similar to: RE: Asterisk-Users digest, Vol 1 #1808 - 13 msgs archives gsm of asterisk ???

I was wondering if someone might be able to offer a suggestion to me about how I might go about dropping a caller into a context specific to their CID. For example, I would like to be able to dial Asterisk from a specific number (a mobile phone) and have it drop me into a context other then the one that normal callers receive that has more options tailored to things I might want to do. I assume

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

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

> -----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

> -----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 =>

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

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

At the moment we're not - the email notification from Comedian Mail has been mostly sufficient. I do however have some Dialogic D/42-NS PBX emulation cards and the plan is to use them to set and unset the MWI lamps based on events pushed out of Asterisk. They may be obsolete hardware but they came in real handy for extracting the voicemail from the old StarTalk NAM too. Take a look at the

For outbound only, I have traditionally recommended VoipJet. They just recently has a spat of issues that seem to have resolved though. I am still using them via their east coast server and it seems to work quite well again. Cost is around 1.3 cents minute I believe. Use IAX and g711 for best quality to VoipJet. Thanks, Wiley -----Original Message----- From:

That's the problem we had with Asterisk and HT on a 2.4 Kernel: whenever Asterisk was staying in the RTP stream, and HT was enabled (on a Dell Dual Xeon system), we had choppy audio. After disabling HT, everything was fine again. Nothing measurable, indeed, but you could definitely hear it. So there *must* be something. -Manuel -----Messaggio originale----- Da: Peter Corlett

I have been playing around with asterisk for a week or so now and haven't had too much trouble getting things to work but one thing seems to puzzle me. I have been patient hoping that there was a configuration error on the server or that the toll-free gateway was down but nothing has changed. I have the following configuration context for IAXTEL: [iaxtel] exten =>

Hi all, Thank you for your help, and sorry for my late answer. Everything works fine by now ! Yes, you have to become a BDC to vampire the accounts ! This is why I was getting an "Access denied" error : I thought my Samba was a BDC, but I forgot to add "domain logon = Yes" in my smb.conf, so Samba was a simple share server. Here is the steps I followed to suck the accounts :

RE Message: 5 Date: Sat, 7 May 2005 23:18:46 -0400 From: Andrew Kohlsmith akohlsmith-asterisk@benshaw.com Subject: Re: [Asterisk-Users] At home Asterisk via Broadvoice? To: asterisk-users@lists.digium.com On May 7, 2005 11:04 pm, John Stegenga wrote: > Broadvoice will give me 2 lines, with 2 phone numbers each - distinctive > ring - for a reasonable fee... Please do a google search for

has anyone done caller announce in MeetMe's before? Dave P >>> brian@bkw.org 5/18/2004 5:50:49 PM >>> With multiple parking lots you can give each person their own lot thus exten 800 for everyone will connect them with just their call passing the lot name which you know for X customer. bkw ----- Original Message ----- From: "Andrew Kohlsmith"

At 1/16/04 7:25 AM, Andrew Kohlsmith <akohlsmith-asterisk@benshaw.com> wrote: >That's pure bullshit -- I use software RAID *specifically* because I value >my data. I don't want to buy two hardaware RAID controllers to have one >sit on the shelf just in case the first dies... and if the second dies >you're SOL because they've lasted long enough that

--On Monday, January 19, 2004 11:01 AM -0500 Andrew Kohlsmith <akohlsmith-asterisk@benshaw.com> wrote: <SNIP'd from the "ADSI phone vs. IP phone" thread> > I'm looking at ADSI phones simply because I don't have to re-tool my > entire building; I can use the existing phone network and (I think) get > all the functionality I need with the (far) cheaper

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:

How do you suggest doing that? How can I convert wav files to gsm files? thanks Shoval Tomer, MCSE IT Manager Softov Advanced System Ltd. Email: shoval@softov.co.il Mobile: 972-55-229220 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031102/f84d7805/attachment.htm

Is it possible to use * as a VOIP gateway? Can I connect asterisk to one of the trunks on my current PBX and on the other side of the world connect another * to the trunk of another regular PBX - is it possible to transfer calls from here to there? I guess I'll need one port FXO card for each asterisk, but I can't figure how to configure the thing. I know I'll need to