similar to: SIP trunk to a Boscom/Claro/IP Gear Robocom

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

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

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:

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:

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

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

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, 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 =======================================

I'm having an issue with CDR. Basically, I expect to have all "legs" of a call having the same linkedid and differing only by the sequence value. That does happen, but I'm getting null dst values after doing an attended transfer. I'm not sure if this is a bug or I'm doing something wrong. I'm running Asterisk 13.2.0. Here's the console log, step by step: First,

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

Hi All, Can Asterisk dial extension which resides in the PABX? (eg. 2000) Sip Phone <-----> Asterisk <------> ATA (FXS) <------> (CO side) PABX <-----> Extension (eg. 1000) (2100 & 2101) can my sip phone call to pabx extension 1000? What will be my dial plan? I know I can connect to 1000 by

Hi All, I would like to explain the layout that i am trying to achive. I am so helpless on this regard. So here is the story ........ " This is with regard to the setup which you can find at the "Asterisk The Future of Telephony" , chapter 11, page # 196-197, I am attaching the picture for your information. Now I am taking a challenging step to of integrate IP PBX with our

-----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

Vinicius Fontes wrote: > Hey guys, I don't know if this is the right place to ask this. I was > thinking about reporting a bug, but maybe it's better to sort out if > this is really a bug or just me being lame. > > I want to record *every* call in my Asterisk box, so I use the > MixMonitor() application like this is my extensions.conf: > > exten =>

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:

All, I'm haveing a bit of trouble getting my head around H.323 and call routing with Gatekeepers, Zones and intra-zone calls - hopefully someone who is more informed in things H.323 will be able to point me in the right direction...? I already have a mature network of Asterisk boxes dotted around the UK and overseas with hundreds of extensions and our own number-plan/dial-plan in the form

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

? Dear all, We are connecting the PABX with Application server.What we are trying is that when a nbr 1800 (this is not registered in PABX) is dialled the pabx should route the call to Application server .The PABX should also have intelligence to route the call by itself for its registered clients.For this scenario to work please guide us what are the files we need to change and other necessary

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, We've been using zaphfc single ISDN cards as cheap Zaptel timing sources for our Asterisk boxes for a long time, and in the asterisk 1.0.x series, had zero problems doing so. I now have some boxes with Zaptel 1.2.x (with a mixture of 1.0.x asterisk and 1.2.x asterisk), and this setup no-longer seems stable - By plugging or unplugging the ISDN cable, and sometimes just randomly the card