similar to: Do you encounter this REC alarm before?

Hi, all, Do you guys happen to know that there are any IP phones have such feature, that it can has some indication for the agent status linked to the phone? E.g some LED show the status, backend we can link the phone to one agent id, then the agent login the system, the 'online' indication will be blinking and on, if logout with type of meeting, then 'meeting' LED will be on,

Hi, all I am a pabx vendor from Singapore. Recently we are going to implement a failover solution for our customers using heartbeat, the asterisk server can failover perfectly, however the g729 codec canot work, because it is binded the mac address, we have bought two set of licenses, can you provide us some workaround for this scenario? Regards, Liangliang

Hi, we are using asterisk 1.2.13. When callbacklogin agent transfer call using SIP phone's transfer feature, he is always in busy status and cannot answer any more incoming call from queue until the transferee hang up the call. -- Regards! Liangliang

Recently, we just migrate our PABX from 1.0.10 to the latest asterisk 1.2.13, most of the features are migrated smoothly, only blind transfer behaves differently, and quite annoying, We use options 't' and 'T' in dial command to enable using asterisk feature to do blind transfer, in 1.0, after you dial the number and followed by a '#' key, asterisk will do the transfer

My MusicOnHold sound is very soft, but when I hear it directly from mp3 playe on desktop, the loudness is quite ok. Wonder whether there is any configuration to change the loudness of MusicOnHold. Regards, Liangliang

Another day, another apparant unexplained hardware incompatibility. I have a TE412P and a TDM400B living quite happily in a whitebox using an Intel motherboard: http://www.intel.com/design/servers/boards/se7230nh1-e/index.htm I tried to move to an IBM x3650 system. It uses a slightly newer chipset, but apparantly it's in the same family. The SE-7230 board has been EOL'd and the

I'm hoping people can suggest some ideas for debugging a problem that I'm having with DTMF. Unlike most of the DTMF problems reported here, it has nothing to do with Asterisk interpreting DTMF. My problem is with the synthesis of DTMF tones on outbound calls on a PRI connected to a TE412P card. I'm running * 1.4.10.1 with Zaptel 1.4.4. It is important to note that these problems

Hi, Recently we got a new feature request from our customer, they want a report to list the duration that agents putting customer on hold, they want to base on this to measure the agents performance. I cannot find any events in cdr, message logs, or manager interface, only when I enable sip debug, then I can see the ReInvite Event in the cli , some thing like the attached logs, is there any

Has anyone successfully run * 1.4 with the following configuration (or something very similar)? HP DL380 G5 (3Ghz Xeon) CentOS 4.5 (kernel 2.6.9-55) Asterisk 1.4.5 (or 1.4.4) Zaptel 1.4.3 (or 1.4.2.1) TE412P TDM400B (2x FXO and 2x FXS modules) I've had this rig running * 1.2.18 with Zaptel 1.2.17.1 for several months without any issues. Upon trying to upgrade to * 1.4.4 and Zaptel 1.4.2.1 a

OK let’s start out with this. I’m not a pro GUI designer… ? Now that that’s done. Welcome to OpenConf. At least that what we call it now. To config an * file click on the filename to the left. For my example use extension.conf. Now you’ll have a FULL text editor and a parsed list of all the [sections] in the extensions.conf file on your left. On the right you will find any numbered var’s

The Gestalt System project aims to develop a free, Open Source, object-oriented software framework which makes use of other free, Open Source software tools for the purposes of managing, analysing, and reporting on quantitative data in a manner which is suited to modern, networked computing environments in which processor, memory, and storage resources are abundant but time is scarce and

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,

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, I am about to purchase an Eurotech PRI E1 GSM & CDMA Gateway to operate with my Asterisk's TE412P interface. Anyone here has any experience of having this combination? Any success or failure stories would be greatly appreciated. Thanks in advance. Ash

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

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 =>

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

Hi, I was able to implement T122p one port PRI and was able to call out, but I am planning to use TE412p (includes echo cancellation) 4 port digital card (PRI), I wanted to know can asterisk support 3 four port PRI cards (12 PRI connections) with proper hardware like dual core quadcore processor and 8gb RAM in one server? Also I was planning to implement using 64 bit architecture with Asterisk:

? 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, 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