Outgoing call issue

Jawad Ahmad · ‎05-16-2016

Hello,

I'm facing issue in outgoing calls, incomming calls working fine. Outgoing calls choose correct dial-peer and interface but ITSP don't receive any call on that SIP Trunk. They receive a call from another SIP Trunk that's terminate on same VG.

Network Lay out is,

IP Phone>>>Site1 CME>>>IPVPN>>>HQ>>>SIP Trunk>>>ITSP

Rajan · ‎05-16-2016

Hi Jawad,

As per these debugs, the ITSP is not getting proper number and hence rejecting the call with 484 - address incomplete.

Since you are saying that we are hitting an incorrect sip trunk, pls let us know which sip trunk should be used for this call and also provide "debug voip ccapi inout" to find out why we are hitting the other sip trunk.

Thanks

Rajan

Jawad Ahmad · ‎05-16-2016

Thanks for you reply.

Yes we receive 484 error call going to another SIP Trunk.

751 trunk shoud be use, 751 trunk is terminate on Gig0/2 and 21,22 dial-peer using for outgoing dial-peer.

You can see attached file of Debug Voip CCAPI Inout.

Rajan · ‎05-16-2016

HI Jawad,

This is what i see in the debugs:

Initially the call is routed using the dial-peer 21 only which is the desired one but it fails with cause code 28 - incomplete number and hence it tries to route the same call using dial-peer 20 as the second option but the result is the same.

002717: *May 17 05:14:23.706: //11720/35BA579F8339/CCAPI/ccSaveDialpeerTag:
Outgoing Dial-peer=21

002748: *May 17 05:14:23.798: //11721/35BA579F8339/CCAPI/ccCallDisconnect:
Cause Value=28, Call Entry(Responsed=TRUE, Cause Value=28)

002770: *May 17 05:14:23.802: //11720/35BA579F8339/CCAPI/ccSaveDialpeerTag:
Outgoing Dial-peer=20

002797: *May 17 05:14:23.878: //11722/35BA579F8339/CCAPI/cc_api_call_disconnected:
Call Entry(Responsed=TRUE, Cause Value=28, Retry Count=0)

So only thing you need to concentrate on is the number you sent in the invite message and make sure that you are sending the number as required by the ITSP in order for the call to work.

001191: *May 16 10:38:10.535: //243/F461EE4180E2/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:909@10.200.7.157:5060 SIP/2.0
Via: SIP/2.0/UDP 10.68.33.210:5060;branch=z9hG4bK7378F
Remote-Party-ID: <sip:4848307@10.68.33.210>;party=calling;screen=no;privacy=off
From: <sip:4848307@192.168.126.1>;tag=103300-A6E
To: <sip:909@10.200.7.157>
Date: Mon, 16 May 2016 10:38:10 GMT
Call-ID: C847EE42-1A6B11E6-81DCB49E-268EAA19@10.68.33.210

HTH

Rajan

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Jawad Ahmad · ‎05-16-2016

but problem is ITSP don't receive any outgoing call from my side.

connectivity everything ok.

Rajan · ‎05-16-2016

Hi Jawad,

I dont think so. For the invite sent, we receive trying and 484 from the provider which means they are receiving the invite.

001191: *May 16 10:38:10.535: //243/F461EE4180E2/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:909@10.200.7.157:5060 SIP/2.0
Via: SIP/2.0/UDP 10.68.33.210:5060;branch=z9hG4bK7378F
Remote-Party-ID: <sip:4848307@10.68.33.210>;party=calling;screen=no;privacy=off
From: <sip:4848307@192.168.126.1>;tag=103300-A6E
To: <sip:909@10.200.7.157>
Date: Mon, 16 May 2016 10:38:10 GMT
Call-ID: C847EE42-1A6B11E6-81DCB49E-268EAA19@10.68.33.210

001192: *May 16 10:38:10.555: //243/F461EE4180E2/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.68.33.218:1052;branch=z9hG4bK7378F
Call-ID: C847EE42-1A6B11E6-81DCB49E-268EAA19@10.68.33.210
From: <sip:4848307@192.168.126.1:5060>;tag=103300-A6E
To: <sip:909@10.200.7.157>
CSeq: 101 INVITE
Content-Length: 0

001193: *May 16 10:38:10.619: //243/F461EE4180E2/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 484 Address Incomplete
Via: SIP/2.0/UDP 10.68.33.218:1052;branch=z9hG4bK7378F
Record-Route: <sip:10.200.7.157:5060;transport=udp;lr>
Call-ID: C847EE42-1A6B11E6-81DCB49E-268EAA19@10.68.33.210
From: <sip:4848307@192.168.126.1:5060>;tag=103300-A6E
To: <sip:909@10.200.7.157>;tag=sbc0803bfp7euta
CSeq: 101 INVITE
Reason: Q.850;cause=28;text="address incomplete"
Warning: 399 - "SoftX3000 R601-CCU Rel POS:[3103] Release from CR"
Content-Length: 0

Another thing I see in the SIP responses from ITSP that 10.68.33.218 (gi0/1) is referenced but you have used gi0/2 for all communication with ITSP for this dial-peer. I am not sure whether this has anything to do with the issue. Else you need to concentrate on the number sent only.

HTH

Rajan

Jawad Ahmad · ‎05-16-2016

actually Via: SIP/2.0/UDP 10.68.33.218:1052;branch=z9hG4bK7378F for other (SIP Trunk)

for this call should goes throuh Via: SIP/2.0/UDP 10.68.33.210:5060;branch=z9hG4bK7378F

Rajan · ‎05-17-2016

"

but problem is ITSP don't receive any outgoing call from my side.

connectivity everything ok."

How you have checked the connectivity ? do you have any firewall in your network. If so, pls allow all required ports for SIP communication.

Jawad Ahmad · ‎05-17-2016

By Ping, tracerout to SIP server.

I've no firewall.

Jawad Ahmad · ‎05-26-2016

Hi Rajan,

Problem was solved, it was routing issue when we configured PBR its solved.