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Twilio incoming call fail on sip trunk on cucme

o0chris0o
Level 1
Level 1

Hi,

I've been trying to resolve this problem for a few days now (I'm starting in the cisco voip). I set up a Twilio elastic SIP trunk, I'm able to make outside call, but the incoming calls are drop, when I call the number, it simply said that the number is not assign. But the call did reach my system, when I run the debug, I see it coming through but I get a SIP 404 not found with reason #1 (number not allocated). Before going further, here's my set up, Twilio--->CUCME (cisco router 2811)---->CUE (auto-attendant-202)---->SCCP extension.

 

The way I setup the call routing is when I call the main number (5145551212) the call is supposed answered by the AA at 202 then from there, the user choose an option then is transfer to either a huntgroup or a DN.

 

When I look at the logs I find out first that when it reach my translation-rule (/15145551212/ /202/) it actually translate it to +202, which I think is the problem. So I went on and read about it and I was able to get rid of the (+), but it goes nowhere from there, the rule to strip the (+) is apply but it doesn't go to 202. Then the call get stuck there so I don't know how to push the call through another translation-rule/profile to finish the leg.

 

Can you guys look at my config and point me where I might have screwed up?

1 Accepted Solution

Accepted Solutions

Ayodeji Okanlawon
VIP Alumni
VIP Alumni

Hi there,

First of all you only need one inbound dial-peer in this scenario. So I suggest you remove dial-peer 101

no dial-peer voice 101 voip

Now, the translation profile applied to dial-peer 100 should look like this:

voice translation-rule 4
rule 1 /^\+5145551212/ /202/

 

If you want all calls coming to go to AA on 202, then you can use this

/^\(.*\)/ /202/

 

Please rate all useful posts

View solution in original post

3 Replies 3

Ayodeji Okanlawon
VIP Alumni
VIP Alumni

Hi there,

First of all you only need one inbound dial-peer in this scenario. So I suggest you remove dial-peer 101

no dial-peer voice 101 voip

Now, the translation profile applied to dial-peer 100 should look like this:

voice translation-rule 4
rule 1 /^\+5145551212/ /202/

 

If you want all calls coming to go to AA on 202, then you can use this

/^\(.*\)/ /202/

 

Please rate all useful posts

Hi Ayodeji,
Thank you very much! It worked! Can you point me to a good ressource to learn more about translation rules.
Thanks
Chris