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2901 CME: Problem with incoming call via SIP Trunk

ooiaiyoon
Level 1
Level 1

Dear All,

I have seen some others posted similar question regarding this but mine still doesn't work by using the reference solution.

Mine is quite standard setup too -> CME setup on my 2901 router, analog phone attach to my FXS port my outgoing calls are working  fine via SIP but my incoming calls are not.  Caller only listen to engage tone and analog phone is not ringing at all. Attached with my config and trace log of ccsip messages. Kindly assist. Thank you so much.

7 Replies 7

EU UC Support
Level 4
Level 4

First error message:

SIP/2.0 422 Session Timer too small

Try to play with session timers:

 

http://www.cisco.com/c/en/us/td/docs/ios/voice/cube/configuration/guide/vb_book/vb_book/vb_1369.html

 

Router(config)# voice service voip

Router(conf-voi-serv)# sip

Router(conf-serv-sip)# min-se time

 

 

Hi EU,

Thanks for your reply.

There is no changes after I add in min-se <time>, call disconnected straight if I add in min-se 90 or little more, call remains engage tone if I put in larger value, so at the end I remove this command line.

I guess something to do with my dial-peer, attached with my dial-peer log, I can't fully interpret the log.

Thank you.

 


 

Hi 

On your dial-peer 198 remove destination-pattern .T and add 

incoming called-number 093080413

 

Please make those changes and let me know if it makes the difference.

 

Please after changes post the output of a debug ccsip messages during an incoming call

 

 

Thanks

 

Regards

 

Carlo

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

Hi Carlo,

I did change but still the same, I still get session timer error in ccsip messages log but I doubt more in 'Dialpeer' log, seems like can't find a way out to reach destination.

dial-peer voice 198 voip
 description SIP IncomP Incoming Call
 translation-profile incoming INCOMING
 session protocol sipv2
 session transport udp
 incoming called-number 093080413
 voice-class sip bind control source-interface GigabitEthernet0/0
 voice-class sip bind media source-interface GigabitEthernet0/0
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad

 

My scenario is

093080413 (local no) called to 0383183648 (remote no) successful (via SIP trunk),

0383183648 (remote no) called to 093080413 (local no) failed (via SIP trunk).

Hi.

Can you please post the output of the following commands:

 

show sip-ua timers

show sip-ua retry

show sip-ua min-se

 

Thanks

 

Regards

 

Carlo

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

I left the place early today, I will let you know tomorrow,

can I know what are we trying to check from here?

thank you.

 

Hi Carlo,

Here it is

CME_2901#show sip-ua timers
SIP UA Timer Values (millisecs unless noted)
trying 500, expires 180000, connect 500, disconnect 500
prack 500, rel1xx 500, notify 500, update 500
refer 500, register 500, info 500, options 500, hold 2880 minutes
, registrar-dns-cache 3600 seconds
tcp/udp aging 5 minutes

CME_2901#show sip-ua retry
SIP UA Retry Values
invite retry count = 6   response retry count = 6
bye retry count    = 10  cancel retry count   = 10
prack retry count  = 10  update retry count    = 6
reliable 1xx count = 6   notify retry count   = 10
refer retry count  = 10  register retry count = 6
info retry count   = 6   subscribe retry count = 6
options retry count = 6

CME_2901#show sip-ua min-se
SIP UA MIN-SE Value (seconds)
Min-SE: 1800

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