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New Member

incoming sip calls are not working but outgoing is working with cme

I have CME setup with voip.ms on my 2800 router, my outgoing calls are working  but my incoming calls are not.  Below is my config, please let me know if it is something with my config:

voice translation-rule 3
 rule 1 /^9142281\(...\)$/ /\1/

voice translation-profile INCOMING_CALL_1
 translate called 3

 

dial-peer voice 1 voip
 translation-profile incoming INCOMING_CALL_1
 session protocol sipv2
 session target sip-server
 incoming called-number .%
 voice-class codec 1
 dtmf-relay rtp-nte
 no vad

Everyone's tags (2)
8 REPLIES
VIP Gold

Dear, Please attach 'debug

Dear, Please attach 'debug voip ccapi inout' for an incoming test call & sh run.

Suresh

New Member

The debug showed no output

The debug showed no output when I called, I verified that I do own this number on voip,ms

 

Attached is my config

VIP Gold

Did you check the same doing

Did you check the same doing 'show log' ? and Can you check below debug for one call as well.

deubg ccsip messages

If you will not get any debug messages, that means call is not hitting to router and in that case you need to ask to service provider.

Suresh

 

 

 

New Member

So I contact voip.ms and I

So I contact voip.ms and I got that issue resolved, I am able to see the call on my gateway.  Turns out it was not yet registered.  But the calls are not redirecting to my extension:  Attached is my output of debug ccsip messages

VIP Gold

Dear, You have configured

Dear, You have configured below translation profile wrongly. It should be "translate called 3" so modify it once and try. If still issue persists then attach 'debug voip ccapi inout'

voice translation-profile INCOMING_CALL_1
 translate calling 3

Suresh

New Member

I made the change, but I am

I made the change, but I am getting no output from debug voip ccapi inout.  What does concern me from debug ccsip messages is:

Aug 31 12:42:04.195: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 400 Bad Request - 'Invalid Host'
Via: SIP/2.0/UDP 107.6.67.238:5060;branch=z9hG4bK000d3c36;rport
From: "+19144410197" <sip:9144410197@107.6.67.238>;tag=as7439b9c1
To: <sip:9142281462@108.54.238.199:1061>;tag=829C8-2532
Date: Sun, 31 Aug 2014 12:42:04 GMT
Call-ID: 2b4b66bd109bfeea6f4e15b403fb5514@107.6.67.238:5060
CSeq: 102 INVITE
Allow-Events: telephone-event
Reason: Q.850;cause=100
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0

 

I also am getting this:

voicertr2#debug ccsip error
SIP Call error tracing is enabled
voicertr2#
Aug 31 12:45:07.359: //-1/xxxxxxxxxxxx/SIP/Error/sipSPI_validate_own_ip_addr: ReqLine IP addr does not match with host IP addr
Aug 31 12:45:07.359: //-1/78AE76E98009/SIP/Error/sact_idle_new_message_invite: Invalid URL in incoming INVITE

 

New Member

I found the issue, I had to

I found the issue, I had to put a destination in my config for the dial-peer.  Below is the config for the incoming dial-peer

dial-peer voice 1 voip
 translation-profile incoming INCOMING_CALL_1
 destination-pattern .T
 session protocol sipv2
 session target sip-server
 incoming called-number .%
 voice-class sip bind control source-interface FastEthernet0/0.50
 voice-class sip bind media source-interface FastEthernet0/0.50
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad

 

according to your config

according to your config

 

voice translation-rule 2
 rule 1 /^.*/ /9142281462/
!
voice translation-rule 3
 rule 1 /^9142281\(...\)$/ /\1/

 

incoming called number is 9142281462

and it is translated to 1462

but you have two numbers 462 462

 

so you can fix it, making it simple and easy to understand

voice translation-rule 3
 rule 1  /^9142281462/  /462/

 rule 2  /^9142281463/  /463/

 

 

if it is not helps

so do

clea logg

deb ccsip mess

make call

sh logg

 

 

 

and dont forget to rate post

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