Re: [asterisk-users] pick up IAX2 calls

Try with ** + iax extension Regards, Luis Morales On Mon, Nov 24, 2008 at 10:20 PM, Bruno Castelo Branco <[EMAIL PROTECTED]> wrote: > Hi > > Somebody knows if pickup call works with IAX2? > I enable *8 in features.conf, but doesn't works with IAX2 extensions. > Any idea? > > thanks > > > > _

[asterisk-users] pick up IAX2 calls

Hi Somebody knows if pickup call works with IAX2? I enable *8 in features.conf, but doesn't works with IAX2 extensions. Any idea? thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSC

Re: [asterisk-users] asterisk and v4l

Jerry Geis wrote: > I am looking for any information on using vlc and asterisk. > > I'd like to take a normal video phone and call into a desktop > with a usb camera. > > Is this possible? Thanks, > > Jerry > I really meant to type v4l not vlc... Again - I am looking at using a USB web camera (v4l

Re: [asterisk-users] IMAP voicemail with Exchange (was: A way to run extenrnotify when IMAP events take place...)

> I have IMAP voicemail working with Exchange 2003 using a single username and > password for multiple mailboxes. > > Sorry to hijack this thread (at least I changed the Subject), but this > really caught my eye. I was under the impression that Exchange's IMAP > doesn't have the master user featur

Re: [asterisk-users] Problem with DAHDI and OSLEC integration.

>Have you copied there the files from the directory drivers/staging/echo >in a recent (that is: >= 2.6.28-rc1) kernel tree? Tzafrir, Thank you for following up on this. I don't have a quick command for only the three files, I just grabbed the tar ball. But like the OP, the only difference was that

Re: [asterisk-users] Problem with DAHDI and OSLEC integration.

On Sun, Nov 23, 2008 at 11:44:04PM +0100, Marco Signorini wrote: > Hi List. > I've bought a new server for my home asterisk installation and I'm > trying to install asterisk with dahdi drivers and OSLEC. To do that I've > got the svn dahdi-linux trunk revision 5366 and the echo subproject from > th

Re: [asterisk-users] sip trunking and call transfer

On Sun, Nov 23, 2008 at 5:54 PM, Eric ManxPower Wieling <[EMAIL PROTECTED]>wrote: > The term you are looking for is "reinvite". Reinvites allow two devices > to send audio directly between the two end points of the call. the > SIGNALING stays on the servers, but the audio can be sent directly >

Re: [asterisk-users] Problem with DAHDI and OSLEC integration.

>Am I doing something wrong? I just posted this exact issue on Wednesday: http://lists.digium.com/pipermail/asterisk-users/2008-November/222063.html I never got any response and Digium came through with keys for my HPEC license in the nick of time. I am not pleased with the admin overhead HPEC re

Re: [asterisk-users] sip trunking and call transfer

Maybe because there is no such thing as a "SIP trunk", at least in the Asterisk world. Most of us call them "peer" or "friend". The term you are looking for is "reinvite". Reinvites allow two devices to send audio directly between the two end points of the call. the SIGNALING stays on the se

[asterisk-users] asterisk and vlc

I am looking for any information on using vlc and asterisk. I'd like to take a normal video phone and call into a desktop with a usb camera. Is this possible? Thanks, Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- ast

[asterisk-users] Problem with DAHDI and OSLEC integration.

Hi List. I've bought a new server for my home asterisk installation and I'm trying to install asterisk with dahdi drivers and OSLEC. To do that I've got the svn dahdi-linux trunk revision 5366 and the echo subproject from the 2.6.28-rc6 Linux kernel sources. As reported in the dadhi README file I'v

Re: [asterisk-users] sip trunking and call transfer

Maybe my question is not clear or is too stupid? (sorry) Maybe this is already done in SIP trunking? Or (worste case) is impossible to do that? Thanks On Fri, Nov 21, 2008 at 8:53 AM, nik600 <[EMAIL PROTECTED]> wrote: > Hi to all. > > i-ve got a question: > > what happen when a call between 2 t

Re: [asterisk-users] How does IMAP notify Asterisk that I've read a message?

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Tilghman Lesher wrote: > In 1.4, use the "checkmwi" option in the [general] section of sip.conf to > specify how often the SIP channel should poll for message count. In 1.6, > the message waiting has been changed to use a notification system by > def

[asterisk-users] Asterisk 1.6, IMAP Voicemail and externnotify

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I have Asterisk sitting between the PSTN and a legacy PBX. Asterisk is doing some IVR work prior to forwarding calls to the PBX and it also acts as the voice mail server for the PBX, with Asterisk configured for IMAP storage. When a call comes in and

Re: [asterisk-users] Collect digits from the Callee after the Call is connected.

Simith Nambiar wrote: > > Hello Doug, > Thank you for your response, if you see my > e-mail above, i wanted the Read to happen after the Call is connected > (i.e after Dial Pen and paper? Seriously, I thought you were just having trouble elucidating your thoughts.

Re: [asterisk-users] How does IMAP notify Asterisk that I've read a message?

On Saturday 22 November 2008 22:17:54 Barry L. Kline wrote: > I have an Asterisk box sitting between the PSTN and a legacy PBX. I > have successfully configured Asterisk to use IMAP for voicemail and have > written the necessary script to turn the MWI indicator (via a .call file > to the PBX) on a

Re: [asterisk-users] SendImage()

On Saturday 22 November 2008 22:18:05 Rob Hillis wrote: > Philipp Kempgen wrote: > > SendImage() in 1.4: > > > > ---cut--- > > SendImage(filename): Sends an image on a channel. > > If the channel supports image transport but the image send > > fails, the channel will be hung up. Otherwise, the di

Re: [asterisk-users] Need Recording Solution in Asterisk

Well, I think this is a little difficult to deploy with a single system. Maybe you will need to deploy some distributed solutions (Asterisk or whatever) to have quality assurance, and coordinated to a central management. You will also need to verify how many concurrent calls between extension

Re: [asterisk-users] Collect digits from the Callee after the Call is connected.

Doug Lytle wrote: > Simith Nambiar wrote: > >> Hello Darrin / Doug, >> Thank you for your response, i find >> that the Read Aplication blocks for input and returns when a DTMF is >> dialled, which is fine. >> My problem is that when i use the Dial Applicatio

Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities

Yehavi, You might want to check out some of the EDUCAUSE mailing-lists to find out what other universities are doing. -- Eric Chamberlain ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- aster

Re: [asterisk-users] How does IMAP notify Asterisk that I've read amessage?

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 [EMAIL PROTECTED] wrote: > Regarding #2 - There is nothing in Asterisk, at this time, that is > able to check the status of a attachment in a message in an external > system. AFAIK. > Ooops... I obviously wasn't clear in what I was asking. When I

[asterisk-users] CDR Design

I've taken the liberty of starting a new thread to discuss the design of the Asterisk CDR mechanism. The discussion has been kindly initiated by murf putting together a proposal: http://svn.digium.com/svn/asterisk/team/murf/RFCs. After reading the proposal I still don't think it's the right way to

Re: [asterisk-users] How does IMAP notify Asterisk that I've read amessage?

>>> On 11/22/2008 at 11:17 PM, "Barry L. Kline" <[EMAIL PROTECTED]> wrote: > -BEGIN PGP SIGNED MESSAGE- > Hash: SHA1 > > I have an Asterisk box sitting between the PSTN and a legacy PBX. I > have successfully configured Asterisk to use IMAP for voicemail and have > written the necessary s

Re: [asterisk-users] hint priority with 50 channels

Ups, some more details. I just tried with different values: 800, 8000, 24000 I did a make clean && make && make install, restarted asterisk etc... Unfortunately it does not seem to work correctly. So I have a hint priorities like this: *1,hint,SIP/1&SIP/2.SIP&100 *2,hint,SIP/1 *3,hint,SIP/1

Re: [asterisk-users] hint priority with 50 channels

I just tried with different values: 800, 8000, 24000 I did a make clean && make && make install, restarted asterisk etc... Unfortunately it does not seem to work. So I have a hint priority like this: *1,hint,SIP/1&SIP/2.SIP&100 If SIP/1 or SIP/2 rings the priority is updated but not for SI

[asterisk-users] Asterisk 1.6 mysql cdr log problem

Hi all! I've installed a new Asterisk 1.6.0.1 with addons and dahdi drivers and tools but my calls aren't logged. I'd enabled mysql log and noticed that asterisk send a 'DESC cdr' so connection is working between asterisk and mysql and I am able to call other phones so Asterisk is working as well.