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Beginner

Help with secondary/internal SIP trunk on UC560

My company has a sip trunk to our voip provider and everything works great.  They don't use credentials to authenticate, just based on our ip address.  Recently we started working with an IVR vendor to setup a sip trunk between our UC560 and an on premise Asterisk server (I don't have control of the Asterisk server).  Here's what I've entered on on the UC560:

dial-peer voice 2222 voip

description IVR System

destination-pattern 351

session protocol sipv2

session target ipv4:10.27.75.25:5060

session transport udp

incoming called-number 351

voice-class sip dtmf-relay force rtp-nte

dtmf-relay rtp-nte

codec g711ulaw

ip qos dscp cs5 media

ip qos dscp cs4 signaling

no vad

When I place a call to 351 I receive a busy signal immediately and the IVR vendor says they don't see any traffic hitting their server.  I've included a couple of debugs captured during the failed call.  If someone could have a look and point me in the right direction that would be awesome.  I probably left some important info out so please let me know if I can capture anything else.

Thanks!

Travis

4 REPLIES 4
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Beginner

Additional information:

Here's what the dial-peer shows for the previous calls -

Last Disconnect Cause is "39  ",

Last Disconnect Text is "bearer capability not authorized (57)",

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

supplementary-service h450.12

no supplementary-service sip moved-temporarily

no supplementary-service sip refer

redirect ip2ip

sip

  registrar server expires max 3600 min 3600

  localhost dns:SIPCARRIERNAME.net

   outbound-proxy dns:SIPCARRIERNAME.net

  no update-callerid

  sip-profiles 1000

sip-ua

keepalive target dns:SIPCARRIERNAME.net

no remote-party-id

retry invite 2

retry register 10

timers connect 100

timers keepalive active 100

sip-server dns:SIPCARRIERNAME.net

host-registrar

Highlighted

Travis,

     According to the voice_debug the UC recieves a "403 From: URI not recognized".  It appears the device responding does not like the URI that the UC sends.  The far end device is not allowing the connection from the UC.  Hope this helps.

James Battisti

Highlighted

James-

Thanks for responding and checking the debug.  Obviously I have to work with the vendor of the IVR system and I'll share this information with them.  In the debug are you able to see what far end device is responding?  I just want to make sure the traffic is destined for the right place and not trying to use my PSTN sip trunk.  Thanks again for your help!

Travis

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Travis,

       Here is where we recieve the message:

018642: Feb 25 14:58:56.083: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 403 From: URI not recognized

Via: SIP/2.0/UDP XXX.XXX.219.20:5060;branch=z9hG4bK1524201F

From: "Server Room" <132>>;tag=13909D50-21F6

To: <351>;tag=metaswitch.

Call-ID: 795A8367-9D9611E3-A30C9961-4C7E51@

CSeq: 101 INVITE

Timestamp: 1393361936

Server:

Organization:

Content-Length: 0

We are recieving it from 10.27.75.25, since that is who the call was sent to.  Hope this helps Travis.