similar to: Generating Recall/Flash using Zaptel

Hi, I've solved the flash transfer problem changing the flash time in the zapata.conf file, I've set: flash = 200 (the defualt was 750 ms) in the extensions.conf the code is for example: exten => 42,1,Flash() exten => 42,2,SendDTMF(42,250) exten => 42,3,Hangup() now the transfer with flash works correctly. About the question whether my PBX expects a hook flash for

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

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 =>

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

Hi I've got an Asterisk box connected to a Siemens Hicom 300 using a Digium TDM400. The Hicom provides the calling extension as DTMF at the beginning of the call followed by two *, as in 3425** when 3425 calls my extension, I can hear all 6 tones if I have a handset connected but using Asterisk's Read application straight after Answer() Asterisk usually only gets the last *, sometimes the

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

Senario is There are 2 asterisk servers 1FXO ports connected to Panasonic PABX on extension 100 on server 1 If someone dial 100 from extension 101, call comes in on ZAP/1 call Dial,IAX2/xxxx on asterisk server 2 and from server 2 Dial/SIP/xxxx, now problem is if SIP/xxxx want to transfer this call to extension 102 then what will be the solution ? rgrds Fregi -------------- next part

Hello, hoping someone out there has some ideas - I have a VOIP line that has call waiting. It is terminated at a Sipura 3000 and the POTS side of that device connects to an FXO port in my * box. I also have a POTS/PSTN line that terminates in another FXO port on my * box. There are two FXS ports which feed cordless phones. I'm using the Zaptel TDM400 card. This gives 2 extensions + 2 lines

Olly Betts <olly at survex.com> wrote: > On Fri, Mar 24, 2023 at 10:37:41AM +0000, Eric Wong wrote: > > Realizing I had documents of hugely varying sizes (0.5KB..20MB) > > and little RAM, I instead tracked the number of raw bytes in the > > text being indexed and flushed whenever I'd seen a configurable > > byte count. Not the most scientific way, but it seems

Hello, I have found a "bug", I think in the way TDM400P cards handle FXO interface disconnect/re-connect problems. Normally I do keep all the wires connected from my CO / PABX quite securely, but I had a need to re-route the cable from one side of the desk to another, and I simply disconnected the RJ-45 connector and plugged it back in. THIS PROMPTLY RESULTED IN VERY VERY SCRATCHY AUDIO

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

Hi there, I have a x100p card in an asterisk box. Does anyone know if it's possible to do a hook flash / recall on an active zap channel? On what I'm trying to do... >From an ordinary analogue pstn telephone I can call someone, press recall, call someone else, press recall 3 & presto we're on a three way chat, with me only using one line - using the telephone company's

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

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

The issue is that delve was renamed to xapian-delve but documentation is still saying that delve is delve. Who has access to update the documentation? http://www.linuxfromscratch.org/blfs/view/svn/general/xapian.html Installed Programs: copydatabase, delve, quest, simpleexpand, simpleindex, simplesearch, xapian-check, xapian-chert-update, xapian-compact, xapian-config, xapian-inspect,