#6 Asterisk 1.8 + Kamailio 1.5

ver devel
open
nobody
core (2)
5
2011-10-26
2011-10-26
Anonymous
No

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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