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Cisco CME refused incoming call

faruk.zaimovic
Level 1
Level 1

Hello, 

I want to test SIP trunk on my Cisco CME. I make call from local phone to out, but i can not make call from outside to my local. 

i run debug, and i found .

%CALL_CONTROL-6-CALL_LOOP: The incoming call has a global identifier already present in the list of currently handled calls. It is being refused.

 

Allow-Events: telephone-event

Warning: 399 10.10.38.18 "Transcoder Not Configured"

 

i attached all debug.

Does anybody have same problem, and please help me.

1 Accepted Solution

Accepted Solutions

Absolute sure that I did not suggest you to add this.

ip address trusted list
ipv4 0.0.0.0 0.0.0.0
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323

Please correct and then get back with the debug in a text file. I don't know why you can't upload a plain text file, that normally works. What text editor do you use, suggest that you use Notepad++ and save it as plain text file.



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View solution in original post

30 Replies 30

With all the debugs you have turned on it makes it harder to pinpoint the cause of your issue as there is so much white noise in the output. Can you please redo the call with these debugs running, debug ccsip message and debug voip ccapi inout, also please share your running configuration so that we can look it through.



Response Signature


Hi,

From the debug it seems that you have not configured any dialpeer matching incoming calls from the provider so the CME uses the default DP which negotiates G729 and uses H323 signaling protocol that’s why you send Internal Server Error message. As @Roger Kallberg mentioned without the running config is hard to tell you exactly where the issue comes from.

Regards

 

Carlo

Please rate all helpful posts "The more you help the more you learn"

faruk.zaimovic
Level 1
Level 1

Hello all,

Thank you very much for help Carlo and Roger.

I attached here SIP-TRUNK tar file, where you can find my running conf and debug ccsip message and debug voip ccapi inout in separate files.  I deleted other phones and users from conf, there is only one user Slobodni 678 and register pool 109 for that user. That users use  cor incoming Test default, i want only to make one local number 678 goes over SIP trunk. 

I made this two dial-peer for SIP. One for in other for out calls

dial-peer voice 200 voip
corlist outgoing Test
translation-profile outgoing Odlazni-HT
destination-pattern 0.T
session protocol sipv2
session target ipv4:10.10.38.5:5060
voice-class codec 10
voice-class sip profiles 1
dtmf-relay rtp-nte sip-notify
no vad
!
dial-peer voice 201 voip
translation-profile incoming Dolazni-HT
session protocol sipv2
session target ipv4:10.10.38.5:5060
incoming called-number .%
voice-class codec 10
voice-class sip profiles 1
dtmf-relay rtp-nte sip-notify
no vad

I hope that I give you enough info.

Thank you very much for effort and help.

I haven’t checked your TAR file and likely will not as I’m kind of restrictive in opening files other than just plain text. Based on the information provided about the dial peers I will suggest that you remove these two lines from your inbound dial peer.

session target ipv4:10.10.38.5:5060
incoming called-number .%

The first line is used on outbound dial peers, as such it’s not needed on an inbound dial peer. The second line is not a good way to match inbound calls, replace this with a match that uses information in the VIA header. See this document for how to, In Depth Explanation of Cisco IOS and IOS-XE Call Routing



Response Signature


Hi,

Luckly with sip we have many options to match an incoming dialpeer and called number it's something that I would use with old pstn connections.

Let's try to match the host portion of "VIA" value of INVITE message:

voice class uri 10 sip

host 10.10.38.5

Now create a DP that should match this portion of invite message

dial-peer voice 220 voip

translation-profile incoming Dolazni-HT (I believe that this is translating the called number +38736325353 with an internal extension)
session protocol sipv2

session transport udp

incoming uri via 10

voice-class codec 10
voice-class sip profiles 1
dtmf-relay rtp-nte sip-notify
no vad

Please let us know

 

Regards

 

Carlo

 

 

Please rate all helpful posts "The more you help the more you learn"

Hello Carlo,

Thank you very much for help.

I added that configuration, and situation is same. My dial peer are now

dial-peer voice 200 voip
corlist outgoing Test
translation-profile outgoing Odlazni-HT
destination-pattern 0.T
session protocol sipv2
session target ipv4:10.10.38.5
voice-class codec 10
voice-class sip profiles 1
dtmf-relay rtp-nte sip-notify
no vad
dial-peer voice 201 voip
translation-profile incoming Dolazni-HT
session protocol sipv2
session transport udp
incoming uri via 10
voice-class codec 10
voice-class sip profiles 1
dtmf-relay rtp-nte sip-notify
no vad

 

 

voice class uri 10 sip
host 10.10.38.5

 

 

Hi Faruk,

With the above configuration applied, please activate a debug ccsip message , place a call and post the output.

 

Thanks

Regards

 

Carlo

Please rate all helpful posts "The more you help the more you learn"

faruk.zaimovic
Level 1
Level 1

Hello Roger,

Thank you very much, I understand you. Sorry because i put in in tar file. This site have some restriction which kind of file you can upload. I upload again running config, debug ccsiop message and debug voip ccapi inout. Thnak you very much. I delete session target ipv4:10.10.38.5:5060
incoming called-number .% 

and situation is same.

now my dial peeer are. 

dial-peer voice 200 voip
corlist outgoing Test
translation-profile outgoing Odlazni-HT
destination-pattern 0.T
session protocol sipv2
session target ipv4:10.10.38.5
voice-class codec 10
voice-class sip profiles 1
dtmf-relay rtp-nte sip-notify
no vad
dial-peer voice 201 voip
translation-profile incoming Dolazni-HT
session protocol sipv2
voice-class codec 10
voice-class sip profiles 1
dtmf-relay rtp-nte sip-notify
no vad


Thank you very much for help

I see you deleted what I suggested, but you have not added what I suggested. Without any configuration to match your inbound dial peer it won’t work.



Response Signature


Please have both of the debugs running and post the output. That’s what I wrote in my first reply.

Also please post the output in a txt file, not a docx.



Response Signature


faruk.zaimovic
Level 1
Level 1

Hello Roger,

Sorry, I attached in txt format. I hope that you will help me.

Thanks for the text files, I will look at the configuration. However it's now the third time I ask you to provide the debugs in one file. I'm sorry, but I will not look at individual files with debug output as you'd want to see the output of these in combination.



Response Signature


Try with adding these changes.

voice service voip
 ip address trusted list
  no ipv4 0.0.0.0 0.0.0.0 !Not a good thing to have as it opens up for toll fraud, you'd want to have the ITSP SIP server IP(s) defined
  ipv4 <ITSP SIP server IP>
  ! add as many lines as you need
 no allow-connections h323 to h323 !Not needed with SIP
 no allow-connections h323 to sip !Not needed with SIP
 no allow-connections sip to h323 !Not needed with SIP
 address-hiding
 mode border-element !Save and reboot once you have added this and the rest of the suggested changes to activate Cube
 sip
  bind control source-interface GigabitEthernet0/0/0
  bind media source-interface GigabitEthernet0/0/0
!
no voice class codec 10 ! You don't need all those codecs
voice class codec 10
 codec preference 1 g711alaw
 codec preference 2 g711ulaw
 codec preference 3 g722-64
 codec preference 4 g729r8
!
no voice class sip-profiles 1 !Not needed when you have turned on Cube functionallity with mode border-element and have address-hiding in global config
!
voice translation-rule 100
 rule 1 /^678$/ /+38736325353/ !Set to match starts and ends with, ie will only match the exact string
!
voice translation-rule 200
 rule 1 /^\+38736325353$/ /678/ !Set to match starts and ends with, ie will only match the exact string and also add start with a plus (+)
 no rule 2 !Not needed as it is the same number as rule 1
!
voice class uri PSTN sip !Defines what to look for the the SIP header, used to match the IP in the VIA header on the inbound dial peer 201
 host ipv4:10.10.38.5
!
dial-peer voice 200 voip
 corlist outgoing Test
 translation-profile outgoing Odlazni-HT
 destination-pattern 0T !No need to have a dot after the zero and before the T
 session protocol sipv2
 session target ipv4:10.10.38.5:5060
 voice-class codec 10  
 no voice-class sip profiles 1
 voice-class sip bind control source-interface GigabitEthernet0/0/2
 voice-class sip bind media source-interface GigabitEthernet0/0/2
 dtmf-relay rtp-nte sip-notify
 no vad
!
dial-peer voice 201 voip
 translation-profile incoming Dolazni-HT
 session protocol sipv2
 no session target ipv4:10.10.38.5:5060 !Used on outbound dial peers
 no incoming called-number .% !Not a good way to match inbound dial peer
 incoming uri via PSTN !Matches inbound dial peer based on information in VIA header
 voice-class codec 10  
 no voice-class sip profiles 1
 voice-class sip bind control source-interface GigabitEthernet0/0/2
 voice-class sip bind media source-interface GigabitEthernet0/0/2
 dtmf-relay rtp-nte sip-notify
 no vad


Response Signature


Hi Faruk.

In your config you are missing a bind interface for SIP service.

under voice service voip

sip

bind all source-interface gi 0/0/2

Apply this and let us know 

 

Regards

 

Carlo

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