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call loop

Hello Guys,

I'm facing a strange issue on a SIP gateway. We received traces from our provider about calls loops. It appears, if we call the main number, this call is directly send back by the gateway to the sip provider and the loops begin.

 

Call flow of a working calls:

 

Sip provider --> Cube/Gateway --> CUCM --> IP Phone

 

Based on my observation, it's like the call never user the configured translation rules.The number is presented with number 0161413 and have to be transform to +43161413

I add some translation rules but it's never used and calls go out to the SIP provider via dial-peer 150 (see attached config).

 

I add a rules and dial-peer for test but it's the same see below the modifiy config part

 

 voice translation-rule 1
 rule 1 /0161413/ /+43161413/
 rule 2 /^0161\(413.*\)$/ /+43161\1/
 rule 3 /^0120\(511.*\)$/ /+43120\1/

dial-peer voice 80 voip
description ** Calls Received from PSTN to primary Callmanager **
translation-profile incoming E164Incoming
destination-pattern +43161413
progress_ind setup enable 3
session protocol sipv2
session target ipv4:10.161.10.170
voice-class codec 1
dtmf-relay rtp-nte
fax rate disable
ip qos dscp cs5 signaling
no vad
!
dial-peer voice 81 voip
description ** Calls Received from PSTN to primary Callmanager **
translation-profile incoming E164Incoming
destination-pattern +43161413
progress_ind setup enable 3
session protocol sipv2
session target ipv4:10.161.10.171
voice-class codec 1
dtmf-relay rtp-nte
fax rate disable
ip qos dscp cs5 signaling
no vad
!

 

 

if someone has an idea...

 

Thanks,

 

Hervé Jacquemin

1 ACCEPTED SOLUTION

Accepted Solutions

Herve,you should create a

Herve,

you should create a inbound dial-peer to receive the calls from provider, example below.

dial-peer voice 151 voip
 description ** Calls from SIP Provider **
 translation-profile incoming E164Incoming
 incoming called-number .T
 progress_ind setup enable 3
 session protocol sipv2
 voice-class codec 1
 dtmf-relay rtp-nte
 fax rate disable
 ip qos dscp cs5 signaling
 no vad.

 

with 'voice translation-profile E164Incoming', you will modify the called number from /0161413/ to /+43161413/.

 

When the call comes in, it should match the dial-peer 151 and translate the called number to E164 format & then it will match the outgoing dial-peer 50 or 51.

 

also please shut down the dial-peers 189 & 155 during the testing.

 

//Suresh

Please rate all the helpful posts.

//Suresh Please rate all the useful posts.
6 REPLIES

Hello Hervé, I think you

Hello Hervé,

 

I think you should create Translation Pattern in CUCM to modify the main number to your internal extension(ex: reception), thus you can avoid the call loop.

 

//Suresh

Please rate all the helpful posts.

//Suresh Please rate all the useful posts.
New Member

Hello Suresh, it changed

Hello Suresh,

 

it changed nothing (i tested), the calls seems to never reach the CUCM. It seems the Dial-peer 150 match the call.

Correct me if I'm wrong, but to match a dial-peer, the gateway check all DP one by one, apply a translation rule (if set up) and check the destination pattern.

 

I have translation rules, I check my DP with/without translation rules apply in incoming and I have the same behavior. The dial-peer 150 match the call each time.

Thanks,

Hervé

 

Herve,you should create a

Herve,

you should create a inbound dial-peer to receive the calls from provider, example below.

dial-peer voice 151 voip
 description ** Calls from SIP Provider **
 translation-profile incoming E164Incoming
 incoming called-number .T
 progress_ind setup enable 3
 session protocol sipv2
 voice-class codec 1
 dtmf-relay rtp-nte
 fax rate disable
 ip qos dscp cs5 signaling
 no vad.

 

with 'voice translation-profile E164Incoming', you will modify the called number from /0161413/ to /+43161413/.

 

When the call comes in, it should match the dial-peer 151 and translate the called number to E164 format & then it will match the outgoing dial-peer 50 or 51.

 

also please shut down the dial-peers 189 & 155 during the testing.

 

//Suresh

Please rate all the helpful posts.

//Suresh Please rate all the useful posts.
New Member

Hello Suresh, Thanks a lot

Hello Suresh,

 

Thanks a lot for your help !! The issue is solved, I tested the new dial-peer (never think about an inbound DP) and works at first shot without shut the 2 other DP.

Best regards,

 

Hervé Jacquemin

Hello Herve,Gald to know the

Hello Herve,

Gald to know the issue if fixed, smiley

how will we send the numbers to provider? E164 format? There are 3 dial-peers pointing to provider with 2 of them with + (E164 format).

 

//Suresh

Please rate all the helpful posts.

//Suresh Please rate all the useful posts.
New Member

Yes we are using E164 mainly.

Yes we are using E164 mainly.

 

It's a cluster over wan and we are using TEHO accros the country :)

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