cancel
Showing results for 
Search instead for 
Did you mean: 
cancel
414
Views
0
Helpful
4
Replies

Help with SIP profile

quevedo_lopez
Level 1
Level 1

Hi,

I'm trying to config a Cisco 2801 IOS 15.1M10 with a SIPTRUNK to one service provider.  The calls are working fine incoming, but incoming not.  The service provider told me that i need to change the extension number and ip address from to a one asigned DID, i've use a guide from the support site but still doesn't works, the outgoing call progress but no audio in both ways.  I attach the configuration and a debub ccsip messages.

Thanks advanced.

PS, update the config and debug in separates files.

4 Replies 4

Vivek Batra
VIP Alumni
VIP Alumni

I can initially see two issues here in the call flow with the assumption that gateway IP address pointing towards CM is 192.168.102.25 and towards service provider is 10.219.254.155;

1. 183 Session Progress sent to CM is sourced from incorrect IP viz 10.219.254.155. This is because dial peers are not bound with specific interfaces. Please corss verify and don't let any dial peer without bind interface command.

Bind all dial-peers towards CM with 192.168.102.25 and dial-peers towards service provider with 10.219.10.250.

2. Another issue I can see is 183 Session Progress sent to CM is with SDP offer. This is not correct behavior. As initial INVITE recieved from CM is without SDP and PRACK/100rel is not being used between CM and gateway, SDP offer must carry in reliable provisional response or final response (200 OK). This can lead to how CM interprets the SDP offer in non-reliable  provisional response but this is not correct behavior as per RFC 3264.

Now there could be many ways to resolve this issue. Either to not use PRACK/100rel at all or enable and use PRACK correctly.

Let's not use PRACK and see if it works for you.

Take following steps;

1. Enable SIP early offer in CUCM (under SIP Profile in CUCM)

2. Disable 100rel in gateway. voice server voip -> sip -> rel1xx disable

3. Don't forget to bind the dial peers with appropriate interface (both inbound and outbound dial peers)

Once done, please share the results.

I've not entertained your original quesiton still. As you said, your service provider wants to see specific number in INVITE. You can simple use voice translation rules to manipulate the number in a way you want.

- Vivek 

Please share further debugs/configuration only as a attachment.

- Vivek

Thank you Vivek let me try those changes. Yes the service provider says that they want to see a specific number in the the INVITE, i've tried just the translation but i'm still getting a disconnect from the SP.

Once you're done with the suggested changes, share the logs and will see furhter.

- Vivek

Getting Started

Find answers to your questions by entering keywords or phrases in the Search bar above. New here? Use these resources to familiarize yourself with the community: