SIP CUBE dial-in problems

[Paul Austin]

Hi Folks, I'm having a real problem with migrating to a new ISP for a customer. The original one works fine but he new one just doesnt want to play ball and I'm at a loss to be honest. So, outbound is fine but inbound is not.

Call made from PSTN, through CUBE and phone on UCM rings, if I answer it, the call is still ringing on the PSTN phone, the call duration time is increasing on the UCM phone then eventually it clears down.

If I make the same call from the PSTN but le it ring without answering, the UCM phone rings for a while, stops but I still get rininging on the PSTN phone for a while before that is cleared down.

What I can see from the CUBE - debug ccsip messages output is that I get a 504 Gateway Timeout

Any ideas?

I have attached the relevant debug file.

Thanks in advance

[Mohammed al Baqari]

Try to apply this config.

voice class sip-profiles 1
response 200 sip-header Require REMOVE
!
voice service voip
sip
sip-profiles 1

Also remove the voice-class bind command from dialpeer 5.

If didn't work please share the latest debugs.

[Paul Austin]

no sorry. The call came in then rang the destination for 3 - 4 times then disconnected, I did manage to answer it once but the call disconnected after 10 seconds or so.

Thanks

[Mohammed al Baqari]

I don't think you removed the voice-class bind command from your dialpeer as I can still see it. Please try to remove it and share the debugs. I don't see the right IP reflecting

[Paul Austin]

yep removed - see config.

Thanks again.

[Mohammed al Baqari]

Ok. Please open a case with your ITSP to find out why they aren't responding to 200OK messages. There is nothing to be done from your side now.

Just for me as a last test, can you please apply this command and share the debugs before you open a case with your ITSP.

voice service voip
 sip
  bind control source-interface g0/2
  bind media source-interface g0/2

[Paul Austin]

OK so what can I say except many thanks for your assistance - this SIP stuff is pretty complex and NOT as straight forward as 1st believed.

[Mohammed al Baqari]

You are welcome. I am sorry for not resolving this.

However, please keep us posted on the forum on how things goes with ITSP and we will be here to assist.

[Vivek Batra]

Hi Mohammed,

Nice job done so far despite of output result.

Would you like to suggest Paul to add following commands under dial-peer 5 as it's the dial peer matching incoming calls (seems 200 OK is being sent using gig 0/1 and we want to send it using gig 0/2);

voice-class sip bind control source-interface GigabitEthernet0/2
 voice-class sip bind media source-interface GigabitEthernet0/2

Although this dial-peer will be matching calls from CUCM too however if it works, we can separate out and have two different dial-peers matching incoming calls from CUCM and service provider.

Thanks

[Mohammed al Baqari]

Thx Vivek. Nice comment. 

Unfortunately the problem not yet fixed. 

I asked Paul to apply the bind command globally for the same reason you mentioned bit didn't work. 

Will interesting if it works per dialpeer

[Paul Austin]

also, if you look into the debug trace, the 200 OK message is not passed back to the ISP following the rining messages.