10-20-2013 09:44 AM - edited 03-16-2019 07:59 PM
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
10-20-2013 03:01 PM
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
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.
10-20-2013 06:35 PM
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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