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different SIP accounts per line / voice dn on outgoing calls

Hi,

I'm looking for a solution to use my different SIP accounts for the different line configurations (dn 1 and dn 2) on the 9971.

With dn 1 the system should use voice translation-rule 3 (phone number 06209123456) where for dn 2 the system should use translation-rule 4 (phone number 05209789012) for all outbound calls.

So far the system is always using the dial-peer voice 1 voip (where translation profile OUT is assigned). Even removing the translation profile assignment in dial-peer voice 1 voip doesn't help.

Configuration is attached.

Any ideas?

Regards, Frank

Everyone's tags (3)
2 REPLIES

different SIP accounts per line / voice dn on outgoing calls

Check this:

Support for Multiple Registrars on SIP Trunks on a Cisco Unified Border  Element, on Cisco IOS SIP TDM Gateways, and on Cisco Unified  Communications Manager Express

http://www.cisco.com/en/US/docs/ios/voice/cube/configuration/guide/vb_9655_ps5640_TSD_Products_Configuration_Guide_Chapter.html

Configuring Multiple Registrars on SIP Trunks

http://www.cisco.com/en/US/docs/ios/voice/sip/configuration/guide/sip_cg-multi-registrars.html

HTH

--
Jorge Armijo

Please remember to rate helpful responses and identify helpful or correct answers.

-- Jorge Armijo Please remember to rate helpful responses and identify helpful or correct answers.

different SIP accounts per line / voice dn on outgoing calls

first of all, what you are seeing is expected behaviour, all outbound traffic will be hitting:

dial-peer voice 2 voip

description **Incoming VOIP Call**

translation-profile incoming IN

session protocol sipv2

session target ipv4:10.10.10.1

incoming called-number .%

voice-class codec 1

voice-class sip dtmf-relay force rtp-nte

dtmf-relay rtp-nte

no vad

This is because your incoming called number .% is a catch all for all inbound dial peers. including a call originated from your 9971. I would do the following (please note that this is only a concept, I havent been able to test it).

dial-peer voice 2 voip

description **Incoming VOIP Call**

translation-profile incoming IN

session protocol sipv2

session target ipv4:10.10.10.1

incoming called-number 06209123456

voice-class codec 1

voice-class sip dtmf-relay force rtp-nte

dtmf-relay rtp-nte

no vad

!

dial-peer voice 3 voip

description **Incoming VOIP Call**

translation-profile incoming IN

session protocol sipv2

session target ipv4:10.10.10.1

incoming called-number  05209789012

voice-class codec 1

voice-class sip dtmf-relay force rtp-nte

dtmf-relay rtp-nte

no vad

these two dialpeers are now PSTN inbound.

create 2 more dial peers:

dial-peer voice 20 voip

translation-profile incoming IN 8x

answer-address   06209123456

voice-class codec 1

voice-class sip dtmf-relay force rtp-nte

dtmf-relay rtp-nte

no vad

!

dial-peer voice 21 voip

translation-profile incoming IN   9x

answer-address  05209789012

voice-class codec 1

voice-class sip dtmf-relay force rtp-nte

dtmf-relay rtp-nte

no vad

now create the two translation profile 9x and 8x, and these are just examples, all they need to do is add either an 8 or a 9 in front of any dialed number, 

now you will need to create two outbound dial peers, one for SIP provider A  (with the 9x destination pattern) and one for provider B (with the 8x destination pattern) and dont forget to strip of the 9 and the 8 respectively.

not really a s3xi solution, but it will work



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