Support Requests item #3429000, was opened at 2011-10-26 23:58
Message generated for change (Tracker Item Submitted) made by nobody
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Category: core
Group: ver devel
Status: Open
Priority: 5
Private: No
Submitted By: Nobody/Anonymous (nobody)
Assigned to: Nobody/Anonymous (nobody)
Summary: Asterisk 1.8 + Kamailio 1.5
Initial Comment:
Hi,
We are having issues where the "OK" or "ACK" is that is coming from the phone is not relayed by OpenSER to Asterisk.
Below is the sip trace... I am also attaching a tcpdump. Please help what we can do.
Received from udp:10.1.10.80:5060 at 26/10/2011 10:22:41:476 (490 bytes):
SIP/2.0 481 Call/Transaction Does Not Exist
Via: SIP/2.0/UDP 10.30.0.64:5060;received=10.30.0.64;branch=z9hG4bK-wowp1kmdy4rl;rport=5060
From: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=6wkdms1r20
To: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as0b87218f
Call-ID: 3c26755bf15c-9iq08xqqblo6
CSeq: 4 INVITE
Server: Asterisk PBX 1.8.7.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
--------------------------------------------------------------------------------
Sent to udp:10.1.10.80:5060 at 26/10/2011 10:22:41:481 (387 bytes):
ACK sip:vm9513261429@10.1.10.83:5060 SIP/2.0
v: SIP/2.0/UDP 10.30.0.64:5060;branch=z9hG4bK-wowp1kmdy4rl;rport
Route: <sip:10.1.10.80;lr=on>
f: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=6wkdms1r20
t: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as0b87218f
i: 3c26755bf15c-9iq08xqqblo6
CSeq: 4 ACK
Max-Forwards: 70
m: <sip:91421@10.30.0.64:5060>;reg-id=1
l: 0
--------------------------------------------------------------------------------
Received from udp:10.1.10.80:5060 at 26/10/2011 10:22:42:130 (868 bytes):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.30.0.64:5060;received=10.30.0.64;branch=z9hG4bK-5evtiw6dm0po;rport=5060
Record-Route: <sip:10.1.10.80;lr=on>
From: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=qi3i8ze6z8
To: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as3f8c0f96
Call-ID: 3c2676547a8d-2t5yi6jok1sv
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.7.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:9513261429@10.1.10.83:5060>
Content-Type: application/sdp
Content-Length: 256
v=0
o=root 1355451627 1355451627 IN IP4 10.1.10.83
s=Asterisk PBX 1.8.7.1
c=IN IP4 10.1.10.83
t=0 0
m=audio 16094 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
--------------------------------------------------------------------------------
Sent to udp:10.1.10.80:5060 at 26/10/2011 10:22:42:132 (385 bytes):
ACK sip:9513261429@10.1.10.83:5060 SIP/2.0
v: SIP/2.0/UDP 10.30.0.64:5060;branch=z9hG4bK-wszafb7cbzpw;rport
Route: <sip:10.1.10.80;lr=on>
f: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=qi3i8ze6z8
t: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as3f8c0f96
i: 3c2676547a8d-2t5yi6jok1sv
CSeq: 2 ACK
Max-Forwards: 70
m: <sip:91421@10.30.0.64:5060>;reg-id=1
l: 0
--------------------------------------------------------------------------------
Received from udp:10.1.10.80:5060 at 26/10/2011 10:22:42:232 (503 bytes):
BYE sip:91421@10.30.0.64:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.10.80;branch=z9hG4bKe723.bf70c1f4.0
Via: SIP/2.0/UDP 10.1.10.83:5060;branch=z9hG4bK69f53cf1
Max-Forwards: 69
From: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as3f8c0f96
To: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=qi3i8ze6z8
Call-ID: 3c2676547a8d-2t5yi6jok1sv
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.8.7.1
X-Asterisk-HangupCause: Protocol error, unspecified
X-Asterisk-HangupCauseCode: 111
Content-Length: 0
Regards,
Rowell
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Module: sip-router
Branch: master
Commit: db308939b551a920b31bdebd76c28b5b104db68c
URL: http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=db30893…
Author: Jason Penton <jason.penton(a)gmail.com>
Committer: Jason Penton <jason.penton(a)gmail.com>
Date: Wed Oct 26 14:32:00 2011 +0200
Dialog: Fixed lurking html tag in XML doc
---
modules_k/dialog/doc/dialog_admin.xml | 2 +-
1 files changed, 1 insertions(+), 1 deletions(-)
diff --git a/modules_k/dialog/doc/dialog_admin.xml b/modules_k/dialog/doc/dialog_admin.xml
index 9bcbea1..86a5ad2 100644
--- a/modules_k/dialog/doc/dialog_admin.xml
+++ b/modules_k/dialog/doc/dialog_admin.xml
@@ -1439,7 +1439,7 @@ dlg_refer("caller", "sip:annoucement@kamailio.org");
dialog created is further processed statefully. Specifically, if a
stateless response is sent out after dlg_manage() is called, the
dialog cannot be handled properly. So make sure that a transaction
- exists or create it explicitly using the tm module.<br>This is a
+ exists or create it explicitly using the tm module. This is a
shortcoming of the current implementation that may be resolved in a
future version hopefully.
</para>