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Dial Peer Matching and Forwarding

cdowling_bm
Level 1
Level 1

Hello,

I need some help with matching and forwarding dial peers. I have an existing non-Cisco PBX which connects to a non-Cisco messaging system via a AS5300 series voice gateway.  I want use the existing messaging platform and gateway for a second non-Cisco PBX. 

PBX1 uses a pilot number of 5555406245 and PBX2 uses 5552226245 to access the messaging platform. Calls to the messaging platform via PBX1 are successful but I can't get the calls from PBX2 to work (the call fails in the handling of the dial peers).   

I have attached trace files for a successful call - PBX1 to messaging system - (ccapi-inout-call-successful-screen.txt) and for an unsuccessful call - PBX2 to Messaging platform (ccapi-inout-call-fail-screen.txt).  

Can anyone help me understand why my calls are failing?

Below is the config for the dial peers

 

PBX1 working dial peers
=======================

inbound from pbx1
-----------------

dial-peer voice 10002 pots
 trunkgroup PRIs
 huntstop
 preference 2
 incoming called-number 55554T
 direct-inward-dial

 

outbound to messaging platform
------------------------------

dial-peer voice 6201 voip
 destination-pattern 5555406245
 session protocol sipv2
 session target dns:vms.sip.noname.com
 dtmf-relay rtp-nte
 codec g711ulaw


PBX2 Dial Peers
===============

inbound from pbx2
-----------------

dial-peer voice 80 voip
 incoming called-number 555222....
 voice-class codec 1
 session protocol sipv2
 session target ipv4:162.XXX.XXX.XXX
 dtmf-relay rtp-nte


Outbound to messaging platform
------------------------------

dial-peer voice 85 voip
 destination-pattern 5552226245
 voice-class codec 1
 session protocol sipv2
 session target dns:vms.sip.noname.com
 dtmf-relay rtp-nte
!

9 Replies 9

Terry Cheema
VIP Alumni
VIP Alumni

You are getting below for your failed call:

Aug  9 21:19:47 ADT: //528966/576578C1A5B1/SSAPP:80:-1/ssaSetupPeer: dialpeer tags in rotary= 85  90
Aug  9 21:19:47 ADT: //528966/576578C1A5B1/SSAPP:80:-1/ssaSetupPeer: cid(528966), destPat(5552226245), matched(10), prefix(bm), peer(6FC3AEA4), peer->encapType (2)
Aug  9 21:19:47 ADT: //528966/576578C1A5B1/SSAPP:80:-1/ssaSetupPeer: cid(528966), can't gateway with peer tag(85))
Aug  9 21:19:47 ADT: //528966/576578C1A5B1/SSAPP:80:-1/ssaSetupPeer: cid(528966), destPat(5552226245), matched(4), prefix(), peer(648AA064), peer->encapType (2)
Aug  9 21:19:47 ADT: //528966/576578C1A5B1/SSAPP:80:-1/ssaSetupPeer: cid(528966), can't gateway with peer tag(90))
Aug  9 21:19:47 ADT: //528966/xxxxxxxxxxxx/CCAPI/ccTDUtilGetDataByValue: CallID[528966], tagID[24], instID[-1]

 

 

Aug  9 21:19:47 ADT: //528966/576578C1A5B1/CCAPI/ccCallDisconnect: (callID=0x81246, cause=0x3 tag=0x0)

CC_CAUSE_NO_ROUTE = 0x3

No route to destination (3)

Ref: http://www.cisco.com/c/en/us/td/docs/ios/voice/monitor/configuration/guide/12_4/vt_12_4_book/vt_ccodes_debug.html

 

You can try to remove the session target, For the inbound dial-peer voice 80 voip. Since this is just inbound dial-peer you dont need that.

 

Try that and make another test call. If still doesn't work take the below debugs:

1) Debug ccsip messages

2) debug voip dialpeer inout

3) debug voip ccapi inout

Turn these on together and attach the output again.

-Terry

Please rate all helpful posts

 

 

Kevin Monteiro
Level 7
Level 7

Hey,

 

This is your call-flow which fails. Need some debugs to help you out.

PBX2>>SIP>>>>(GW)AS54XX>>>>SIP>>Msging-System.

 

debug voip ccapi inout

 

debug ccsip message.

 

If you could attach the "show tech" that would be great.

 

-Kevin

 

Hi,

 

From the logs, I can see that your gateway isn't able to reach the destination peer which is 'vms.sip.noname.com'. Can you make sure that your gateway is able to resolve the domain name 'vms.sip.noname.com'? Try simple ping and see if this name can be resolved.

 

My preference to use IPs instead of names unless needed.

 

I initially thought the same but the OP has listed he is using the same gateway for both PBXs, Since its working for one PBX, it should be able to resolve the DNS

If that's not the case, if these logs are from different gateways, then definitely make sure DNS is resolved correctly.

-Terry

cdowling_bm
Level 1
Level 1

Thanks for the help so far.  
Terry, I tried removing the session target and the call failed.  The gateway can get busy at times but I am attaching the debugs as suggested (unedited so nothing gets missed). The test call is made from 4415000808 to 4412226245.
Kevin - Show tech is also attached
All - The DNS is resolving fine.  This is the only gateway being used for both PBXs and the one messaging platform.

Hi,

 

What is the SIP bind interface used in your configuration for control and media? Try to ping from your router and share the results

 

ping vms.sip.noname.com source #your_bind_interface#

 

I don't think this will work based on your logs. Also, if you can post your entire config will be great

Hi Mohammed.  The SIP bind is indeed used for both control and media.  A ping to the DNS name from the bind interface works fine.  See attached file.  For the complete config, see the show tech output in the above post.

Thanks.

Hi,

 

Sorry for missing show tech provided earlier. Couple of points:

 

1. Your gateway is missing the command 'allow-connections sip to sip'

voice service voip
 allow-connections sip to sip

2. You need to enable CEF on your interfaces using the command 'ip route-cache cef'. I have seen many problems in VGW when CEF is disabled

3. Can you ping the IP 162.255.216.18 using loopback 1 as source. I see that your gateway is sending 'Not Found' SIP message but this IP isn't sending ACK. I want to make sure that reverse reachability is present.

Hi Mohammed,

The VG does not have the command "allow-connections sip to sip" under voice service voip.  There is a SIP command which allows more configuration but there is no allow command.  It's a pretty old gateway.

I haven't enabled CEF yet so I will have to get back to you on that but I did do a ping as you suggested.  I can ping 162.225.216.18 from Loopback1 no problem.