Hi, We have set up our Asterisk server, our extension.conf and sip.conf according to http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html?page=4 It's quite basic, and extension.conf is set up properly. The difficulty we are now encountering is in sip.conf, in trying to get any softphone to register at our own Asterisk server. We have searched the mailing list, and find bits and pieces of information that kind of help, but are very much not complete. The successes we've had include: - registering extensions to Free World Dialup. Once these were registered we could make calls to our fwd numbers and have them routed through to our asterisk server. This would put us through to voicemail. - have Asterisk route calls to a softphone. Essentially, we just set up an extension in sip, hard coded the IP and Asterisk would route a call from the Free World Dialup to this extension. The softphone receives the call and can communicate somewhat. This sounds like it works nicely, but it is still lacking. Our problem: The problem is this... Asterisk routes the call to the hard coded IP. The softphone is just sitting there listening for calls, but not registering with the Asterisk server. Also, we cannot make calls from one extension to another through our own Asterisk server. The questions: 1) How do we set up specific softphones to register to the Asterisk server and make calls to extensions through the Asterisk server? The softphones we are trying include: SJPhone, Kphone, Linphone. If you know of any others that work for you we'd sure like to know. 2) How do we configure sip.conf to do this? If you have this working for you, please give us specific details concerning the setup of your sip.conf and especially the corresponding set up for your softphone. We're very new to Asterisk and would very very much appreciate the help. Thanks, Tim & Rika _________________________________________________________________ MSN 8 with e-mail virus protection service: 2 months FREE* http://join.msn.com/?page=features/virus
>Hi, > >We have set up our Asterisk server, our extension.conf and sip.conf >according to >http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html?page=4 >It's quite basic, and extension.conf is set up properly. The >difficulty we are now encountering is in sip.conf, in trying to get >any softphone to register at our own Asterisk server. > >We have searched the mailing list, and find bits and pieces of >information that kind of help, but are very much not complete. > >The successes we've had include: > >- registering extensions to Free World Dialup. Once these were >registered we could make calls to our fwd numbers and have them >routed through to our asterisk server. This would put us through to >voicemail. > >- have Asterisk route calls to a softphone. Essentially, we just >set up an extension in sip, hard coded the IP and Asterisk would >route a call from the Free World Dialup to this extension. The >softphone receives the call and can communicate somewhat. This >sounds like it works nicely, but it is still lacking. > >Our problem: > >The problem is this... Asterisk routes the call to the hard coded >IP. The softphone is just sitting there listening for calls, but >not registering with the Asterisk server. Also, we cannot make >calls from one extension to another through our own Asterisk server. > >The questions: > >1) How do we set up specific softphones to register to the Asterisk >server and make calls to extensions through the Asterisk server? >The softphones we are trying include: SJPhone, Kphone, Linphone. If >you know of any others that work for you we'd sure like to know. > >2) How do we configure sip.conf to do this? > >If you have this working for you, please give us specific details >concerning the setup of your sip.conf and especially the >corresponding set up for your softphone. > >We're very new to Asterisk and would very very much appreciate the help. > >Thanks, >Tim & RikaThe examples in the onlamp article are sufficient to allow a standards-compliant SIP phone to register with your Asterisk system. This is really a question about your particular softphone, and not about Asterisk. I'd suggest grabbing a copy of the xten Xlite (http://www.xten.com/) phone and seeing if that registers successfully, as I have that one working quite well. For testing, make sure there are no NATs or firewalls in between your Asterisk server and your client. JT
hello, I am working with asterisk and looking for some stats about operators, then i've found that there is no real time of the call in asterisk when i use an autoattend context. looking into the cdr.c i can see that applications can call a ?set destination? or something to update the CDR record so you can know the real destination of the call, but i can't found something to make the apps(queue,dial, etc.) to update also the real time of the answer for that call. When the exten,s,1, is executed the answer time is setted and it remains that way, so if for example, a person dials a PBX, the autoattend starts telling him about the menu and the extensions and the person just dial an extension, lets say it took him 15 secs, then the Dial app is executed, for example it could be a queue app, and the extension start ringing, for lets say 5 sec and we have 20 secs so far and no real answer for that call, when anotehr person actually answers the call and they talk about 20 secs, the CDR will tell me that the specific call i'm talking about had 40 secs with 40 secs billables, when the real thing is that it was 20 secs what the real call last, i mean for real when a person actually gives attention to the caller. anyone has opinions? i think it could be very usefull, cause sometimes you need to know, for example, if operators are answering the calls for real or not, or if they just let it ring. with actual statistics.. i can't know that. thanks in advance.