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

Asterisk Asterisk Support
1

I have Asterisk 1.8.10.1 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@207.227.228.29,300”) in new stack
[09:25:33] == Using SIP RTP CoS mark 5
[09:25:43] – Called SIP/1814449XXXX@207.227.228.29
[09:25:45] – SIP/207.227.228.29-00000089 is ringing
[09:25:45] – SIP/207.227.228.29-00000089 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.

2

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.

3

David55,

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.

4

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.