search for: hookflashes

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,

...hat conversation is going on, Ted calls in on another line and selects the ACD option for Bob, and Bob sees Ted's CLID on his phone and hears the CW tones. 3. Bob wants to hang up his call with Carol to talk to Ted. *** Now the possible scenarios, and their apparent resolutions *** A. Bob hookflashes and takes the call from Ted. BUT NOW CAROL'S PHONE IS STUCK OFFHOOK "LISTENING" TO MUSIC ON HOLD FROM THE THREE-WAY CALL CAUSED BY THE HOOKFLASH. That surely isn't what we want. . . . B. Bob hangs up for longer than a hookflash period. Picks up the phone and gets a fresh di...

I'm sure dumb when it comes to describing things that happen on my system. I'm making an outbound call on my ATA186 when another call comes in. I first get the nasty CID screech and then the periodic call-waiting tone. So far, so good. Then I hookflash, and just like it's supposed to, the waiting caller is on the line. But during the duration of that conversation, my console

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

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

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.

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

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:

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, 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

I repost a message I put a week ago: I have a Cisco IAD 2431 which has MGCP protocol; I cannot make it to work againts Asterisk; at least there is some MGCP conversation between them but when I offhook a phone attached to IAD I get no tone at all. As anybody managed to get working Asterisk against an MGCP Cisco gateway ? Which MGCP version should I use ? Also I recently

Hi, I'm using zaptel with three way calling and call transfers with a hookflash. If I try transfering a call to an extension that is engaged I get an engaged tone. This is fine, but how do I get back to the caller? If I hookflash again I seem to put on a three-way call and the caller can hear the beeping. I can press hookflash again but I'd prefer the caller to hear only the hold

Hello, I've been trying to send traffic to a Cisco Call Manager 3.2, but with no luck. Here's whats happening: * Call gets to CCM * Call gets to the gateway * Rings a couple times on destiny * Call gets hungup. On the CCM I get the following error: MediaManager - ERROR wait_AuConnectErrorInd On the Gw (Cisco AS5300) I get a disconnect cause of 2F (Resource not available) On asterisk:

Hi all, I'm using following software and equipment and I have very strange behavior: Asterisk CVS-NHEAD-05/30/05-16:42:41 H323 gatekeeper - GnuGK 2.2.2 IAX2 client - Firefly 1.9.8 build 3945 H323 client - SJPhone Build 1.50.271d H323 gateway - Welltech Wellgate 3504A When I dial from Firefly (IAX2) -> SJPhone (H323) everything works as expected. When I dial from SJPhone (H323) ->

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

...ou may or may not be seeing right now. AFAICT, transfers pretty much result in confused CDR's. I gave up totally on generating separate CDR's for any 3-ways that might occur. Such 3-ways will end up being billed to the dialing parties. Here's an interesting situation: A calls B, A then hookflashes, and then A calls C, and hookflashes again. It's now a 3-way call, between A, B, and C. A then drops out and B and C converse. My goal with this situation was to have two CDR's, one for A->B and one for A->c. Since A placed both calls, it seems only just that A pay for B's and C...

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

...ur stations. When a call that has been rung in using that macro transfers the call things work just fine as far as the "other" instrument ringing. But once the ring timeout has expired, the call then drops into the *original station's* voicemail. E.g. Tammy picks up the call and hookflashes, then dials Jim's extension. Jim's phone rings for 20 seconds. But if he's not at his desk, the call then goes to *Tammy's* voicemail. I've gone through the WIKI and mail archives looking for the solution, but it's sort of hard to conjure the correct search string, I h...

The solution to the problems with the Grandstream 1.0.4.39 firmware is to use inband (in-audio) DTMF. Neither the RFC2833 nor INFO seem to work. However, this presents another problem. When I'm using g729 to place a call, I get the warning "Unable to process inband DTMF" because inband is not supposed to work with g729 (although it does seem to work when I've tried it so far).