Steve
2005-Jun-01 16:36 UTC
[Asterisk-Users] Newbie Question: HOWTO make outgoing call on SIP account from internal extensions?
Over the past 2 weeks I have been able to compile and get an asterisk system up & running on a debian Linux box. I have setup 5 internal sip clients on the lan and all works great! I can also call from outside (PSTN) into the system and reach extensions and services no problem. All is up & running behind a nat firewall with proper ports forwarded and locked down on each device to work properly. For outbound calls to the PSTN I'm using a single SIP provider account... I have read LOTS of docs and played quite a bit to get this far.... The only thing I have not been able to figure out is how to set it up so the internal extensions can actually make an outbound call via our SIP provider account. I've very confused by the docs I have read and some of them pertaining to this matter are not perfectly clear (to me) on if what they are doing is what I think they are doing. I'm probably totally off here and doing the wrong thing because whenever I try it My system no longer even works with the sip provider at all and won't register... -------------- Here's what works: register => 2135551212:2135551212@sipproviderexample.com -------------- Just having that single line in place allows incoming calls to everything and works flawlessly but please forgive me for asking..... register => 2135551212:2135551212@sipproviderexample.com/2200 ALso works perfectly and rings in to extension 2200 How in the world to you make it ring into more than one extension? :-) ---------------------------- OK here's what messes it all up (and I admit I'm clueless here) register => 2135551212:2135551212@sipproviderexample.com [sipproviderexample.com] type=peer host=10.77.77.133 fromuser=2135551212 secret=2135551212 fromdomain=sipproviderexample.com adding this secttion breaks it and I really do not understand what it's even for... does it work with the register line somehow? or is it totally seperate? what is it for? All the docs I have looked at seem to suggest adding this extra section but do not really seem to explain it or what exactly it does. I'm not sure what it's for or if it has anything to do with making outbound sip calls from the internal extensions. when I add it my sip provider account stops working and I get registration retries and timeouts without any successful registrations after that. I'm just looking for a good pointer in where to go for an example of how to use my provider account for outbound connections... I understand the dialplans themselves but do not know how to associate them with the actualy sip provider account for an outbound call. Thanks.... I'll keep reading until I figure these things out but any pointers to specific documentation that answers any of these questions would be very much appreciated.... I *think* I am familiar with just about all of the standard asterisk documentation I have been able to find... More than likely I am just missing some key points that I have read but have misinterpreted! Take care! Steve
Ronald Wiplinger
2005-Jun-01 17:20 UTC
[Asterisk-Users] Newbie Question: HOWTO make outgoing call on SIP account from internal extensions?
Steve wrote:> > I have read LOTS of docs and played quite a bit to get this far.... >Good, keep playing!!! (a lot of your typing time deleted)> > ---------------------------- > OK here's what messes it all up (and I admit I'm clueless here) > > > register => 2135551212:2135551212@sipproviderexample.com > > [sipproviderexample.com] > type=peer > host=10.77.77.133 > fromuser=2135551212 > secret=2135551212 > fromdomain=sipproviderexample.com > > adding this secttion breaks it and I really do not understand what > it's even > for... >What does it mean for you, that this "breaks" it. Did it work before? What is your "jump in point" to the dial plan? You do not have a context= line. So you may jump into the context=default as usually mentioned in sip.conf [general] context=default ; Default context for incoming calls Do you have something in the dialplan like: [from-sipproviderexample.com] ; if you would use this as your context= in sip.conf, otherwise the next lines in the default context of your extensions.conf exten => _X.,1,Dial(SIP/601,60,tr) ; if any number calling from this provider should ring extension 601 or exten => 2135551212 ,3,Dial(SIP/601,60,tr) ; if only a dialed in number 2135551212 of your provider should dial extension 601 Keep playing with: context=xxxxx and [xxxxx] exten => something with and without starting _ Hope it helps, ... bye Ronald> does it work with the register line somehow? or is it totally seperate? > what is it for? > > > All the docs I have looked at seem to suggest adding this extra > section but do > not really seem to explain it or what exactly it does. > I'm not sure what it's for or if it has anything to do with making > outbound sip > calls from the internal extensions. > > when I add it my sip provider account stops working and I get > registration > retries and timeouts without any successful registrations after that. > > I'm just looking for a good pointer in where to go for an example of > how to use > my provider account for outbound connections... > > I understand the dialplans themselves but do not know how to > associate them > with the actualy sip provider account for an outbound call. > > > Thanks.... > > I'll keep reading until I figure these things out but any pointers to > specific documentation that answers any of these questions would be > very much > appreciated.... > > I *think* I am familiar with just about all of the standard asterisk > documentation I have been able to find... > > More than likely I am just missing some key points that I have read > but have > misinterpreted! > > Take care! > > Steve > > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > >-- Ronald Wiplinger (CEO of ELMIT) http://www.elmit.com +886 (0) 939--77-55-16 or FWD 511208 - I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org PS: Spam prevention! Our system is protected with a spam prevention program. If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again.