There's a setting, I think it's "canreinvite=no" in sip.conf that prevents asterisk from dropping out of the media path. It's an issue with calling card apps, that's where I've seen the discussion before. You'll probably find details on the wiki.
Dave On 1/16/06, Liviu Toma <[EMAIL PROTECTED]> wrote: > Thanks for the response, Dave. > I've noticed that on many of the calls my asterisk says something > "attempting native bridge between X and Y". > Does that mean that it tries to connect the 2 peers without staying in > between ? It does that with SIP calls as well as calls to PSTN. > If that's the case, is there a way to disable this ? This used to give me a > lot of trouble back when I was using wildcard clone modems, when I called in > from PSTN and dialed an extension that was being forwarded back to PSTN > through another modem, it would do a native zaptel bridge for the 2 calls > and even if I hung up the 2 calls, the modems would never drop the line. I > never found a solution to this, but I stopped using modems eventually. > Just FYI, the connection between Asterisk and the Cisco router is pretty > simple, the Cisco can do SIP and has all the popular codecs built in. > However, it doesn't seem to be able to register or accept registrations by > itself (I think it can do that only in conjunction with a Cisco Call Manager > server), so pretty much the peers connected to it need to have fixed IP > addresses and that's how they get authorized (there no authentication, or, > if there is, I haven't been able to find how to use it). Anyway, the reason > I've been using it was that I had the router already, and the PRI card cost > me a lot less (~$120) than it would cost for a PRI card for Asterisk, and it > seems to work really well. > > Thanks again, > Liviu > > ----- Original Message ----- > From: "Dave Donovan" <[EMAIL PROTECTED]> > To: "Liviu Toma" <[EMAIL PROTECTED]> > Sent: Monday, January 16, 2006 10:01 PM > Subject: Re: [on-asterisk] No sound on either inbound or outbound > > > I don't know much about the specifics of connecting Asterisk to a > Cisco box but I'll tell you the first two things that come to mind. > > 1) Firewall: It's obvious, you've probably already checked. It also > doesn't sound likely since it works in some scenarios but not others. > > 2) Reinvite: check and make sure that your asterisk box is staying in > the media path. After the call is nailed up, do a "sip show channels" > and see if your asterisk box is still part of the call. It's possible > that the SIP client is trying to talk directly to your Cisco box and > that's why the behaviour differs when you send it to an internal > phone. In the case of the internal phone, Asterisk would stay in the > media path. > > Good Luck, > Dave > > On 1/16/06, Liviu Toma <[EMAIL PROTECTED]> wrote: > > Hello all, > > > > I have a problem that is giving me headaches for quite a while. > > I have an Asterisk server running at one location, registering to > > different > > SIP providers. The server has a routable IP address, doesn't use NAT. > > In a different location I have a Cisco router with VoIP capabilities, > > connected to the PSTN through a PRI. The Cisco router also has a routable > > IP > > address. > > In my Asterisk server, I am using the Cisco for connecting to the PSTN. > > Basically I want all inbound calls from different SIP providers forwarded > > through the Cisco router to the PSTN and then my swll phone. > > Now here's the problem: let's say my Asterisk is connected to SIP provider > > X. If I call my account from another account of the SIP provider X, > > everything works OK, I get the call forwarded to my cell and the sound > > works > > both ways. But it somebody calls my account at SIP provider X from a > > regular > > PSTN line, at least one of the ends cannot hear the other (or maybe > > doesn't > > work both ways). > > If instead of forwarding the calls through my Cisco router I send them to > > an > > internal extension where I use a Sipura adapter, the sound is OK in both > > cases. > > Any idea where to start troubleshooting this ? > > > > Thanks, > > Liviu > > > > > > --------------------------------------------------------------------- > > To unsubscribe, e-mail: [EMAIL PROTECTED] > > For additional commands, e-mail: [EMAIL PROTECTED] > > > > > > > -- > David Donovan > Consultant > Fulcrum Solutions > > -- David Donovan Consultant Fulcrum Solutions
