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Disconnected SIP Call

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Guilherme1000

Guilherme1000

Technical User
Apr 17, 2012
270
BR
Hello guys

I have a strange issue.
we have here an ipoffice500v2 7.0(36) 2 combo cards phone2 PRI phone 30.

we have a SIP trunk linked to a IP Cellphone Interface.

The IP call is made correctly but some calls are automatically disconnected every 30 seconds.

3 calls, 1 got disconnected.
seems like the pabx answers with an "ok" instead of "ack" every time the call is disconnected.

this is the system monitor when the call is disconnected.

please help!

11866706mS Sip: License, Valid 1, Available 16, Consumed 0
11866710mS Sip: 17.1129.0 43 SIPTrunk Endpoint(f541ec94) received CMSetup
11866711mS Sip: 17.1129.0 43 SIPTrunk Endpoint(f541db30) SetLocalRTPAddress to 192.168.1.55:49152 (index 0)
11866712mS SIP Call Tx: 17
INVITE sip:87521512@192.168.1.59 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.55:5060;rport;branch=z9hG4bK396ae76fab6e973ce8a35b3ccabd9918
From: "Extn0035" <sip:0035@192.168.1.59>;tag=1d2560e4110b811e
To: <sip:87521512@192.168.1.59>
Call-ID: c1f9987a1326024ed5ad48f0efdf5dd6@192.168.1.55
CSeq: 521759880 INVITE
Contact: "Extn0035" <sip:0035@192.168.1.55:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Content-Type: application/sdp
Supported: timer
Content-Length: 298

v=0
o=UserA 1745641786 240594919 IN IP4 192.168.1.55
s=Session SDP
c=IN IP4 192.168.1.55
t=0 0
m=audio 49152 RTP/AVP 18 8 0 4 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
11866713mS SIP Tx: UDP 192.168.1.55:5060 -> 192.168.1.59:5060
INVITE sip:87521512@192.168.1.59 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.55:5060;rport;branch=z9hG4bK396ae76fab6e973ce8a35b3ccabd9918
From: "Extn0035" <sip:0035@192.168.1.59>;tag=1d2560e4110b811e
To: <sip:87521512@192.168.1.59>
Call-ID: c1f9987a1326024ed5ad48f0efdf5dd6@192.168.1.55
CSeq: 521759880 INVITE
Contact: "Extn0035" <sip:0035@192.168.1.55:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Content-Type: application/sdp
Supported: timer
Content-Length: 298

v=0
o=UserA 1745641786 240594919 IN IP4 192.168.1.55
s=Session SDP
c=IN IP4 192.168.1.55
t=0 0
m=audio 49152 RTP/AVP 18 8 0 4 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
11866728mS SIP Rx: UDP 192.168.1.59:5060 -> 192.168.1.55:5060
SIP/2.0 181 Call Is Being Forwarded
Via: SIP/2.0/UDP 192.168.1.55:5060;rport;branch=z9hG4bK396ae76fab6e973ce8a35b3ccabd9918
From: "Extn0035" <sip:0035@192.168.1.59>;tag=1d2560e4110b811e
To: <sip:87521512@192.168.1.59>;tag=93c17ed2-17f1-4401-a6c1-daf86be59929
Call-ID: c1f9987a1326024ed5ad48f0efdf5dd6@192.168.1.55
CSeq: 521759880 INVITE
User-Agent: Direction
Content-Length: 0

11866730mS SIP Call Rx: 17
SIP/2.0 181 Call Is Being Forwarded
Via: SIP/2.0/UDP 192.168.1.55:5060;rport;branch=z9hG4bK396ae76fab6e973ce8a35b3ccabd9918
From: "Extn0035" <sip:0035@192.168.1.59>;tag=1d2560e4110b811e
To: <sip:87521512@192.168.1.59>;tag=93c17ed2-17f1-4401-a6c1-daf86be59929
Call-ID: c1f9987a1326024ed5ad48f0efdf5dd6@192.168.1.55
CSeq: 521759880 INVITE
User-Agent: Direction
Content-Length: 0

11866744mS SIP Rx: UDP 192.168.1.59:5060 -> 192.168.1.55:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.55:5060;rport=1060;branch=z9hG4bK396ae76fab6e973ce8a35b3ccabd9918;received=192.168.1.59
From: "Extn0035" <sip:0035@192.168.1.59>;tag=1d2560e4110b811e
To: <sip:3187521512@192.168.1.59:5060>
Call-ID: c1f9987a1326024ed5ad48f0efdf5dd6@192.168.1.55
CSeq: 521759880 INVITE
Content-Length: 0

11866746mS SIP Call Rx: 17
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.55:5060;rport=1060;branch=z9hG4bK396ae76fab6e973ce8a35b3ccabd9918;received=192.168.1.59
From: "Extn0035" <sip:0035@192.168.1.59>;tag=1d2560e4110b811e
To: <sip:3187521512@192.168.1.59:5060>
Call-ID: c1f9987a1326024ed5ad48f0efdf5dd6@192.168.1.55
CSeq: 521759880 INVITE
Content-Length: 0

11866749mS SIP Rx: UDP 192.168.1.59:5060 -> 192.168.1.55:5060
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.55:5060;rport=1060;branch=z9hG4bK396ae76fab6e973ce8a35b3ccabd9918;received=192.168.1.59
From: "Extn0035" <sip:0035@192.168.1.59>;tag=1d2560e4110b811e
To: <sip:3187521512@192.168.1.59:5060>;tag=8a12502b9ed73dee142aafd44523c639
Call-ID: c1f9987a1326024ed5ad48f0efdf5dd6@192.168.1.55
CSeq: 521759880 INVITE
Contact: <sip:3187521512@192.168.1.62>
Supported: replaces
Content-Type: application/sdp
Content-Length: 143

v=0
o=call 706779 706780 IN IP4 192.168.1.62
s=-
c=IN IP4 192.168.1.62
t=0 0
m=audio 10008 RTP/AVP 18
a=rtpmap:18 G729/8000
a=ptime:20
11866751mS SIP Call Rx: 17
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.55:5060;rport=1060;branch=z9hG4bK396ae76fab6e973ce8a35b3ccabd9918;received=192.168.1.59
From: "Extn0035" <sip:0035@192.168.1.59>;tag=1d2560e4110b811e
To: <sip:3187521512@192.168.1.59:5060>;tag=8a12502b9ed73dee142aafd44523c639
Call-ID: c1f9987a1326024ed5ad48f0efdf5dd6@192.168.1.55
CSeq: 521759880 INVITE
Contact: <sip:3187521512@192.168.1.62>
Supported: replaces
Content-Type: application/sdp
Content-Length: 143

v=0
o=call 706779 706780 IN IP4 192.168.1.62
s=-
c=IN IP4 192.168.1.62
t=0 0
m=audio 10008 RTP/AVP 18
a=rtpmap:18 G729/8000
a=ptime:20
11866753mS Sip: 17.1129.0 43 SIPTrunk Endpoint(f541db30) SetRemoteRTPAddress to 192.168.1.62:10008 (index 0)
 
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I'm not 100% sure but it looks like re-invite is turned on. If so, try turning that off. Also, what is at 192.168.1.62? A gateway?

-----------------------------------
atcom_logo_small.jpg

Calgary Telephone Systems, Avaya LG Asterisk (FreePBX) VOIP & TDM
 
Upvote 0 Downvote
Hello atcom,

re-invite is already turned off.

the ipoffice(192.168.1.55) sends the call to a portability server(192.168.1.59)that only analyses the digits and send to the ip phone interface(192.168.1.62).

* version 6.1 worked fine, but 7.0(36),7.0(32) and 8.0(42) did not, I didnt try other versions.

Everytime the call is droped I receive this mensage:
SIP/2.0 481 Call Leg/Transaction Does Not Exist

what that means?
 
Upvote 0 Downvote
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