similar to: Hanging up one call when you have call waiting

I am working with the telantek.adsi file, and I was wondering how I would create a softkey for Transfer. I tried making a key definition and using SENDDTMF "#", but that didn't work. Is there another way I could do this? Also, does anybody have any ADSI scripts for use with Asterisk that they would like to share? Thank you for your time. __________________________________ Do you

I recently set up Asterisk with an Adtran TA750. All is well except the phones do not show the MWI. I have configured zapata.conf properly, as all phones will receive a stutter dial tone if there is a message waiting in it's assigned mailbox. Does anybody know how I might fix this problem? Thank you for your time __________________________________ Do you Yahoo!? Yahoo! SiteBuilder - Free,

I have just received a T100P and an Adtran TSU 600 in the mail. I seem to be having a problem with the T100P card. So far I have done the following: vi zaptel.conf fxoks=1-22 fxsks=23-24 ... vi zapata.conf ... signalling=fxo_ks ... channel => 1-22 ... signalling=fxs_ks ... channel => 23-24 I then run modprobe zaptel modprobe wct1xxp ztcfg -vv There are no errors to report. In

I've tried to ask this before but didn't want to waste a lot of bandwidth. However, I have to give the params in order to make the exact question understood. We have 2 ZAP FXO connected to two phone lines(2xX100P) 3 phones connected to a TDM400P with FXS modules Usually a caller will be on ZAP/2 and the first outbound line in the g1 group is on ZAP/1. There are two (main) call waiting

Hi I am setting up an Asterisk System in an office environment, Incoming and Outgoing calls are working ok, but i am having a few strange problems regarding call waiting. With the SNOM 200 (firmware 2.02t) phones, if you are on a call and a 2nd call comes in, the call waiting beep is played and the light flashes, but if you hang up the 1st call, instead of the phone ringing, it connects the

I keep getting segmentation faults when I do a reload. Here are the core file outputs from gdb: (I have three of them and they produce the same output) (gdb) core core.6044 Core was generated by `asterisk'. Program terminated with signal 11, Segmentation fault. #0 0x401519fc in ?? () I have no idea what that means, but if somebody could point me in the right direction, that would be

I just received an unlocked ADSI phone and I am playing with the ADSI script. I was wondering how I can include Voicemail functions (Check new messages, Delete message) into the soft buttons. I checked in app_voicemail.c and it looks like these functions have already been programmed. Is there a voicemail.adsi script somewhere? If not, then how do I get the functions I want onto my phone? Thank

I was just wondering if there was a way that you could have two calls on one line and switch between the two without initiating a threeway conversation? I would imagine that Flash is the way to do this, but when I Flash twice, a 3-way call is initiated. If I turn threeway off, then I can't transfer. Also, is it possible to hang up one of the calls, and then continue talking to the second

Hi all, How can I implement a full-featured Call-Waiting behavior on the Asterisk level (e.g. I don't want to relay on end-equipment capabilities)? I found it very strange that such a basic feature is not built-in in Asterisk (and I've googled a lot in search for this). Here is what I need: SomeuserX is calling MyUserA. They are on conversation (assumption: voice is via the Asterisk)

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

I had a similar issue with sending a flash to the PSTN for call waiting. I found that my dial plan in the extensions.conf file was not allowing me to dial *xx. Once I corrected my dial plan I was able to dial *0, *69, *78, *79, etc. Training the wife on how to actually use it was an entirely different issue. I have not tried to enable 3-way calling via the PSTN. "Jeff Otterson"

I have been looking for the answer to this question for a while. Google-ing and reading the archives of Asterisk-Users has not enlightened me. It seems that this question has been asked many times, and many times it has gone unanswered. I have call waiting and three way calling on my PSTN line from Verizon (the local telco). This is connected to a FXO port on a TDM400P. I also have

I have only been able to confirm this issue with my IAXy as it is the only ATA I have. I am running 1.2.5 stable. Example: User @ exten 100 (iaxy) receives call from PSTN (call 1). While on the call, another user from PSTN (call 2) calls 100 (iaxy) sending a call waiting beep to the iaxy. User 100 answers the call from PSTN call 2 putting PSTN call 1 on hold with music. While call 1 is on

Hi, When I use the analog phone connected to Zap/1 how do I transfer & hold the caller ? When I hit the flash key, all that happens is the caller hears a beep (sounds like DTMF). But no stutter dial tone on the Zap/1 Port, just continuing conversation with the caller. What could be wrong here? Cheers Jay

I realized today that my call waiting isn't working properly. If I am using the FXS attached phone and a call comes in the FXO, it just goes directly to voicemail, with no indication (call waiting beep). If I flash there is a second dial tone, and I can initiate a second call. If I am dialed out through the FXO and another call comes in I hear the call waiting beep, when I flash over I

Is there anyway to turn off the call waiting beep in the grandstream and/or cisco ata186? I have a dial statement in my extensions.conf that rings 5 phones at the same time by combining them with the & in the dial statement. i.e.) exten => blah,blah,Dial(SIP/GS1&SIP/GS2&SIP/GS3&SIP/ata186a&SIP/ata186b,25,t) If one of those lines is being used, then the user gets a really

Hi, I hope someone could help me, i have a x100p interface for testing purpose and on each incomming call I redirect the call to handytone 388 atas, the problem comes when i'm during a call and another call comes in, i hear the call waiting beep (comming from the zap channel), but I can't catch the call as usually using flash+2 (my pstn call wait sequence), because when i flash the

Hi, I have (to me) an interesting problem. There are 3 physical extensions, 11, 12 and 13. All hang off Sipura adapters. There is also extension 10 which simply uses 'Dial(SIP/11&SIP/12&SIP/13)' to call all phones in the house. Incoming calls from outside get sent to 10 in order that they can be answered from any phone.. Now - if (say) 11 is on a call externally, and 12 calls

Hi, I'm running 1.2 beta1 in a mini call center. I have 3 queues with 10 operators, and I'm running into some trouble because when all the operators are busy answering call asterisk still sends them more, resulting in a "beep beep" (call waiting) over and over again in Xlite audio. An easy solution woud be the use of a "single line" user agent, like firefly, still

Hi, I am having problem with my Asterix server. It does not hand up after play the voicemail. The scenario is this: 1. I make a call to Asterisk's PSTN number; 2. After recording, I hang up and make the same call again. The first call would go through nicely with the voicemail recording, but the second call will hit a message saying "the other party is busy". The only way to fix