A trace of the whole call setup to hangup would be very helpful On Tue, Oct 6, 2009 at 8:31 AM, Peter den Hartog <[email protected]>wrote:
> > Thanks alot for you reply. > > Asterisk is used because we have some agi stuff happening on incomming > calls. The sip trunk is registered on Asterisk. If i dial out, opensips > uses > Asterisk because the extention is not in opensips (if i understand it > correctly) then Asterisk just uses his own sip trunk to dial outside. > > But for me it would be fine to use Opensips directly to make the connection > with the sip trunk, we can leave asterisk out for now. > > 1. There is two way audio, i can hear the other person talking, and he can > hear me 2. > 2. no reinvite, i see a ok, and then a bye > 3. i don't know this yet, i can test it, i think i saw a empty ACK > > > > Brett Nemeroff wrote: > > > > I guess the question here is, what is asterisk doing for you? I > personally > > would prefer the sip trunks right on opensips.. Asterisk is a kinda funny > > bottleneck in your architecture unless it's acting as some sort of media > > server (or TDM gateway). > > Some potential issues: > > 1. Do you have 2 way audio, some providers (gateways) will disco the call > > if > > there is one way audio for X seconds. > > 2. Do you see any reinvites happening? Some providers will re-invite > calls > > after they are up and if the reinvite fails, it will tear down the call. > > 3. Where is the BYE coming from? Do you see any other signaling after the > > 200OK/ACKs you get? Do you see retransmissions of either the 200OK or > ACK? > > If the signaling indicating the call was connected doesn't finish a > proper > > ACK in both directions, the call will likely get hung up on. > > > > > > On Tue, Oct 6, 2009 at 8:17 AM, Peter den Hartog > > <[email protected]>wrote: > > > >> > >> I'm trying to intergrate opensips with a allready running Asterisk > >> server. > >> The two servers are both on the same machine. > >> > >> I can recieve calls fine, Asterisk send them to my opensips > installation, > >> and the opensips forwards the phone call to the right user. I can call > >> between the users on the network, with out any issue's so far so good. > >> > >> I have a sip trunk registered on Asterisk, and i use that for my in and > >> outgoing calls. > >> > >> But when i make an outside call, the call ends after 17 seconds. Looking > >> at > >> the sip messages i see that i recieve a bye, then the call is gone. > >> > >> Am i doing something wrong, should the sip trunk be directly in > opensips? > >> and add that as a rewritehost? Or is this an Asterisk issue? > >> > >> My opensips is running on port 5090 (so are the phones) and my > >> asterisk+outside trunk is on 5060. > >> -- > >> View this message in context: > >> > http://n2.nabble.com/17-sec-recieve-a-bye-and-a-hangup-tp3774964p3774964.html > >> Sent from the OpenSIPS - Users mailing list archive at Nabble.com. > >> > >> _______________________________________________ > >> Users mailing list > >> [email protected] > >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > >> > > > > _______________________________________________ > > Users mailing list > > [email protected] > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > -- > View this message in context: > http://n2.nabble.com/17-sec-recieve-a-bye-and-a-hangup-tp3774964p3775048.html > Sent from the OpenSIPS - Users mailing list archive at Nabble.com. > > _______________________________________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-bin/mailman/listinfo/users >
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