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different SIP accounts per line / voice dn on outgoing calls

Frank Hohenadel
Level 1
Level 1

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

2 Replies 2

Jorge Armijo
Level 4
Level 4

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.

Dennis Mink
VIP Alumni
VIP Alumni

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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