similar to: Asterisk BlindTransfer behaves differently in version 1.0 and 1.2

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

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

We deployed a PABX in China, orginally it used Netcom????'s E1, the zaptel.conf is as following: span=1,0,0,ccs,hdb3,crc4 bchan=1-15 bchan=17-31 dchan=16 loadzone=cn defaultzone=cn However, recently customer changed to use China Telecom??????'s E1, it always show REC, RED/REC, RED, cycling alarm when I run zttool in console. They sometimes still can make call, but the quality was quite

Hi everybody, I'm not a programmer, so I really don't know this, but is it possible to somehow backport ${BLINDTRANSFER} variable functionality to 1.0.X versions of *. I need this really badly ... thanks for your replies. Ivan

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

Hi, When a SIP hardphone is transfering a call while ringing (caller and callee don't speak to each other) using phone's Transfer key, it seems BLINDTRANSFER remains empty. Though I can see a 302 MOVED TEMPORARILY message coming in. Is there a work around or something obvious I'm missing (it's the first time I'm playing with Dialplan transfert features. context mylocal {

I've found on wiki that there is a variable called ${BLINDTRANSFER} which should contain the channel (or a number) of user that made a blind transfer of a call to another extension. Also I've found a patch for chan_sip to add support for ${BLINDTRANSFER}, but it's not working at all (chan_sip crashing), so I guess it is intended for CVS-HEAD version. Has anyone tried to backport it

I am having issues with Blind Transfer on asterisk 1.8 If I call from one Grandstream phone to another and us the transfer key to do a blind transfer everything works fine. When calling in on a sip trunk and then trying to use the transfer key to transfer from Grandstream phone to Grandstream phone the call just hangs up. It did not do this on Asterisk 1.4.x or 1.6.2.x . If we use

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

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 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

? 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

http://voip-info.org/wiki/view/Easy+PABX With Easy PABX you can create your own virtual PABX online in just minutes. Easy PABX is based on Asterisk and best of all - it's completely free. Regards thorben.dk

I am installing one asterisk, to establish connection with my PABX Siemens, in ISDN, link went up normally, also I obtain to internally call the branches the PABX, normally, but when I try to dial for the PSTN, through pabx with the command exten = _ 19xxxxxxxx, 1, dial(zap/g2/${EXTEN}, 30) asterisk, reports me the following error: -- Executing Dial("SIP/8110-a729",

Hi I am in the middle of move a small business over from legacy PABX + PSTN lines to VOIP infrastructure. I borrowed a spa9000 to place between the PABX and the PSTN lines. I have had this going for a while (>5 months) and it has been working fine (some issues with echo and other minor things), which is why I am moving to asterisk. I bought a tdm410 with 3 fxo + fxs. The fxs is connected to