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Hi every all, since a few weeks I came back to asterisk and tried to install version 1.6. The installation went fine so I decided to buy new dids on Voxbone. I have added the sip peers of Voxbone Belgium1 like this in the sip.conf [81.201.82.39] host=dynamic type=friend insecure=very context=your_context canreinvite=no qualify=no deny=0.0.0.0/0.0.0.0 permit=81.201.82.39/255.255.255.255 but unfortunately when I receive a call I got this nice error: handle_request_invite: Failed to authenticate user "075XXXXXXXX" <sip:075XXXXXXXX at voxbone.com>;tag=76596. I am in doubt now because...

I've just started consulting for a SME client based in Reston Virginia. They don't know it yet but they are going to need a hosted asterisk service and some DID's. Email me if you are able to provide 10 DID's in Reston (must be able to be ported away!!) and hosted Asterisk with end user configurable IVR etc. Probably only 5-8 users at the moment BUT... they'll be

I've read through the archives and have picked up that * does not need a gatekeeper to talk directly with an H323 handset to send and receive calls. I'm trying to go PSTN----*-----H323 and all the examples that I can find use a gatekeeper. Are there any examples or hints for doing it without the gatekeeper? many thanks in advance Brian

Hi, I have a wav file recording that i want to use on my voicemail, how can i set this up? thanks!

Hi I'm trying to get the spa 8000 used with a fax machine using t38 passthru i have tried with 1.6.2 and 1.8.3 and is still a no go the provider i use is 012 in israel wich supports t38 (i use it with ffa) could anybody give me a clue how to get this working if it should t38pt is set to yes in sip.conf Thanks, Israel -------------- next part -------------- An HTML attachment was

Hi Timo.... I try to change my plugin to read the input email to increase a header line and one \r\n. But i need help to get this input.... My plugin is based in zlib plugin. In "static int zlib_permail_get_stream" function, have it. if (imail->data.stream != NULL || (_mail->uid == 0 && zuser->save_handler == NULL)) { return

Hello, I'm using Asterisk with Realtime extensions and ODBC CDR. And I'm have some trouble with the CDR userfield that is not changed when using the SET command in the realtime dialplan. In my dialplan (extensions.conf, the file) I'm setting the userfield like this : exten => s,n,Set(CDR(userfield)="X") Later, my dialplan switches to the realtime part and this is an

...ling > > List - Non-Commercial Discussion" > > <asterisk-users at lists.digium.com> > > > Date: Friday, August 15, 2008, 12:25 PM > > > Check if you have some rule to dial under brad1 > > context > > > > > > dialplan 91xxxxxxxxxxx at your_context > > > > > > Regards > > > > > > Felippe Silvestre > > > > > > > > > -----Original Message----- > > > From: asterisk-users-bounces at lists.digium.com > > > [mailto:asterisk-users-bounces at lists.digium.com] >...

...n and again for everyuser. Maybe someone else out there knows someting else that can help. Don't set many "outgoing" context for every user in sip.conf!!!!! just set one and point all users to that one. If you need your user to have acces to other contexts just add include => your_context at the end of whatever context you want (btw can add more than one inlcude's ) Alyed ------------------------------------------- Hi, Hope someone can help me-Asterisk isn't behaving as I would expect and I think it's down to my contexts. There are two things I can't fath...

I have some troublesome numbers that I would like to capture the SIP dialogue when I am calling them. When I am about to dial the number, is there any way to turn on SIP debugging in the dial plan before I make the call? (and turn it off after the call is completed?) -------------- next part -------------- An HTML attachment was scrubbed... URL:

I need help creating a channel for my phone device (Quicknet PhoneJack). I have installed and loaded the driver and phone devices listen in /dev (phone0 - phone15). [phone.conf] mode=dialtone format=slinear device => /dev/phone0 fxoks=2 ;Quicknet PhoneJack [extensions.conf] ... exten=>_NXXNXXXXXX,1,Dial,Phone/phone0 ... When I try to make a call, I get the following output: Executing