search for: terisk

> -----Original Message----- > From: Andrew Kohlsmith [mailto:akohlsmith-asterisk@benshaw.com] > Is the channel physically being hung up before the * tone is heard? Good question. If it is, Asterisk doesn't detect it -- the PBX doesn't support Kewlstart-style disconnect notification. The sequence I hear on the extension, when I plug in an analog phone, is the cl...

...Local/4444 at office > &Local/5555 at office&Local/6666 at office,d) > same => n,Hangup() > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk. > org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/...

...t=3B cheers=2C<br>&gt=3B Paul.<br= >&gt=3B <br>&gt=3B --<br>&gt=3B ___________________________________________= __________________________<br>&gt=3B -- Bandwidth and Colocation Provided b= y http://www.api-digital.com --<br>&gt=3B New to Asterisk? Join us for a li= ve introductory webinar every Thurs:<br>&gt=3B http://www.as= terisk.org/hello<br>&gt=3B <br>&gt=3B asterisk-users mailing list<br>&gt=3B= To UNSUBSCRIBE or update options visit:<br>&gt=3B http://lists.digium.c=...

<snip> Replaces: pickup-9582-c0a80101-d-4 at 192.168.101.102 <snip> This Replaces header refers to RFC3891 which is not yet supported in Asterisk (see http://www.voip-info.org/wiki/view/Asterisk+SLA) This INVITE fails with : <snip> chan_sip.c: Trying to pick up 7792 at subs <snip> app_directed_pickup.c: No target channel found for 7792. If I'm dialing *87792 instead of using BLF, then I'm entering the dialplan part in...

Same here Michael and the PocketPC version seems unaudible with any codec; early days trying that though. Simon -----Original Message----- >From: "Michael Van Donselaar"<mvand@neb.rr.com> >Sent: 22/04/03 04:10:24 >To: "asterisk-users@lists.digium.com"<asterisk-users@lists.digium.com> >Subject: Re: [Asterisk-Users] Xten - Free windows SIP client > >On Fri, 18 Apr 2003 08:35:08 -0700, you wrote: > >>I have tried the Xten this morning, the sound is very good! >> >...

I have an idea, tell me if this wouldn't work... I know it's really ugly, but it might help some people until we can get round robin DNS checks for peers... Since * does not do GetHostByName() again until you reload your config, and BroadVoice and I'm sure other sip providers are using round robin DNS, why not create 2 [<your server here>-out] contexts in sip.conf, and then in

I am trying to setup a couple of virtual pbx's off of my one may asterisk box. So far I have been able to segment most everything via the Dial plan. My only question/problem has to do with the # Transfer function. I had set up # Transfers prior to segmenting the dial plan, and I cannot remember how I was able to specify which context to use when the user presses #. I...

I'm trying to use OpenBTS with Asterisk. I have two phones that are connecting to OpenBTS correctly, but on the Asterisk side the phones can't call each other. I followed this guide: http://gnuradio.org/redmine/projects/gnuradio/wiki/OpenBTSSettingUpAsterisk I set up two phones in sip.conf and extensions.conf. In my SIP output I se...

...mly believe that the few problems we are having is indicative of bad config files. Let me explain what the issues are first: 1) Our agents for the main call center are responsible to make calls when they have not already received an ACD call. However it seems that if they make an outbound call asterisk is still routing inbound calls to them. The ACD call beeps at them via the call waiting features then if the agent does not answer the ACD call it logs the agent out. I am just trying to figure out how I can tell the system that the extension is busy. Should I be using the new replacements to in...

...uot;. 09 00 00 3d 37 54 68 65 20 4e 75 46 6f 6e 65 20 ...=7The NuFone 4e 65 74 77 6f 72 6b 27 73 20 48 2e 33 32 33 20 Network's H.323 43 68 61 6e 6e 65 6c 20 44 72 69 76 65 72 20 66 Channel Driver f 6f 72 20 41 73 74 65 72 69 73 6b 00 00 19 31 2e or Asterisk...1. 30 2e 30 20 28 4f 70 65 6e 48 33 32 33 20 76 31 0.0 (OpenH323 v1 2e 31 32 2e 32 29 00 00 00 c0 a8 01 6b 06 b8 00 .12.2)......k... 80 d4 bf de d3 8a d9 11 8e 4c 00 01 02 3f c2 76 .........L...?.v 00 5d 0d 80 07 00 c0 a8 01 97 80 36 11 00 30 d4 .]........

...h323.conf, sip.conf and ooh323.conf. If I am using gatekeeper (gnugk) then I am able to call successfully to h323 users using SJphone. And same for SIP users also. But when I disabled gatekeeper and trying to call using gateway with sjphone then every time whatever number I dial the call goes to asterisk and some computerized information regarding asterisk is coming. I am putting my h323.conf and ooh323.conf h323.conf ; The NuFone Network's ; Open H.323 driver configuration ; listenAddress=10.142.17.68 listenPort=1720 connectPort=1720 ;TCP tcpStart=10000 tcpEnd=20000 ;UDP udpStart=10000...

First, here's what I want to do / what I have: X100P and a Quicknet PhoneJack. I want to be able to pick up the analog phone (connected to the phonejack) and dial another computer (with the same hardware) or just make a regular phone call which will be decided by asterisk depending on the phone number dialed. I know that this won't be taking full advantage of asterisk, but I'm just trying to get a connection before I go into deeper configuration. So far, I have asterisk running, and I get a dialtone when I pick up the phone ... however, when I dial a di...

Hi .. I've just received in the post my Wildcard card with a single FXO and three FXS daughter cards. I've identified a dedicated PC to function as the * machine and installed the card. I've installed Fedora Core 2 on that machine. I've downloaded the * software and the zaptel drivers. And now, to be quite honest, I haven't got much of a clue what to do next! I've

...locations I would also run *, and hook it up to an extension on an existing PBX. Excuse the complete newbie question, but how many 'wires' do I need to bring between the PBX and the * box to support multiple simultaneous calls? These calls would come from any extension on the TDM pbx to asterisk to the call center. In a typical scenario there would NOT be a lot of simultaneous calls unless the system we're supporting went down hard. How would / could? one configure * at the remote location to communicate with * at the call center? How would / could? one configure * at the remote loc...

Hi All, I am trying to get Asterisk up and running on my new Mandrake 9.1 install. I've installed Linux in the "standard" mandrake security mode, and "su" to do my attempts at install. I managed to obtain the source from CVS, and have been able to compile Zaptel. I then ran insmod zaptel, and also make con...