similar to: Using MAC or extension number as SIP identifier

I am trying to setup an extension of *7XXX that will allow me to dial *7 and then any extension and use the Pickup application to pickup a ringing phone. Ideally it will also check if the phone is ringing somehow and then either dial the extension or pick it up if it is ringing. But I can't get that far. If I use *7268 specially it works fine, but as soon as I introduce any wild

Hi, Been playing with Call parking, and I can`t help but wonder if I am doing something incorrectly. The way I understand it (using default config in features.conf), is I would transfer a call to extension 700, which would park the call, tell me "701". I could then hang up, go fetch the fright person and tell him "call 701 you have a call waiting for you". The way I

Hi all, I would like to ask please if there is a way to play a ring back tone from asterisk when the customer try to make a call...I already added the ringing function to the context in extensions .conf and it work perfectly...But the issue that the asterisk server is stoping playing back his own ring back tone as soon as it detect a ring back tone coming from the carrier side... Is there a way

Dear All, i have the following context defines in etensions.conf: [a2billing] exten => _X.,1,Gotoif($[${EXTEN} = 111] ? 21) exten => _X.,2,DeadAGI,a2billing.php exten => _X.,3,Wait,2 exten => _X.,4,Hangup exten => _X.,21,Playback(AR_GetGiveToID) exten => _X.,22,Wait(2) exten => _X.,23,Record(/tmp/asterisk-recording:ulaw,,5) exten => _X.,24,Wait(2) exten =>

Hello, We are using asterisk 1.6, sangoma pri card, and Cisco 7960 phones. When we make or receive calls there is a delay before voice is heard. Anyone have any ideas on where to start to debug or has anyone seen this before. We have played with settings on pri, asterisk, and phones with no change. Thanks for your help and ideas in advance. Neal -------------- next part -------------- An

Hi everyone, Currectly I'm having some troubles to get correct status of my calls throug ISDN lines, when outbound calls don't get its destination I always receive NO ANSWER as ${DIALSTATUS} despite the fact I know the target number doesn't exists or is busy at that time. Maybe there is something I must change in my zaptel.conf or zapata.conf, current configs follows: ####

Hi. In creating a custom extension, and dialing SIP/222/333#444, seems the party receives only "333" What should I do to send the # symbol? or better, where can I find that syntax? Googled a lot, nothing. Thanks! -- Rodolfo Alcazar Responsable red y datos Deutsche Gesellschaft f?r Technische Zusammenarbeit (GTZ) GmbH Programa de Apoyo a la Gesti?n P?blica Descentralizada y Lucha

Hello, I need to test canreinvite=yes with 2softphones and 1 asterisk. I want to have that : http://www.panoramisk.com/wp-content/uploads/2007/05/asterisk-call-flow-outb ridge.png But I have that http://www.zimagez.com/zimage/canreinvite.php Canreinvite=yes work for all phones or just asterisk?... Can you help me? Thank you -------------- next part -------------- An HTML

Hi, I'm using app_pppd with a Digium-PRI-card for PPP connections. I had some strange problems with some IP packets passing and some not, e.g. ftp login went well, but as soon as I tried to up- or download a file, noting was transferred. I finally guessed, it must have to do something with the packet size. Then I started pppd with the parameters mtu 296 and mru 296 as in further times with

What is the best way to have 1 phone check multiple voicemail accounts. I am using polycom 650 phones, and am wondering if mwi can work when checking multiple accounts. -Chris Sent from my BlackBerry? wireless handheld

Can anyone tell me what this message means? Got event 17 (Polarity Reversal)... I'm running DAHDI 2.0 with a TDM401 card. Asterisk version 1.6.0. It appears that I get this Polarity Reversal each time an inbound call hangs up. This results in another ring, but no one is there. It appears as an unknown caller, but I believe its a phantom. Thanks, Jim [Oct 10 12:47:54] NOTICE[6669]:

Given the following SRV records: _sip._udp.tollfree.sip-happens.com. 38400 IN SRV 10 0 5060 sometimes.sip-happens.com. _sip._udp.tollfree.sip-happens.com. 38400 IN SRV 20 0 5070 ares.sip-happens.com. Why is asterisk (1.4.17) not honouring the priority and not failing over to using other records when a connection fails? For a given call to tollfree.sip-happens.com ares.sip-happens.com was chosen

Thanks for all the comments on the 3Com phones. Thankfully, there is a large number of phones out there to dig through looking for the right solution. What I have not been able to find, after spending all weekend looking, is a good solution for an attendant console. We have 2 receptionists that need to be able to view all 60+ phones (we could probably weed it down a bit if we had to,

Hi all, We have just set up trixbox latest with a Rhino r1t1 card, hooked up to a plain E1 PRI line. We call fine SIP to SIP, but as soon as we make a call from SIP to PSTN all sounds become unintelligible screeching or static kind of noise on both ends, when we call PSTN to SIP the PSTN side seemingly OK at least we hear no screeching sound, but the SIP side is a even worse screeching

2008/9/29 Alex Balashov <abalashov at evaristesys.com> > Try this: > > exten => _XXXX,1,Set(THISTECH=${CUT(CHANNEL,/,1)}) > exten => _XXXX,n,NoOp(Technology is ${THISTECH}) > exten => _XXXX,n,Set(THISCHANNEL=${CUT(CHANNEL,/,2)}) > exten => _XXXX,n,NoOp(Channel is ${THISCHANNEL}) Hi, I don't have any spare zaptel enabled system I could try this on, but I

Is there a way to force a channel to continue in the dialplan after the remote end hangs up? Specifically, I am trying to play around with setting up a fax server. I can receive the fax, but sometimes the sending fax hangs up before my System command for printing can run and the fax never prints. I know I can work around by setting up a custom context and use the 'h'

Hi, I'm using the following setup : Alice ---- IPPhone ------<LAN>----- Media gateway ----<PSTN> ------- Phone ---- Bob For certain calls, users complains about echo : they can ear their own voice in their handset, though media gateway echo cancel is turned on. I'm wondering how this echo cancelation engine is supposed to work. My understanding of echo is that most probably,

Hi I have problem with TE121 Digium card. I connected it to modem keymile Music 200 (provided by telco) but I can see 2 red lights on modem (both bellow words rx) and my card is red too. I tried to make experiment and made loopback (pins 1 4 , 2 5) and put it in card and card become green (I hope that is sign card is ok) but on CLI i can see following error message WARNING: We think we are cpe but

Hi, Before diving into this, I would very pleased to know if someone could yes or no, successfully compile mISDN 1.1.8 on Lenny (latest RC1 or beta2 version) ? Regards After a fresh install on Lenny, I can reproduce at will : apt-get install build-essential linux-headers-2.6.26-1-686 cd /usr/src wget http://www.misdn.org/downloads/mISDN.tar.gz tar xvf mISDN.tar.gz cd mISDN-1_1_8 make ....

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