551211" <sip:13605551211 at z.z.z.z>;tag=as660257ef
Contact: <sip:13605551210 at x.x.x.x>
Call-ID: 2623f3bb4a5b79b67a7d559b69d3ac7f at z.z.z.z
CSeq: 103 ACK
User-Agent: ASTERISK
Content-Length: 0
BYE sip:13605551211 at z.z.z.z SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK3b46b0a6;rport
Max-Forwards: 70
From: <sip:13605551210 at x.x.x.x>;tag=as3caea504
To: "13605551211" <sip:13605551211 at z.z.z.z>;tag=as660257ef
Call-ID: 2623f3bb4a5b79b67a7d559b69d3ac7f at z.z.z.z
CSeq: 104 BYE
User-Agent: ASTERISK
Content-Length: 0
SIP/2.0 200 OK
Via: SIP/2.0/UDP
x.x.x.x:5060;branch=z9hG4bK3b46b0a6;rport;received=x.x.x.x
From: <sip:13605551210 at x.x.x.x>;tag=as3caea504
To: "13605551211" <sip:13605551211 at z.z.z.z>;tag=as660257ef
Call-ID: 2623f3bb4a5b79b67a7d559b69d3ac7f at z.z.z.z
CSeq: 104 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:13605551211 at z.z.z.z>
Content-Length: 0
INVITE sip:13605551212 at x.x.x.x SIP/2.0
Via: SIP/2.0/UDP z.z.z.z:5060;branch=z9hG4bK31287ca4
From: "13605551211" <sip:13605551211 at z.z.z.z>;tag=as3d7e03cf
To: <sip:13605551212 at x.x.x.x>;tag=as10598c8a
Contact: <sip:13605551211 at z.z.z.z>
Call-ID: 55a3b4f81ba6febf010fc5f44862be2c at z.z.z.z
CSeq: 104 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 359
v=0
o=root 3032 3036 IN IP4 z.z.z.z
s=session
c=IN IP4 z.z.z.z
b=CT:384
t=0 0
m=audio 10568 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
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
m=video 16642 RTP/AVP 34
a=rtpmap:34 H263/90000
a=sendrecv
SIP/2.0 100 Trying
Via: SIP/2.0/UDP z.z.z.z:5060;branch=z9hG4bK31287ca4;received=z.z.z.z
From: "13605551211" <sip:13605551211 at z.z.z.z>;tag=as3d7e03cf
To: <sip:13605551212 at x.x.x.x>;tag=as10598c8a
Call-ID: 55a3b4f81ba6febf010fc5f44862be2c at z.z.z.z
CSeq: 104 INVITE
Server: ASTERISK
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:13605551212 at x.x.x.x>
Content-Length: 0
SIP/2.0 200 OK
Via: SIP/2.0/UDP z.z.z.z:5060;branch=z9hG4bK31287ca4;received=z.z.z.z
From: "13605551211" <sip:13605551211 at z.z.z.z>;tag=as3d7e03cf
To: <sip:13605551212 at x.x.x.x>;tag=as10598c8a
Call-ID: 55a3b4f81ba6febf010fc5f44862be2c at z.z.z.z
CSeq: 104 INVITE
Server: ASTERISK
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:13605551212 at x.x.x.x>
Content-Type: application/sdp
Content-Length: 385
v=0
o=ASTERISK 1597817427 1597817432 IN IP4 y.y.y.y
s=ASTERISK
c=IN IP4 y.y.y.y
b=CT:384
t=0 0
m=audio 27614 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
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
m=video 16961 RTP/AVP 34
a=rtpmap:34 H263/90000
a=sendrecv
ACK sip:13605551212 at x.x.x.x SIP/2.0
Via: SIP/2.0/UDP z.z.z.z:5060;branch=z9hG4bK0e6dba92
From: "13605551211" <sip:13605551211 at z.z.z.z>;tag=as3d7e03cf
To: <sip:13605551212 at x.x.x.x>;tag=as10598c8a
Contact: <sip:13605551211 at z.z.z.z>
Call-ID: 55a3b4f81ba6febf010fc5f44862be2c at z.z.z.z
CSeq: 104 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
BYE sip:13605551212 at x.x.x.x SIP/2.0
Via: SIP/2.0/UDP z.z.z.z:5060;branch=z9hG4bK4ea0c262
From: "13605551211" <sip:13605551211 at z.z.z.z>;tag=as3d7e03cf
To: <sip:13605551212 at x.x.x.x>;tag=as10598c8a
Call-ID: 55a3b4f81ba6febf010fc5f44862be2c at z.z.z.z
CSeq: 105 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
SIP/2.0 200 OK
Via: SIP/2.0/UDP z.z.z.z:5060;branch=z9hG4bK4ea0c262;received=z.z.z.z
From: "13605551211" <sip:13605551211 at z.z.z.z>;tag=as3d7e03cf
To: <sip:13605551212 at x.x.x.x>;tag=as10598c8a
Call-ID: 55a3b4f81ba6febf010fc5f44862be2c at z.z.z.z
CSeq: 105 BYE
Server: ASTERISK
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:13605551212 at x.x.x.x>
Content-Length: 0
======================================================================
----------------------------------------------------------------------
(0105007) svnbot (reporter) - 2009-05-19 09:47
https://issues.asterisk.org/view.php?id=14244#c105007
----------------------------------------------------------------------
Repository: asterisk
Revision: 195451
_U branches/1.6.1/
U branches/1.6.1/channels/chan_sip.c
------------------------------------------------------------------------
r195451 | file | 2009-05-19 09:47:47 -0500 (Tue, 19 May 2009) | 21 lines
Merged revisions 195449 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
................
r195449 | file | 2009-05-19 11:43:54 -0300 (Tue, 19 May 2009) | 14 lines
Merged revisions 195448 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r195448 | file | 2009-05-19 11:41:45 -0300 (Tue, 19 May 2009) | 7
lines
Fix a bug where direct RTP setup would partially occur even when
disabled if the calling channel was answered.
(issue https://issues.asterisk.org/view.php?id=13545)
Reported by: davidw
(issue https://issues.asterisk.org/view.php?id=14244)
Reported by: mbnwa
........
................
------------------------------------------------------------------------
http://svn.digium.com/view/asterisk?view=rev&revision=195451
Issue History
Date Modified Username Field Change
======================================================================
2009-05-19 09:47 svnbot Checkin
2009-05-19 09:47 svnbot Note Added: 0105007
======================================================================
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