02-18-2014 02:13 PM - edited 03-16-2019 09:48 PM
Hello all,
I am setting up a CME using dual registrations. With the generous help of people on this forum I have gotten the dual registrations to work. But now I need to know how to set up the dial-peer to use the registration information. For example, I have this set up:
sip-ua
authentication username 5555555555 password 7 08114342101A0A1A43
...
registrar 1 ipv4:11.11.11.11:6034 expires 3600
registrar 2 ipv4:22.22.22.22:6035 expires 3600
How do I set up a dial-peer to send traffic to one of the registrations? I've tried this, and it does not work:
dial-peer voice 105 voip
description **Outgoing Call to SIP Trunk**
translation-profile outgoing PSTN_Outgoing
destination-pattern 91%...........
session protocol sipv2
session target registrar ????WHAT DO I USE HERE???
voice-class codec 2
dtmf-relay rtp-nte
no vad
Thanks in advance.
Solved! Go to Solution.
02-18-2014 03:19 PM
Hi Tod,
Kindly go through this post, hope it answers your question:
http://tekcert.com/blog/2011/02/03/cme-configuration-example-sip-trunks-viatalk-and-voipms
Rate the post accordingly.
Regards,
Kevin
02-18-2014 05:08 PM
Tod,
you should use two different dial-peers and configure options ping on them When any of the links to the dial-peer is down, the dial-peer will be shutdown, hence there is no delay whatsoever
Please rate all useful posts
"The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
02-18-2014 02:22 PM
Hi Tod,
The "session target registrar " point to the SIP-TRUNK to the PSTN, as detailed exaplaination:
To designate a network-specific address to receive calls from a VoIP or VoIPv6 dial peer, use the session target command in dial peer configuration mode. To reset to the default, use the no form of this command.
A ideal situation would be to use session target ipv4:
dial-peer voice 105 voip
description **Outgoing Call to SIP Trunk**
translation-profile outgoing PSTN_Outgoing
destination-pattern 91%...........
session protocol sipv2
session target ipv4:11.11.11.11:6034 <<(1st SIP-TRUNK)
voice-class codec 2
dtmf-relay rtp-nte
no vad
dial-peer voice 106 voip
description **Outgoing 2ND Call to SIP Trunk**
translation-profile outgoing PSTN_Outgoing
destination-pattern 91%...........
session protocol sipv2
session target ipv4:22.22.22.22:6035 <<(2ND SIP-TRUNK)
voice-class codec 2
dtmf-relay rtp-nte
no vad
Rate the post accordingly.
Regards,
Kevin
02-18-2014 02:43 PM
Thanks for the quick reply Kevin.
My concern with using the two dial-peers and specific IP addresses is that if one of the trunks is down, there will always be a post dial delay while the CME fails over to the next dial-peer, correct? I was hoping to find a way to avoid that.
02-18-2014 03:19 PM
Hi Tod,
Kindly go through this post, hope it answers your question:
http://tekcert.com/blog/2011/02/03/cme-configuration-example-sip-trunks-viatalk-and-voipms
Rate the post accordingly.
Regards,
Kevin
04-15-2018 04:16 PM
02-18-2014 05:08 PM
Tod,
you should use two different dial-peers and configure options ping on them When any of the links to the dial-peer is down, the dial-peer will be shutdown, hence there is no delay whatsoever
Please rate all useful posts
"The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
02-19-2014 05:38 AM
Good stuff +5
Regards,
Yosh
02-20-2014 02:19 PM
Yes. Thanks for the timely info. It is much appreciated.
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