A trace of the whole call setup to hangup would be very helpful<br><br><div class="gmail_quote">On Tue, Oct 6, 2009 at 8:31 AM, Peter den Hartog <span dir="ltr"><<a href="mailto:peterdenhartog@gmail.com">peterdenhartog@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;"><br>
Thanks alot for you reply.<br>
<br>
Asterisk is used because we have some agi stuff happening on incomming<br>
calls. The sip trunk is registered on Asterisk. If i dial out, opensips uses<br>
Asterisk because the extention is not in opensips (if i understand it<br>
correctly) then Asterisk just uses his own sip trunk to dial outside.<br>
<br>
But for me it would be fine to use Opensips directly to make the connection<br>
with the sip trunk, we can leave asterisk out for now.<br>
<br>
1. There is two way audio, i can hear the other person talking, and he can<br>
hear me 2.<br>
2. no reinvite, i see a ok, and then a bye<br>
3. i don't know this yet, i can test it, i think i saw a empty ACK<br>
<div><div></div><div class="h5"><br>
<br>
<br>
Brett Nemeroff wrote:<br>
><br>
> I guess the question here is, what is asterisk doing for you? I personally<br>
> would prefer the sip trunks right on opensips.. Asterisk is a kinda funny<br>
> bottleneck in your architecture unless it's acting as some sort of media<br>
> server (or TDM gateway).<br>
> Some potential issues:<br>
> 1. Do you have 2 way audio, some providers (gateways) will disco the call<br>
> if<br>
> there is one way audio for X seconds.<br>
> 2. Do you see any reinvites happening? Some providers will re-invite calls<br>
> after they are up and if the reinvite fails, it will tear down the call.<br>
> 3. Where is the BYE coming from? Do you see any other signaling after the<br>
> 200OK/ACKs you get? Do you see retransmissions of either the 200OK or ACK?<br>
> If the signaling indicating the call was connected doesn't finish a proper<br>
> ACK in both directions, the call will likely get hung up on.<br>
><br>
><br>
> On Tue, Oct 6, 2009 at 8:17 AM, Peter den Hartog<br>
> <<a href="mailto:peterdenhartog@gmail.com">peterdenhartog@gmail.com</a>>wrote:<br>
><br>
>><br>
>> I'm trying to intergrate opensips with a allready running Asterisk<br>
>> server.<br>
>> The two servers are both on the same machine.<br>
>><br>
>> I can recieve calls fine, Asterisk send them to my opensips installation,<br>
>> and the opensips forwards the phone call to the right user. I can call<br>
>> between the users on the network, with out any issue's so far so good.<br>
>><br>
>> I have a sip trunk registered on Asterisk, and i use that for my in and<br>
>> outgoing calls.<br>
>><br>
>> But when i make an outside call, the call ends after 17 seconds. Looking<br>
>> at<br>
>> the sip messages i see that i recieve a bye, then the call is gone.<br>
>><br>
>> Am i doing something wrong, should the sip trunk be directly in opensips?<br>
>> and add that as a rewritehost? Or is this an Asterisk issue?<br>
>><br>
>> My opensips is running on port 5090 (so are the phones) and my<br>
>> asterisk+outside trunk is on 5060.<br>
>> --<br>
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><br>
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