cancel
Showing results for 
Search instead for 
Did you mean: 
cancel
535
Views
0
Helpful
13
Replies

SIP Call Not Working

Hello Everyone,

I have two CME sites connected via P2P. I have configured dial peer for calling between two sites. Calls from one site are working but from other site I am not able to make calls. Anyone can help me with this issue?

I am attaching debug ccsip all herewith.

 

Thanks

13 Replies 13

Vivek Batra
VIP Alumni
VIP Alumni

Can you also share the SIP traces of failed call?

I have attached debug ccsip all. Any other debug output you require in addition to this?

Hi.

Please activate a debug ccsip messages on both sites and make a call from the site where calls fail.

 

Thanks

 

Regards

 

Carlo

Please rate all helpful posts "The more you help the more you learn"

Hi.

Can you please post both vg config?

 

Thanks

Regards

 

 

 

Carlo

Please rate all helpful posts "The more you help the more you learn"

The configs are attached herewith.

I am not able to make calls from Router2 site to Router1.

 

Hi.

Modify your dial-peer 6 on router 2 as follows:

dial-peer voice 6 voip

session protocol sipv2

 dtmf-relay rtp-nte
 codec g711ulaw
 no vad

 

Try to call from R2 to R1 and let me know

HTH

 

Regards

 

Carlo

Please rate all helpful posts "The more you help the more you learn"

Hi,

Since the call is not working with SIP, so I have enabled H323 on the dial peer so that calls work fine.

I have tried the method that you have described but no success.

Hi.

Can you please activate a debug voip dialpeer inout on both routers and make a call from R2 to R1 and post the output?

Thanks

 

Regards

 

 

Carlo

Please rate all helpful posts "The more you help the more you learn"

Hi Carlo,

 

This is the output of the debug from where calls are not working. There is no output on the other router.


Log Buffer (6000000 bytes):

001892: Mar  2 2015 11:44:57.370 IST: %SYS-5-CONFIG_I: Configured from console by TSAL.ADMIN on vty0 (192.168.100.12)
001893: Mar  2 2015 11:45:08.450 IST: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=1039, Peer Info Type=DIALPEER_INFO_SPEECH
001894: Mar  2 2015 11:45:08.450 IST: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=1039
001895: Mar  2 2015 11:45:08.450 IST: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Result=Success(0) after DP_MATCH_DEST
001896: Mar  2 2015 11:45:08.450 IST: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:
   dialstring=NULL, saf_enabled=0, saf_dndb_lookup=0, dp_result=0
001897: Mar  2 2015 11:45:08.450 IST: //-1/xxxxxxxxxxxx/DPM/dpMatchPeers:
   Result=SUCCESS(0)
   List of Matched Outgoing Dial-peer(s):
     1: Dial-peer Tag=6

 

Thanks

Hi.

Can you please add session protocol sipv2 to dialpeer 6 and collect a debug voip diaper inout and a debug ccsip message on both router and post the result here?

 

Thanks

 

 

Regards

 

Carlo

Please rate all helpful posts "The more you help the more you learn"

Hi Carlo,

I had taken the debug by adding session protocol sipv2 under the dial peer 6.

When I enable the debug ccsip messages on the router from where call is failing I don't get any output for that debug.

 

It seems issue related to SDP.

As mentioned by Carlo, debug ccsip messages command will generate the SIP traces and will help in further diagnosis.

Hi,

I have tried enabling debug ccsip messages. But whenever I make a call nothing is shown in that debug. I am able to get the output of debug ccsip all only.

When I enable H323 under that dial peer, the call works fine. But I am not able to make calls using SIP.