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SIP CUBE dial-in problems

Paul Austin
Level 4
Level 4

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

24 Replies 24

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.

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

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

yep removed - see config.

 

Thanks again.

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

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.

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.

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

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

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

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