SIP incoming calls called number not correct

[corycandia]

Experts,

I am experiencing an issue with incoming SIP calls that me and Google can't resolve.

I have SIP service setup to have one incoming number XXX-660-5634.  Using a translation rule just for testing, I've removed the area code from the number and created a parellel hunt-group with this as it's pilot.

When calls come in, they randomly come in using other dial-peers as the called number.

Example (debug ccsip calls):

Jan 18 22:02:19.251: //13116/9D66B0448800/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x4B780EC0
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : XXX-6605015 <--------(**This number is not associated with the SIP service, or anything else, it's a cell phone)
Called Number : 104  <-----------(**This is one of my extensions, the translation rule should make this 660-5634
Source IP Address (Sig ): 192.168.3.1
Destn SIP Req Addr:Port : 162.243.210.200:5060
Destn SIP Resp Addr:Port : 162.243.210.200:5060
Destination Name : 162.243.210.200

What gives?  It randomly chooses dial peers, sometimes the called number is my 9911 dial-peer, sometimes it's a dial-peer with the incoming line from a POTS line.  I can't figure out what it's doing that.

[Dennis Mink]

Can you add a debug ccsip message output of a 'failed' call to this port so we can see what the INVITE from google contains?

cheers

[corycandia]

Thanks for taking an interest.

I have attached the output during an incoming call.  I have tried looking at the output from messages, but I don't have the experience to interpret the output.

I see that the 'to:' specifies one of the other numbers I have.  The 8109324 number is a POTS line.  It's one of the dial-peers that gets randomly selected when the SIP line '660-5634' gets a call.

I don't know how to determine why it's selecting these other numbers.

I have seen it said that when the CME doesn't have a matching rule, it uses rule 0 or selects one for you, but I thought I did the translation rule correct to say any number coming in from the SIP service, change it to 660-5634?

Any ideas?

[Dennis Mink]

the SIP invite shows a call for: XXX8109324 coming from your SIP provider

what needs to happen is that dial peer 2 needs to get hit:

dial-peer voice 2 voip
description INCOMING CALLS FROM GVGW
translation-profile incoming INCOMING-GVGW
session protocol sipv2
session target sip-server
incoming called-number .%
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
codec g711ulaw

you would need to apply 

voice translation-profile INCOMING-GVGW
translate called 1 

to that dial peer 2 so that inbound calls get translated into 6605634 

you can use debug dial peer, whilst making a inbound test call to see if you are hitting the correct dial peer.

also take the % off of the incoming called-number .% statement.   I never use it

Please rate if useful

[corycandia]

For some reason, debug dialpeer doesn't show any output when I have a call coming in.

I check that I already have the translation rule applied, at least I think.

I'm looking at the dial-peer:

dial-peer voice 2 voip
description INCOMING CALLS FROM GVGW
translation-profile incoming INCOMING-GVGW
session protocol sipv2
session target sip-server
incoming called-number .
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
codec g711ulaw
no vad

Now looking at the translation profile and rule:

voice translation-rule 1
rule 1 /.*/ /6605634/

voice translation-profile INCOMING-GVGW
translate called 1

I believe what you have recommended is already set.

I reviewed sip messages when a call is inbound and attached that.  It looks like the translation rule is still being ignored, so maybe I screwed that part up.  It's also weird that the called number is always a dial-peer entry.  The provider isn't trying to send the call to a particular end point right?

I included the output of sip-us register status, which I am still researching and trying to understand.

[corycandia]

While reviewing Cisco's CME example config again, I noticed they added:

no-reg both

or

no-reg primary

to ephone-dns.  I wonder if that's what I'm screwing up?  They are registering on the service provider proxy and it's randomly choosing them instead of one main number for the DID (XXX-660-5634)?

[corycandia]

I think I found this issue might be related and the cause of all my life's problems.  Being rookie in any subject sucks.

https://supportforums.cisco.com/discussion/11337566/pots-dial-peer-trying-register-sip-registrar

I started adding no  sip-register to those dial-peers and they started disappearing from the results of show sip-ua register status

Then, there was nothing.  I added an ephone-dn with the number XXX6605634 and then that finally became the called number on the show ccsip calls.

I'm not sure why it doesn't show up as 660-5634 since the translation rule should be changing it.