SIP trunk configuration

[k.elnokrashy1]

I have a cucm 9  connected to 2921 router MGCP Gate way 
we use  FXO line 

we tried to replace the FXO by SIP trunk 

but when we try to make a call from out side to the call manager 

the telephone ring but when we pick up the phone we don not hear any thing and the calling person still hearing the ring

and from inside to outside we hear the wrong number message  

I attached the configuration file - the ccsip debug text

I appricaiate your help 

[Rajan · Accepted solution]

Hi

For the incoming call from Huawei SoftX3000 V300R010 (SBC) into the gateway, the gateway receives an invite and send the same to the other side. Otehr end responds with a trying and ringing. When teh gateway sends this 180 ringing message to the SBC, it did not receive any response and timed out resulting in call disconnect with error code 102.

Could you check with the SBC side on this. Outbound call is in the debug provided. Can you provide the call details to check.

HTH

Rajan

[k.elnokrashy1]

this is the call detail 

[Vivek Batra]

Can you please add SIP traces of outbound call.... - Vivek

[s.ranjbari]

Hi Rajan.
I have this issue too, did you resolve it ??

[Vivek Batra · Accepted solution]

Rajan is right what is happening here. To add a note, although incoming INVITE mention the support of 100rel but this can be seen frequently that it's not supported by many providers. 180 Rining being sent by gateway in result expecting 100rel in *require* header.

You would like to disable 100rel under voice service voip and then re-check.

voice service voip

 sip

  rel1xx disable

- Vivek

[k.elnokrashy1]

thanks 

now i can inter establish a call from outside to inside 

but another problem appear

the peson inside cucm  can not hear the outside 

but the out side can hear 

[Vivek Batra]

Can you please share the SIP trace of that call including show voip rtp connection...

- Vivek

[k.elnokrashy1]

here they are

[Vivek Batra]

There are no sip traces attached.

- Vivek

[k.elnokrashy1]

i am sorry 

here it is 

[Vivek Batra]

This is not complete SIP trace. If you see, there is no INVITE coming from service provider and forwarded to CUCM...

- Vivek

[k.elnokrashy1]

hi vivek 

i am so sorry again 

here it is 

[Vivek Batra]

Well, so issue is with SIP bind interface. 200 OK (answer) is being sent to service provider using gig 0/0 interface, however I believe that service provider facing interface is gig 0/2.241. That might be the reason there is no ACK from service provider and hence no media coming from service provider end. However service provider has published correct IP address in SDP hence your internal users audio is being sent to service provider network correctly.

Is the dial-peer 1974 (/voip) which you're expecting to match for inbound calls from service provider? If yes, add the SIP bind command under dial-peer 1974;

  voice-class sip bind control source-interface GigabitEthernet0/2.241

  voice-class sip bind media source-interface GigabitEthernet0/2.241

Let us know how it goes.

- Vivek

[k.elnokrashy1]

hi vevk 

it still not work see the attached trace