similar to: Extension No.s not being received correctly.

Currently I am creating .wav files and then converting them via SOX to .au file format, then running them through a gsm codec convertor which all works fine except that it sounds like the recording was made with a sock in my mouth !! Could someone in * land help me to get a good sound quality with gsm format. Thanks in advance. -------------- next part -------------- An HTML attachment

Hi, Does Icecast support streaming on demand as Shoutcast does? i.e. if I have some MP3 files in a directory, can these be streamed if I link to them on a web page. Thanks, Glen ----------------------------------------------------- Design Solution Limited t: +44 (0)1502 513008 f: +44 (0)1502 588622 e: info@designsolution.co.uk w: http://www.designsolution.co.uk Nouvotech

Hi, I have recently upgraded to Tiger and needed a way to launch the rsync daemon on system-startup. Tiger includes the launchd which is responsible for launching daemons. I have made my config file available to download for anybody who is interested in doing the same: http://www.designsolution.co.uk/resources/rsync/ Perhaps this may be added to the resources section of the rsync site?

Overlap receiving timeout, plus dialplan latency, causes network to retry SETUP https://issues.asterisk.org/view.php?id=16789 This patch removes the requirement that some may have found that you need to insert a Proceeding() statement very early in your dialplan, otherwise an inbound overlap call may retry and fail. Our experience was from a PRI connected PABX, if we took too long doing

Hi, We are trying to port over a PABX to our network. Both PRI's seem to be live however, whenever someone dials out from the PABX Asterisk happens to report : -- Extension '' in context 'samsungincoming' from '736327438' does not exist. Rejecting call on channel 0/31, span 2 If crc4 is turned off, it reports a yellow alarm. Any suggestions? Regards, Sahil

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 everyone, I hope someone can help me with a problem I'm having with Cisco 7940 phones on the SIP 8.12 image. When I place a call from one of the handsets, the call proceeds as normal for 20 seconds and is then terminated by Asterisk (1.4.26.2): [Oct 3 10:08:55] WARNING[1650]: chan_sip.c:1981 retrans_pkt: Maximum retries exceeded on transmission 00215553-

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

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 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

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

Hello all, It 's dificult to explain; The system I need is an box option (based on *), that I would add to an existing PABX (ie: Nortel with 600 ext). I need two E1/T2 card to plug the system between Telco (FT) and PABX (Nortel)! One card for France Telecom Side (E1a) and one other to Nortel Side (E1b). --------- -------- ----------- Telco FT

? 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

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",

Stephan, Ok, I'll re-state the problem... I have two devices that I want to talk to each other: 1. an Asterisk PBX 2. a Damm Cellular TETRAFLEX digital radio system (www.damm.dk) both devices are effectively "gateways" because they have many subscribers behind them. The Damm Cellular system controller is based on Windows-XP Embedded and its sub-systems used the OpenH323

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

My problem is that I'm trying to do a flash on an active ZAP channel to transfer a call, but every time the flash is performed the caller that im trying to transfer gets disconnected. Here is a longer explanation of whats going on. I have a situation where I am linking asterisk upto a PABX via FXO modules. Calls will come in via the traditional PABX and be transfered to the asterisk system

Hi, I am trying to integrate Asterisk with a very old PABX I have here for test purposes. I have it linked with and FXO module. Now the test scenario I am building goes like this: Incoming call on Legacy PABX --> Call Transferred to Asterisk --> Announcement Played --> Call Transferred to SIP Xtn --> If call is unanswered perform a hook flash on active zap channel and return it to