How Does a SIP Trunk Dial-Peer Use a Registrar?

[tod.higman]

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.

[Kevin Monteiro]

Hi Tod,

The "session target registrar "  point to the SIP-TRUNK to the PSTN, as detailed exaplaination:

session target (VoIP dial peer)

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: of the ITSP:

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

[tod.higman]

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.

[Kevin Monteiro · Accepted solution]

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

[Ayodeji Okanlawon · Accepted solution]

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

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"The essence of christianity is not the enthronement but the obliteration of self --William Barclay"

[yahsiel2004]

Good stuff +5

Regards,

Yosh

[tod.higman]

Yes.  Thanks for the timely info.  It is much appreciated.