similar to: Zap callgroup/pickupgroup question

The problem: T100P connected to an Adit600. Channel 1-16 are FXS, 17-24 FXO. I have Zap/24 in callgroup 3 and Zap/1-16 in pickupgroup 3. When a call comes in on Zap/24 I cannot pick it up with *8 from Zap/1-16. *CLI> show version Asterisk CVS-04/27/04-23:48:08 built by root@tuck on a i686 running Linux The zapata.conf and extensions.conf are located here:

Hi all How can I make * ring one phone then if no answer Go to a different extension ?? Any help always appreciated Regards Mick

Hi, I asked a time ago what were callgroup and pickup group used for. I have done some proofs and all, and I'm not sure if I have pick the idea up well!! That's what I understand: For example: group=1 callgroup =2 and pickupgroup=2 and my phone is a membership of the group 1. that's mean that when a phone that belong to group 2 is ringing, I'll be able to answer this call dialing

I have just started to play with callgroups and pickupgroups.. I updates my * from CVS this morning (about 15 mins ago).. I have placed callgroup=1 and pickupgroup=1 into each of my 3 phone configurations in sip.conf.. I place a call from phoneA to phoneB, then I go to phoneC and dial *8# , the call does not get picked up by phoneC and continues to ring on phoneB.. Have I not configured

HI,ALL, I have multiple PSTN lines registered as multiple SIP channels (e.g. SIP/line1, SIP/line2, SIP/line3, etc...), on the multiple gateways( I uses the SIPURA3000). I wants to arrange them into an ordered hunting group for outbound calling. I used http://www.voip-info.org/wiki/view/hunt-dial+macro for reference. My configure files like blew.

Hello. The procedure so that it works you can find in: http://www.voip-info.org/wiki-Convert+WAV+audio+files+for+use+in+Asteris k a the files .wav chmod 755 file.wav sox file.wav -r 8000 file.gsm resample -ql chmod 755 file.gsm in extensions.conf xxxx=> xxx,x,playback(file) Ing Javier Rios Ing de Proyectos 04167285748 212 2637246 /2637187 -----Original Message----- From:

I have problems using the call pickup under Asterisk 1.8.4.2. I have another Asterisk with 1.6 - and it is working fine with the same settings. I have setup the same callgroup and pickupgroup for all extensions in sip.conf - just to make things simple for testing. The sequence *8 seems to be completely ignored by Asterisk - the client shows "Call answered" when dialing *8 while the

I'm trying to use Jeremy's suggestion of dialling using just the peer name instead of user:pass@peer but I'm running into some really funky issues. It does the same thing with both VoicePulse and another * server I have. [voicepulse] type=peer host=gw5.voicepulse.com trunk=yes user=USERNAME pass=PASSWORD and in my dialplan: Dial(IAX2/voicepulse/${EXTEN:2}@VPWS,90,r) The log shows

The answer is yes, I think, but I don't recall precisely how off the top of my head, and I'm walking out the door in a moment. The phone will hold more than a dozen distinct ring tones which you can create for yourself, and you can have asterisk direct it to use a ring tone independently of line appearance. The most direct way would be with the SIP Alert-Info field, but the phone itself

Hey, Thanks for the input Andrew. I did all you suggested but noticed that when I did the loopback test, the output *was not* there as you mentioned ("I'm set to pri_net, but the other side thinks it is pri_net!"). In fact, the same message as before kept repeating every second or so: >> Unnumbered frame: >> SAPI: 00 C/R: 0 EA: 0 >> TEI: 000 EA: 1

In response to the postings by Andrew Kohlsmith and Ernest W. Lessenger: Andrew, I modified the exten line in extensions.conf as you suggested. Unfortunately, It still does not work... Ernest, I spent approx. 4 hours reading list archives (and anything else Google served up) on how to configure iax.conf and extensions.conf to work with Voicepulse. Then, I sent an email to voicepulse

> -----Original Message----- > From: Andrew Kohlsmith [mailto:akohlsmith-asterisk@benshaw.com] > Is the channel physically being hung up before the * tone is heard? Good question. If it is, Asterisk doesn't detect it -- the PBX doesn't support Kewlstart-style disconnect notification. The sequence I hear on the extension, when I plug in an analog phone, is the click of the

Has anyone ever heard of this so called Dynamic Impedance matching on the ADIT 600? I called their support and they've never heard of it. We are of course having echo problems are on the far end due to digital/analog conversion on the local end using a channel bank. We have purchased an ADIT 600 and yes the complaints are "far less" however we're still getting them. While I have

> -----Original Message----- > From: Andrew Kohlsmith [mailto:akohlsmith-asterisk@benshaw.com] > Sent: Tuesday, 25 November, 2003 08:56 > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] Picking a channel (FXO port or SIP) for outbound calls > > > > Yep, we use it for international calling. Works great: > > exten =>

Hi guys! I?m using Asterisk 1.2 with mISDN support. I have problems with Pickup calls with my Grandstream Buttons . I set up on Dial Plan this: Exten => _**XXX,1,Pickup(SIP/{EXTEN:2}) but it doesn?t work if the call comes from mISDN. So, I wanna do something to this: Exten => _**XXX,1,SendDtmf(*8#) because if I introduce *8# into my telephone i can pickup a call from everywhere. BUT the

On Friday, June 10, 2005 3:16 AM, Andrew Kohlsmith [SMTP:akohlsmith-asterisk@benshaw.com] wrote: > On Friday 10 June 2005 04:08, Terry H. Gilsenan wrote: > > Received: from source ([81.56.129.44]) by exprod5mx8.postini.com > > ([64.18.4.10]) with SMTP; Fri, 10 Jun 2005 00:29:16 PDT > > > > Your MTA claimed it was called "SOURCE" but rDNS tells the recipient

On 6/17/23 08:47, Steve Matzura wrote: > > Both Background() and WaitExten()  allow the caller to enter DTMF > digits. Asterisk then attempts to find an extension in the current > context that matches the digits that the caller entered. If Asterisk > finds a match, it will send the call to that extension. > > > My question then is, is "*" a valid exension, as

Hi All! Have a problem with remote call pickup via sip. When 1 sip phone is ringing and I'm trying to pickup a call from another sip phone by dialing *8# I'm getting: -- Sent into invalid extension '*8#' in context 'from-sip-post' on SIP/ciscok-8d39 such configs: zapata.conf ------ context=inbound-analog callgroup=2 channel=2 ------ sip.conf ------ [ciscok] type=friend

Been learning * now for a couple of weeks and have all basic features running including VM, MoH, FX lines, iaxtel, and FWD. However, I seem to be lacking documentation on a lot of technical things and am wondering if I overlooked something that is obvious to others. (I do have the Handbook, have been doing a fair amount of google searches, and read the README.* files.) Examples, Where should I

That is how many old PBX phone systems work and it is that way our users are used to working with the phone system. Another issue with the way Asterisk callparking currently works is that there is only one call-park orbit, you cannot use a different set of numbers for a different call park instance(i.e. 700 goes to 701-720 AND 740 goes to 741-750). We also have several Grandstream phones which