SIP trunking with not working

This should be a relatively simple config, but I can’t get it working.

I have EdgeWater VOIP gateway -> Asterisk

Using config from

I am testing outgoing calls with

exten => 555,1,Dial(SIP/trunk-out/+CELLPHONE)

Outgoing calls terminate after 23 seconds exactly every time.

Incoming calls give disconnected message / signal. Nothing happens in asterisk console during incoming calls.

OK… so where does fit in the picture? You might try the following Dial() format to see if this works better. Not sure what your version actually generates in it’s INVITE.

exten => 555,1,Dial(SIP/+CELLPHONE@trunk-out,60) 

I assume CELLPHONE is actually a 10 digit number, correct? If you are using, I believe they expect their INVITE in +1NPANXXXXXX format. is the provider of the SIP trunks. CELLPHONE in my rule is an actual number in the proper format.

Thanks for the idea, but it doesn’t help. Same problem. engineers tell me that my PBX is issuing a second INVITE when the call cuts off. Old PBX problem?

Since I have been backwards and forwards with the configurations they advise and I am not finding any information about this kind of configuration I assume the whole process is something simple that is just broken. I am going to recreate my server from scratch and see if that fixes it. If not, I will have to pay digium for support.

Doesn’t just about everyone have to do this with Asterisk? Why isn’t there documentation?

If you are willing to give me SSH into the server I could take a look. I do about 2 million minutes/month over the network, so I am pretty familiar with their gateways and behaviors.

actually… just thought of something. Based on your complaint, I am going to jump to a conclusion and guess you have a firewall and/or NAT issue.

Are you sure UDP port 5060 is making it to your asterisk server? It sounds like it might not be. Perhaps 23 seconds is the default timeout on a Dial() if no ACK is received. Certainly the dead air + no console makes me think the SIP INVITE is not making it to your system.

Is your server NAT’d? If so, can you confirm you are passing through port 5060 UDP on your router?

If your server is not NAT’d, can you confirm your firewall is allowing traffic on UDP 5060 from the SIP gateways?

I have completely disabled my firewall with no changes. The gateway device has a public and private IP and is VOIP aware so traffic should make it through okay. My firewall logs don’t show any blocked traffic either.

I would love to give you SSH access if you have time. I am still working on building a new server and almost have that up for testing as well. ctarbet at wssd-k12-id-us

Ougoing issue resolved by upgrading to latest Asterisk 1.6 on Fedora 11.

Incoming resolved as well after some configuration changes on the new server.

Hello Ctarbet,

I got your post by searching for what I am looking. I am looking for with Asterisk. Can you please post the working config of sip.conf or extensions.conf files. It would be really great if you post here or viewtopic.php?f=1&t=89406

Thanks in advance.


We’ve since left and I don’t have the config files anymore. :cry:

They had poor support and blocked telephone conference numbers like WebEx.

Thank you :smile: