Moving from T1 to SIP trunk causes 10s Delay after Dial

I have Asterisk running on a Dell PowerEdge server with a Digium T1 card. Everything has worked well for years, now we are changing Telco providers to a Level3 SIP trunk. Everything works as expected except for a delay of 10 seconds immediately after the Dial command [09:25:33], before the Called response [09:25:43]. The Asterisk box is directly on the Level3 private cloud, and Level3 doesn’t see the SIP request come in for them until I show the Called response [09:25:43]. I don’t see this same delay when another SIP device like a phone is called.

[09:25:33] – Executing [91814449XXXX@unrestrictedInternational-L3:3] Dial(“SIP/9476-00000088”, “SIP/1814449XXXX@,300”) in new stack
[09:25:33] == Using SIP RTP CoS mark 5
[09:25:43] – Called SIP/1814449XXXX@
[09:25:45] – SIP/ is ringing
[09:25:45] – SIP/ is making progress passing it to SIP/9476-00000088

I’m sure this is just some setting setting on my end that I’m not familar with because of my lack of working with SIP trunks.

My guess is that it is trying to resolve or reverse resolve an address and you have a DNS configuration that is timing out rather than returning “not found”.

Otherwise, you are going to have to turn up the debugging to give a better idea of the step that is causing the problem.


I have my Verbose at 15 Vs and I have nat=yes and canreinvite=no in the SIP config for the phone I’m using to test with. If I’ve read the docs on SIP.conf correctly, that’s what I should set to avoid a reverse DNS lookup.

The yes option on nat= is deprecated, and canreinvite may not even be valid, and I am not aware of anything in nat= that affects DNS lookups, and directmedia certainly doesn’t. 1.8 is, at best, on security fixes only, but may not fully have implemented the replacements for nat=yes.