similar to: Weirdness on "hookflash call pickup"

I think I've managed to figure out that there are two ways to transfer a Zap call, using hookflash (defined in zapata.conf) or the # key (the t and T options of the Dial command in the dialplan), but not why there are two ways to do this, nor what the difference is between them. Is there something that explains this? thanks -------------- next part -------------- An HTML attachment

Don't know what else to call this. Googling and some time on the IRC channel haven't gotten me anywhere. Here's the sitch, which is a bit complicated but is something my customers are in fact encountering on an everyday basis: 1. Bob is on a Zap channel talking through the PSTN to Carol. Both have the misfortune, like so many of us, of having LECs who do not offer disconnect

Here is a question that needs a few opinions... Recently we installed a couple of FXS gateways into a site with a SIP Proxy/Softswitch other than Asterisk. One of the things that the users on this site need to do is receive calls on single line phones on the FXS gateways and then hookflash and transfer them to other SIP users. We found that the FXS units, true to their nature as VoIP gateways,

Hi, Has anyone out there managed to do a hookflash transfer with a Micronet 5050s gateway ? We have just tried out this gateway and it seems to do everything we need except this particular feature. Also if you have succeeded where is the hookflash time specified in the gateway. There does not appear to be any parameter for this. Perhaps it is not supported at all. Any help appreciated.

I wonder if I'm the only one whose customers are having trouble with hookflash on their TDMXXX cards. The problematic situation of record for us is a user who is on a call, and then wants to do one of two things: Hang up that call and take another one coming in Hang up that call and make another new call What happens is that instead of seeing the event as a hangup, asterisk perceives

Bear with me this is a bit long winded. I am having some issues making automated outbound calls over Broadvoice from my Asterisk 1.4.2 server. For reference, none of the below issues happen when I make the calls to VoIP phones attached to the Asterisk server. What I am trying to do is call, using a .call file, out via the SIP trunk we have setup, and when the party picks up use AMD to

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

I am using asterisk at home with a Cisco ATA186 and a clone X100P card. My inbound telco line is plugged into the X100P card. My telco line is also plugged into other phones in the house for now so someone else can answer the phone without asterisk being involved. What I would like to do is if someone has answered the call on a normal phone in the house I would like to be able to join the call

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, Is there a way to change the hookflash timing with the TDM400P? Allready been searching the mailing list/google etc but i can't find anything ;-( I tried flash= in zapata.conf, but that only works with the T1/E1 cards. Greetz, Caspar -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, My configuration is SipPhone<-->*1<--->*2. My asterisk version is 1.4beta3. I installed pwlib,openh323,chan_h323. When i call from SipPhone--(SIP)-->asterisk1---(H323)-->asterisk2, there is no audio. Using 'rtp debug', I can see that rtp packets are being received. Rtp packets are being exchanged. I also tested chan_ooh323, but to fail. Can anyone recommand best

hi .. i've just installed and licensed an instance of the G729 codec. I am trying to connect through asterisk to Go2Call server .. According to their info it involves dialling extension 729 on voip01.go2call.com, to get the IVR. my extensions.conf shows : exten => s,2,Dial(OH323/h323:729@216.52.153.206) which I think is correct, I have G729 enabled in the OH323.conf file and it seems to

Asterisk SVN-trunk-r7353M (will be moving to 1.2.2 this weekend) E1 connected to Sangoma A102 SIP phones (Cisco 7960) I've been making a call from my mobile to the office, when, suddenly the conversation is terminated and replaced by a "fax-type" sound. This has happened to me several times over the past year, so it's not the version of asterisk (we've had cvs trunk and

In the simplest terms I can think of, I'm going to describe what I want to do and I want to know if it's possible in the current version of asterisk. Can I take a T38 call from an ATA, convert that back to analog and have asterisk screech that out on a POTS line to a remote fax machine. Would it work? And could I receive an incoming fax the same way? Please don't talk to me

I just deployed * on my home system last Sunday. 2x since then the Zap hardware seems to have malfunctioned on some way. One time it would just screech out one FXS, even though it would ring. The other time * would bridge to my FXO but it never got out on the line. I have a new TDM400 with 3 FXS and 1 FXO. Both times I tried unloading the zaptel drivers (which worked) and reloading them,

Hi Friends, Hope you would help me out here, I have searched the asterisk user list for hours and also read the readme and test files that comes with the driver. I need a very simple scenario. I have SIP clients and want to use oh323 to dial out to PSTN using a h323 gateway. a)If I set the extention.conf like this: exten => _87.,1,Dial(OH323/16.52.153.206) oh323 dials out (I can ring a

Hi, I've googled it and look in voip-info.org <http://voip-info.org> without any success. Hope someone can point me to the right direction. I saw a couple similar questions, but don't see any answers. Fedora Core 2 2 X100P(clone) PSTN Asterisk 1.07 Everything seems to be running fine, but on occasion, Asterisk answers the call with high pitch screech sound (like fax or modem

When I make a regular call from my SIP phone connected to my Asterisk server I have no issues, however when I make a call using Originate : 'Channel'=>"SIP/1XXXXXXXXXX@sip.broadvoice.com", 'Context'=>'mycontext', 'Exten'=>'899', 'Priority'=>1, 'Callerid'=>'whatever')); It creates a screech sound when the

Hello, We are using asterisk 0.4.0 on debian sid with Cisco 7960 and ATA186 phones. All sip entries have: callgroup=1 pickupgroup=1 However I am unable to remotely pickup a ringing phone using *8#. I get fast busy tone. Is there some flag to add in extensions.conf ? Thanks in advance,

Is it possible to send an external hookflash command to the Digium FXO card from the asterisk PBX? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031007/fe5be94d/attachment.htm