search for: hookflash

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, saw the hookflash and passed a SIP INFO (event hookflash) back to the Proxy. The Proxy sent this message on to the calling SIP phone which replied that this "feature is not implemented.&...

...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...

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 is going crazy, spewing this message: -z: No such file or directory -z: No such file or directory -z: No such file or directory It doesn't seem to hurt any...

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, aste...

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 was...

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...

...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 used for the blind transfer (hookflash) --- [from-sip] exten => _NXXX,1,Flash() exten => _NXXX,n,SendDTMF(${EXTEN}) exten => _NXXX,n,Hangup() [from-zap] exten => s,1,Dial(SIP/sv0071iv) exten => s,n,Dial(SIP/sv0072iv) exten => s,n,Goto(AA,s,1) [AA] exten => s,1,Wait(.5) exten => s,n,Background(vm-whichbox) exte...

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: http://lists.digiu...

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, however, it's not working. I'm not sure if it's me, the CPE device, Asterisk, my dialplan code or a combination of all. I am running As...

...-- Call token is ip$localhost/29426 -- Call reference is 29426 -- DTMF Payload is [pt=101] -- Called 3500@192.168.1.150 Setting capabilities to 0x8 (alaw) Capabilities in preference order is (alaw) Allowed Codecs: Table: G.711-ALaw-64k <1> UserInput/hookflash <2> UserInput/RFC2833 <3> UserInput/dtmf <4> Set: 0: 0: G.711-ALaw-64k <1> 1: UserInput/hookflash <2> 2: UserInput/RFC2833 <3> UserInput/dtmf <4> -- Sending SETUP message -- Transmitting...

...Listener:81249e8 H323 Awaiting TCP connections on port 1720 > 0:00.011 OpenH323 Wrapper H323UDP Binding to interface: 0.0.0.0:5000 > 0:00.011 OpenH323 Wrapper H323 Added capability: G.729{hw} <1> > 0:00.012 OpenH323 Wrapper H323 Added capability: UserInput/hookflash <2> > 0:00.012 OpenH323 Wrapper H323 Added capability: UserInput/basicString <3> > 0:00.012 OpenH323 Wrapper H323 Added capability: UserInput/dtmf <4> > 0:00.012 OpenH323 Wrapper H323 Added capability: UserInput/RFC2833 <5> > 0:05.829...

...address 192.168.65.200 255.255.255.0 duplex auto speed auto ! interface FastEthernet0/1 no ip address shutdown duplex auto speed auto ! ip http server ip classless ip route 0.0.0.0 0.0.0.0 192.168.65.1 ! ! ! dialer-list 1 protocol ip permit ! ! call rsvp-sync ! voice-port 2/0 no vad timing hookflash-in 750 ! voice-port 2/1 no vad timing hookflash-in 750 ! voice-port 2/2 no vad timing hookflash-in 750 ! voice-port 2/3 no vad timing hookflash-in 750 ! voice-port 2/4 ! voice-port 2/5 ! voice-port 2/6 ! voice-port 2/7 ! voice-port 2/8 ! voice-port 2/9 ! voice-port 2/10 ! voice-port 2/11 ! vo...

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 o...

...ted endpoint. 0:01.160 H323 Cleaner H323 Started cleaner thread 0:01.161 OpenH323 Wrapper H323 Started listener Listener[ip$*:1720] 0:01.161 OpenH323 Wrapper H323 Added capability: G.729{hw} <1> 0:01.161 OpenH323 Wrapper H323 Added capability: UserInput/hookflash <2> 0:01.161 OpenH323 Wrapper H323 Added capability: UserInput/basicString <3> 0:01.161 OpenH323 Wrapper H323 Added capability: UserInput/dtmf <4> 0:01.161 OpenH323 Wrapper H323 Added capability: UserInput/RFC2833 <5> 0:01.162 OpenH323 Wra...

...gatekeeper. == New H.323 Connection created. -- asterisk-1234 is calling host petew -- Call token is ip$localhost/13054 -- Call reference is 13054 -- DTMF Payload is [pt=101] -- Called petew Allowed Codecs: Table: G.711-uLaw-64k <1> UserInput/hookflash <2> UserInput/dtmf <3> Set: 0: 0: G.711-uLaw-64k <1> 1: UserInput/hookflash <2> 2: UserInput/dtmf <3> -- Sending SETUP message =-= In OnAlerting for call 13054: sessionId=101 -- Ringing phone for &quot...

...323 Awaiting TCP connections on port 1720 0:00.214 OpenH323 Wrapper H323UDP Binding to interface: 0.0.0.0:10000 0:00.243 OpenH323 Wrapper H323 Added capability: G.711-ALaw-64k{hw} <1> 0:00.279 OpenH323 Wrapper H323 Added capability: UserInput/hookflash <2> 0:00.279 OpenH323 Wrapper H323 Added capability: UserInput/basicString <3> 0:00.279 OpenH323 Wrapper H323 Added capability: UserInput/dtmf <4> 0:00.279 OpenH323 Wrapper H323 Added capability: UserInput/RFC2833 <5> 4:0...

...at all! I worked all last week to try to iron out the 1.4 zap-transfer CDR issues. I have 12 cases I test with involving hook-flash and #-blindxfers, and so far, I've got 9 of the 12 working OK (as far as I'm concerned.), but I have 3 cases that come up with problems. For instance, if you hookflash, and dial a number, the CDR's will be different, if you hang up before the dialed party answers, versus hanging up afterwards... The diff between a blind xfer and an attended xfer (without the 3 way), I guess, but I lose the calling channel name... I'll try to sort all this out, and then I&...

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...

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).