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CME to SfB Integration-Issue with intersite calls

Dear Team,


Kindly note that we are facing issues in inter site calls  , from SfB to CME.  Please note that all other calls , like internal,PSTN calls working well from SfB.Issue persits only for intersite calls,which is with the below call flow.

 

SfB --> CME --> ICT -->CUCM.

I will copy the relevent configuration here, and from the SfB side user will be dialing 4 as prefix for reaching CME. site access code from CME to CUCM is 63XXXX.

 

voice translation-rule 13
 rule 1 /^4\(.*\)/ /\1/
 rule 2 /\+971/ /9/
 rule 4 /^\+/ /900/

 

voice translation-rule 12
 rule 1 /^.*\(....\)/ /\1/

 

voice translation-profile fromSfB
 translate calling 12
 translate called 13

 

dial-peer voice 9 voip
 description fromSfB
 translation-profile incoming fromSfB
 preference 1
 destination-pattern .T
 b2bua
 session protocol sipv2
 session target ipv4:XXXX:5060
 session transport tcp
 incoming called-number .T
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad

I will attach the logs for a working call (SfB to CME ) and also non working one. (SfB to CUCM through CME).

Appreciate for any suggestions.

 

25 Replies 25

Nipun,

 

Thank you for your comments, now the its ringing,but once we take the call ,there is no RTP establishing. I have taken the logs as suggested by you and attached.

 

calling party: +97148049122 (from SfB)

called number: 4641833 , 4633505.

 

There are multiple clusters ,but the behavior is same.I had tried to bind the all packets to CME in sip config.but no change happens.

 

Please suggest.

Remove "h225 connect-passthru" from voice service voip/h323 and then test again please.



Nipun,

 

Tried that and its not working.

Nipun,

 

Infact I registered the CME as ICT trunk in CUCM. Do you believe is this can cause this ?

 

Changing as H323 gateway is not successful so far.

Nipun,

 

Thanks a lot for pointing , sip to h323 missing in the configuration. I can test only on tomorrow and will let you know.

Hello ,

 

Issue still there and experts please help.

Can you take another set of logs with the command removed ? save them in a text file rather than doc please.

Hello Nipun,

 

I have uploaded the fresh logs. Please note the command for passthrough in voice service voip has been removed  currently.

 

Call flow

 

SfB --> CME-->ICT(H323)--> CUCM Extension

 

Calling party: +97148049122

Called party: 4638742

Hello Nipun, I can see the Bye msg which sending CME to the SIP server.
CSeq: 102 BYE
Reason: Q.850;cause=47
But please note I am using a single codec , G711ulaw in all environment. Transcoder was also registered in CME with 10 sessions , even tried by voice-class codec doesnt fixed my issue

Hello Nipun,

 

If I try to register the CUCM as SIP trunk to CME , this particular scneario is working . But we do have lot of intersites most of them are H323 ICT or H323 gateway. Is there any way to get it fixed with H323

I don't the GW initiate a TCS which is part of the issue. Can you enable "inbound fs" on for this h.323 router on CUCM and also make sure that "Wait for Far end cap" is unchecked.
Test it.

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