similar to: take-over && transfer question

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

Comments? Feature request: Add the ability for the "T" and "t" suffixes in a Dial command to call an extension directly (if specified) instead of going only to the hardcoded "transfer" command. Feature request: Flash events, when presented inside of an existing call, will call a pre-specified extension just like the "T" and "t" request

Need help on this.. I have configured all my internal extensions as SIP phones, i have one ATA-186 and one softphone. When I try to transfer calls between internalt, I use *1 as it configured on feature.conf. But my external line configured a IAX. When there is an incoming call from outside, the line 1 from my ATA186 rings. If I want to transfer my call using *1 is not possible. So I need to

I don't know how I managed to mess up sending this last time, but somehow it got attached to the AgentCallbackLogin thread. Since the indended audience may not see it there, please indulge me by tolerating this second copy: Here's a little tidbit about the non-functional flash key on the Budgetone 100's. I have 20 of these phones. On some, the flash key works, and on some it

Hello all, Does anyone know how to successfully transfer a call using the GrandStream Budgetone 100 phones? I've read multiple posts talking about hitting flash, the dialing, then flash again, etc. Some posts talk about using the transfer button, then dial, then flash. Anyways, it seems that I am able to put the caller on hold (whereas they hear hold music) by pressing flash or

I am new to asterisk and have been playing with it for just a few days. The GrandStream BudgeTone-102 equipment I have allows me to do a blind transfer, but I am interested to know how to do something which isn't blind. I have managed to enable the call parking which allows me to send a call to extension 700 for somebody to collect, but it still seems rather awkward. I also have

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 have a SIP phone which does not natively support SIP transfers (REFER etc...). So far all that is possible is to enable blind transfers using the t and T arguments in Dial from the # DTMF key. The phone has an R button on it and this can be setup to either send an RFC2833 hook flash message (value 16) or a SIP INFO message which you can edit the contents of (since there seems to be

Hello list, I was looking for a way to implement non-blind call transfers with *, i.e. the usual behaviour of most PBXs when pressing the flash button: - A and B are talking - A pushes flash - A is free to compose a new number - B hears music on hold - A's call is answered by C - A hangs up - B and C are in conversation As much as I can understand, * only supports blind transfers, where if

I have two call waiting problems. I have a POTS line into and FXO port and telephones on an FXS port 1) I can't seem to use the flash button(on the phone) to answer a call waiting call. I see the callerid coming though and here the call waiting tone, but I just can't seem to answer it. The flash button seems to have no effect. I have: callwaiting=yes in zapata.conf

I am having trouble getting transfers to work when a zap channel is part of the call. I have a couple SPA-2000's and some X100P cards as my setup. This is what I'm trying: Dial number from phone: -- Executing Dial("SIP/206-2c61", "Zap/1/#######") in new stack Currently on call: -- Called 1/####### Press flash to place call on hold with SPA-2000: --

Good day all We have Grand Stream BT-100 phones The transfer button work well, for blind transfer What the users want to do is, when a call comes in and asked to be transferred to another extension,for example 100,they 1ste want to speak to exten 100,then have the option transfer or not to transfer the call to this extension Currently they must pus "flash" for a new line speak to the

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

Hi all, I curious to know if other UIP200 users have this same issue: You flash (XFER button) to consult-transfer a caller to another extension. If the transfer target party is unavailable (ie: voicemail), there appears to be no way to get the original caller back. If it's a known limitation, has anyone come up with a functional work around? Thank -- ..................................

Regarding this article (1) I have one question to make. What can I do to record the call if the agent makes a transfer using the "flash" button instead of "transfer button" or using blindxfer or atxfer defined in features. conf If the agent makes the transfer with "flash", the comunication between the person who is calling and is already in the queue and the target

This is my setup: SPA-2000 -> Asterisk -> X101P (x4) -> PSTN 3-way calling works fine if I use flash and dial just local extensions. Or even if I use flash and dial one local extension, and one remote party over the PSTN. However, as soon as I dial from my SPA-2000 out over the PSTN, and hit flash the call hangs-up about 50% of the time. The other 50% of the time it puts the call on

Hi, I facing a problem with flash hook. When ever I do a flash hook to place an extsing call on hold, the call gets disconnected. The debugs on Asterisk shows that 'on hook event detected' when I press the flash button on the phone. The setup is like this Asterisk box with T1 cards and FXS cards. The T1 card is connected to an IAD and configured for ISDN PRI lines. Analog phones come

Hi. I was wondering if there's a way to do supervised transfers on the budgetone 102 . Blind works ok, but can't do supervised. I thought that with the flash button that could be possible, but seems I'm wrong .... matteo.

Hi guys, I'm trying to reuse Mitel 5215 from proprietary system now into Asterisk :) ! I've them already with SIP and handling calls sucessfully! I've followed instructions from: http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+Mitel+5220 Additional Servers * Outbound Server: Off * Outbound Server URL: blank * Outbound Server Port: blank * Voice Mail

Hi all, i am using CVS-HEAD-08/13/04-12:00:00-BRI-stuffed-0.1.0-RC4a with Cisco ATA-186 3.1.1 atamgcp We are used to make an special ;) blind transfer like (Flash)Number(Hangup before anyone answers or ring). Then * crashes (see below) if the man in the middle is an cisco-ata-186-mgcp If one waits until the last one rings, then hangup, everything is fine. If one waits until the last one