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VoIP Dialling out using different usernames

I have a CUCM 9.1 with a CUBE running on Cisco ISR 2811 version 15.0. We have two sets of numbers on the system and I want to be able to have option of calling out using different sip accounts if possible.

Extensions with 610 to 619 calling out using 01234 local number and Extensions 860 to 869 calling out using 01608 number. This is what I have but it does not work. Where am I going wrong?

 

dial-peer voice 3010 voip
 description **01608 Out**
 destination-pattern 0.T
 session protocol sipv2
 session target sip-server
 incoming called-number 86[1-9]
 voice-class codec 1  
 voice-class sip early-offer forced
 dtmf-relay rtp-nte sip-notify
 ip qos dscp cs5 media
 ip qos dscp cs4 signaling
 no vad
 authentication username 889d0f09 password 7 ****


dial-peer voice 3011 voip
 description **01234 Out**
 incoming called-number 610
 session protocol sipv2
 session target sip-server
 voice-class codec 1  
 voice-class sip early-offer forced
 dtmf-relay rtp-nte sip-notify
 ip qos dscp cs5 media
 ip qos dscp cs4 signaling
 no vad
 authentication username 0f5982b0 password 7 **** 


dial-peer voice 100 voip
 description **from CUCM for Outbound calls**
 session protocol sipv2
 session target sip-server
 destination-pattern 0.T
 voice-class codec 1  
 dtmf-relay rtp-nte sip-notify
 ip qos dscp cs5 media
 ip qos dscp cs4 signaling
 no vad

5 Replies 5

kkoeper12
Level 3
Level 3

Can you better define "does not work"? You mean the calls fail or they all go out one dial-peer?

Dial peer 3011 is not set up for outbound calls, you will need to add a destination pattern if you want to use it to call out.

You are matching incoming extensions 861-869 on dial-peer 3010 and are matching incoming extension 610 on dial-peer 3011 but this will have no effect on which outbound dial-peer is chosen. Since you have no preference configured and you have an equal match on the destination pattern of 0.T for dial-peers 3010 and 100 the CUBE should load balance the outgoing calls between these 2.

If you want a certain range of incoming extensions to use a certain outgoing dial-peer you could use incoming voice translation patterns assigned to the dial peers to differentiate them. ie add a "9" in front of outgoing calls from extensions 861-869 and add an "8" in front of outgoing calls from ext 610-619 then change the destination pattern for dial peer 3010 to be 90.T and add a destination pattern for dial peer 3011 of 80.T.

You will also need an outgoing voice translation pattern to strip the 8 and 9.

 

 

 

Sorry for a late reply. I want to be able to use some extensions use one dial-peer to call out with and some with another. I have two different SIP accounts and want two departments to have their own outgoing CLI.

 

At the moment both ranges seem to call out with the second sip account. 

Hi Dmitry Golovenkin,

 try the following:

 

dial-peer voice 3010 voip
 description **Receiving calls from 86x to 01608 Out**
 session protocol sipv2
 session target sip-server
 answer-address 86.
 voice-class codec 1  
 voice-class sip early-offer forced
 dtmf-relay rtp-nte sip-notify
 ip qos dscp cs5 media
 ip qos dscp cs4 signaling
 no vad
 authentication username 889d0f09 password 7 ****

dial-peer voice 3011 voip
 description **Receiving call from 61x to 01234 Out**
 answer-address 61.
 session protocol sipv2
 session target sip-server
 voice-class codec 1  
 voice-class sip early-offer forced
 dtmf-relay rtp-nte sip-notify
 ip qos dscp cs5 media
 ip qos dscp cs4 signaling
 no vad
 authentication username 0f5982b0 password 7 ****

dial-peer voice 100 voip
 description **from CUCM for Outbound calls**
 session protocol sipv2
 session target sip-server
 destination-pattern 0.T
 voice-class codec 1  
 dtmf-relay rtp-nte sip-notify
 ip qos dscp cs5 media
 ip qos dscp cs4 signaling
 no vad

 

Hope this helps.

 

Thanks for your help but that did not make any difference.

Hi Dmitry Golovenkin,

 could you send a log of the call? I would like to check the flow.

 

debug voip ccapi inout

 

Regards.

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