similar to: Flashing Active ZAP Channels

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",

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 Guys!! I wrote a little patch for asterisk 1.2.5 and I will maintain it for future release unless somebody explains me how we can ask people at Digium to add it to the source tree... We are planning on using Asterisk as our main PBX for the office over the next few weeks. Our current setup uses TDM400 cards to bring our 8 lines into Asterisk, our Telco provides us an option for three

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 am trying to setup a simple queue in Asterisk and I'm having a small problem. Our callers come in through a Bosch PBX and are immediately transferred to an Asterisk menu/IVR. If they select the option to call a SIP phone directly (eg. entering the operator's SIP extension) then the callee/operator can transfer the call to a phone within the Bosch system. What Asterisk does is

List, Having a little trouble with the following. Let me prefix by saying I have blind transfers working from the following setup. Inbound call [from-zap] (SIP/sv0071iv) answers. Zaptel -> Asterisk -> SIP extension SIP extension then blind transfers [from-sip] --- SIP extension -> Asterisk -> Zaptel During this whole process, the original channel off the trunk (lineside T1) is

Update: I can reproduce the error by putting the reception phone (call queue 0) in Do Not Disturb mode, then call in from outside using a mobile, then pick up the call from the 2nd phone in the queue. It will cause the error only if I hang up _before_ the mobile hangs up. The error doesn't seem to happen if the external call hangs up, or if the call is answered by the reception phone (first

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 there Background: I want to add DDI and voicemail to users on an existing analogue pabx.. It does not support ISDN. I have 10 DDI numbers via IAX which I am having sent to my Asterisk box. I have 2 X100P cards connected to 2 analogue extension ports of my main legacy analogue pabx. I have set up voicemail for each of my DDI numbers, and when a call comes in for the person at pabx

I've got a piece of CPE equipment that has an FXS port that I have tied to an FXO port on a TDM400 clone card. Normally, if I go off-hook with a standard telephone connected to it, I get a dialtone. If I dial a digit, and send a hookflash, the device will provide a dialtone back for the next available channel on the device. I'm trying to recreate this same behavior with Asterisk,

Hi all I've got a setup where I use a Sangoma PRI card driven via Sangoma WanPipe to connect to a legacy Samsung PABX (I'm not sure which model) form Asterisk 1.8.11.0. The reason is the customer has a large installed base of Samsung phones physically connected to it and on each users desk. They did not want to spring for a complete replace of all their Samsung phones with generic, and

Problem: Flash on X100P does not flash. Phone line has Call Transfer. With this line plugged into a regular phone, it can receive a phone call. Then, depress the hook momentarily, release. Dialtone is now available. Dial a different number. Call is answered. Hook Flash again, now in a three way call. Hang up. The other two parties are still in communication. Now, plug same line into the X100P.

I've tried to transfer a call using the Flash command, but with my configuration it doesn't work. I have a traditional PBX connected with a zap channel to Asterisk that acts like an IVR: TELCO line --> traditional PBX (FXS) --> (FXO) Asterisk >From the TELCO line I can make a call to the traditional PBX and reach Asterisk, the IVR system on Asterisk answers the call and I can

We use an IPcortex PABX running Asterisk 1.2.39-BRIstuffed-0.3.0-PRE-1y-y. We have recently implemented Call Queuing for our main incoming line with hold music. The call queue type is: Ring all - One call at a time (no position announcement). Since implementing this feature we've been receiving the below error daily in the mornings and lunchtime when the queue will jump to the next available

Hi, I'm having a problem with the queue behaviour in my place: I have two ISDN channels to the outside (Zap/1) and two channels two a Siemens Gigaset (Zap/4). I also use a SIP gateway to call outside and have a couple of IP phones around as well (SIP). The Gigaset has about 5 phones connected to it (+base station). Whenever two people are using those, I always am blocking two internal

Greetings. I am attempting to configure a system based on Asterisk 1.2.3 to be used as a backup should our aging voice mail/auto attendant system fail, which seems increasingly likely given its advanced years. The first part of this task is getting the auto attendant feature to work correctly, which I would have figured to be relatively easy. I have successfully built a menu structure, but cannot

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

Previously incorrectly sent to asterisk-dev list, sorry. I tried today while connected to a Jtec QSIG E1 card, with DAHDISendCallreroutingFacility with the following test dialplan: Extension 4888 is on the Fujitsu [incoming] exten => 8688,1,Answer() exten => 8688,n,Playback(connecting) exten => 8688,n,DAHDISendCallreroutingFacility(4888,8688) exten => 8688,n,Playback(goodbye)

I'm having problems using the following. [sip] exten => _*4.,1,macro(test,${EXTEN:2},${CALLERIDNUM}) [macro-test] exten => s,1,Answer exten => s,3,Flash exten => s,3,Dial(SIP/${ARG2},30,t) exten => s,4,Dial(SIP/${ARG1},30,t) exten => s,t,Hangup exten => s,i,Hangup exten => s,h,Hangup I know I must be missing something simple, but here is the output from

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