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Hi, I am looking for a very low cost way of receiving and sending T38 fax reliably. Is there any possible solution using Asterisk as the PSTN SIP gateay and Digium E1/T1 card? Is there other open source package that can help to accomplish this purpose? Regards, Mark -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi guys! We have a call centre that has been moved across from an old Ericsson MD110 PABX to an Asterisk server with those in the call centre using X-Lite as their softphone. I'm trying to get Agent presence configured so that X-Lite gives the operators a visual indicator of their status - logged on, off and on "pause". I'm using chan_agent for the agents, so agents are

I have a DID from IPKall.com which is forwarded to my asterisk box. Then this extension should call my ip phone using Dial application. Everything works fine, except when I pickup the phone, I can talk, the other party can hear me, but I cannot hear anything the person says on the ip phone. Then after a couple of seconds, the call hangs up. I don't know why. Here is the message I get:

Here are some very small header patches to make header dependencies less visible. If other developers would like to use cxref to produce cross-referenced html source, I can post a Makefile to do this. Sorry about not being able to send it as an attachment. You can also get it at http://216.164.32.243/~kruus/vorbis/ for now, but my net connection is slow. 'cxref' suits my purposes,

Yell at me if you will, but I hate func_realtime - it's not very usable nor is it change-friendly (update your database and your dialplan completely breaks). I'm getting a new 1.6 box built out and working, and wanted to emulate the functionality of APP_realtime somehow, so I started digging around in the func_realtime source - here's what I came up with: For 1.6.0, look at line 86

asterisk-users-request@lists.digium.com wrote: > Date: Sun, 22 Apr 2007 19:38:04 +1000 > From: Rob Hillis <rob@hillis.dyndns.org> > Subject: Re: [asterisk-users] Softphone that supports central > provisioning? > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <462B2CFC.50709@hillis.dyndns.org> > Content...

hi, there is T.38 fax gateway for asterisk http://bugs.digium.com/view.php?id=12931 please test it and report bugs for people from http://www.voip-info.org/wiki-Asterisk+T.38+Bounty if you still want donate t.38 development please contact me at cervajs at fpf.slu.cz --------------------------------------- Marek Cervenka =======================================

Hi guys! I'm having one or two issues with queues hosted by an Asterisk machine where the clients are on a legacy PABX - at least for the interim. I fully expect most of these issues to be non-resolvable, but thought I'd at least ask to find out if there is some way of working around the issues. The legacy PABX is an NEC 7400 ICS connected to Asterisk via an E1 ISDN link. Calls are

Hello users, Can you share with me your experiences with Asterisk 1.6? Is it stable enough for commercial service? Thanks. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080901/d849d412/attachment.htm

...re off by one hour. I double checked the times on the server and they are correct. I also noticed that the files that are off by and hour appear to fall within daylight savings time. Is this a common problem with SAMBA and Windows, or is there something I have configured incorrectly? -- Richard Hillis 23 Walnut Knls Canton, MA 02021 /phone/ 781-562-1374 /fax/ 781-562-1374

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi guys, I've been working on a little dialplan fragment for roaming extensions, however the customer wants us to set the MWI indicator for the roaming extension that has just logged in. We're using MySQL realtime, so I've figured out that RealTimeUpdate will happily update the realtime database with the correct mailbox. My problem

Good day Im look at http://www.voip-info.org/wiki-Asterisk+GUI And I see there are a few GUI for asterisk What do you guys prefer? What is the best and simplest? Id like something that give me access to backend for a little bit of customization Thanks for you help and time -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello all, I need to conect an Asterisk with an Alcatel OmniPBX 4400 using an E1 port. It is the first time I make this kind of connection and I do not know exactly how to get it working. Someone has experience with this kind of connection? Could you paste a zapata.con and zaptel.conf files with QSIG configuration? Any clue will be wellcomed. Thanks Voipcrazy -------------- next part

Hi, I've noticed that sometimes instead of getting a regular ring tone when calling out on a Zap channel, I get this obnoxious loud noise which forces me to hang up. Is this a problem in the Zaptel driver? I seem to recall that ringback tones are generated by zaptel when dialing out from a SIP phone over a Zap trunk. Thanks.

Hi All When I dial a number it reaches the asterisk switch as abc at xyz@123.com what I need to do is to parse the abc and send it to my pstn gateway as in exten => _.,2,Dial(SIP/${EXTEN}@pstn.gw) this does work but I do have a varying number of numbers before the @ exten => _.,1,Dial(SIP/${EXTEN:0:12}@pstn.gw) Well can I use some kind of regular expression to take all numbers before

I have followed all the install note for A2billing and have everything installed and configured and my asterisk works except the callingcard application. Added the following [callingcard] ; CallingCard application exten => 777,1,Answer exten => 777,2,Wait,2 exten => 777,3,DeadAGI,a2billing.php exten => 777,4,Wait,2 exten => 777,5,Hangup I am using 777 as the calling card

Hi, I'm receiving this : [Aug 28 02:25:16] NOTICE[5895] chan_sip.c: Received SIP subscribe for peer without mailbox: 9163 I've read this : http://lists.digium.com/pipermail/asterisk-users/2008-May/211701.html I typed this: asterisk -rx "reload" asterisk -rx "voicemail show users" ... and got : default 9163 john doe 0 This "default

Dear, I'm having problems with the configuration of this gateway(GrandStream GXW 4108), I used the instructions from GrandStream but it doesn't work. Someone has a good configuration for this gateway? Thanks in advance, Nelson -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I'm running asterisk realtime, i had prob when a user does not unregister properly. I tested with SPA942 and a PAP2, when phone is registered, i call using the SPA using x-lite no problem, but when i unplugged the power, it does not unregister properly, so asterisk think SPA942 is still registered, when i call using x-lite, asterisk tries to call it.so it gets stuck at [Aug 11

Hello, I am having some troubles with Snom phones and maybe someone can help me. Let me say this: BLF and pickup works great with Polycomes and Grandstream etc... So I think my problem might not be Asterisk related but I am not 100% sure. The snom phones subscribe to my extensions (hint priority) as expected. The light blinks (ringing) or is turned on (in the call) as expected. My problem is to