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Calls dropped after 925 seconds

ambleside
Level 1
Level 1

Outbound only, 925 or 926 seconds after the call is placed, it drops on the callee end.  UC still keeps it up but the callee can't hear anything or can the caller.  It's gets classified as a type 16 disconnect - normal callee hang-up - but it's not.

Any ideas? I've looked for a timer of some sort with 925 as the key # but I can't find it.

100% of all calls, outbound only (inbound callers can talk as long as they like without disconnect.

No VAD is set on the lines. Ideas?

2 Replies 2

Joseph Chambers
Level 1
Level 1

SIP?

Yes.  Narrowed down the issue but not the fix.

Outbound call, carrier sends a set of re-invite message @ 12 minutes into the call which the UC just ignores...Never answers or ACKS at all - it's not even in the logs. But it's in the packet capture!  3 minutes later the call is dropped.  I got these messages from wireshark captures at the wan port.

Here is my idea:

SIP invite sets Min-SE of 1800 seconds.

Response does include any message that implies that gateway will control re-invites, therefore they are the responsibility of the calling party (UAC), right?

It sends an invite for the same session which we ignore.  We haven't hit the timer to send the gateway the message yet so we haven't...

Anyone concur?

-Aaron

Here is the initial invite:

INVITE sip:17035551212@204.255.255.228:5060 SIP/2.0

Via: SIP/2.0/UDP 204.255.253.58:5060;branch=z9hG4bKC79CD2

From: "Phillip Park" <4155551212>;tag=436084A4-21AE

To: <17035551212>

Date: Tue, 14 Dec 2010 20:36:57 GMT

Call-ID: BAB7DFB8-6F811E0-A5A8A0FB-CD34337C@204.255.253.58

Supported: 100rel,timer,resource-priority,replaces,sdp-anat

Min-SE:  1800

Cisco-Guid: 3060812702-0116920800-2778964219-3442750332

User-Agent: Cisco-SIPGateway/IOS-12.x

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

CSeq: 101 INVITE

Max-Forwards: 70

Timestamp: 1292359017

Contact: <4155551212>

Expires: 180

Allow-Events: telephone-event

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 226

v=0

o=CiscoSystemsSIP-GW-UserAgent 4476 1061 IN IP4 204.255.253.58

s=SIP Call

t=0 0

m=audio 17054 RTP/AVP 0 101

c=IN IP4 204.255.253.58

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

Here is the acceptance:

SIP/2.0 200 OK

To: <170355512120>;tag=3aafb255

From: "Phillip Park"<4155551212>;tag=436084A4-21AE

Via: SIP/2.0/UDP 204.255.253.58:5060;branch=z9hG4bKC7B0

Call-ID: BAB7DFB8-6F811E0-A5A8A0FB-CD34337C@204.255.253.58

CSeq: 103 PRACK

Contact: <3CB4C_ERVAIFIZAQI8O_GAMTQGRQ7GFM4W_RQ>

Organization: MetaSwitch

Server: DC-SIP/2.0

Supported: 100rel, resource-priority

Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info

Content-Length: 0

12-minutes later, here is the invite back.

INVITE sip:4155551212@204.255.253.58:5060 SIP/2.0

To: "Phillip Park"<4155551212>;tag=436084A4-21AE

From: <17035551212>;tag=3aafb255

Via: SIP/2.0/UDP 204.255.255.228:5060;branch=z9hG4bK-d87543-1a547d4f13b1da27481f-1-cHBmNWIyN2IwYmZkMGFmNDcyMWNlOQ..-d87543-

Call-ID: BAB7DFB8-6F811E0-A5A8A0FB-CD34337C@204.255.253.58

CSeq: 50543139 INVITE

Contact: <3CB4C_ERVAIFIZAQI8O_GAMTQGRQ7GFM4W_RQ>

Expires: 180

Max-Forwards: 69

Content-Type: application/sdp

Organization: MetaSwitch

Supported: 100rel, resource-priority

Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info

Content-Length: 167

v=0

o=- 3868851470 3868875795 IN IP4 204.255.255.228

s=-

c=IN IP4 204.255.255.228

t=0 0

m=audio 44276 RTP/AVP 0 101

a=ptime:20

a=rtpmap:101 telephone-event/8000

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