similar to: No subject

using Asterisk. =20 Is this all you want Asterisk to do? (eg as an application service rather than provide telephony for the rest of the library as well), or are you looking to have it replace your existing telephony equipment? =20 As a suggestion if you google Trixbox and Nerd Vittles you will find a fairly detailed explanation of how to set your Trixbox server (a version of Asterisk) up to

don't have a public facing web page but you are looking for people to click on but a personalized list of numbers. In order for someone to access this directory you are going to be asking for a username/password correct? If so just tie the username to a selection of 'my location' checkboxes that I tick and then the app remembers this location next time I log in (eg server side

community there is a real possibility this may come off so if you have an interest in this space and want to contribute to the discussion then this is your opportunity to do so. =20 I look forward to all opnions on this topic. =20 The slide deck for the agenda of this call is located here http://voipusersconference.org/2008-05-09-Slides=20 Cheers, Dean=20 ________________________________

display, accelerometer/motion sensor being the first 4 for release (though 81 have been mocked up so far). The long term concept is if you want a 'weather station with live video feeds and gps location control you can add various modules together to deliver what you are looking to achieve. I have high hopes for the concepts, and wish the guys well as it seems their hearts are in the right

might be your best bet to get the information you want. I'd look at voip-info.org for information. _____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Olivier Sent: Wednesday, September 16, 2009 9:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to list ongoing calls

"Why Siphon doesn't allow to receive a call when it doesn't run Apple doesn't accept (for the moment) an application runs in the background= . So, when Siphon doesn't run, the SIP server of your provider doesn't know= your iPhone." --_000_EC80F07C30CE3E46B2AD6B4407BE086F0C2AAD0248cworksmailcwo_ Content-Type: text/html; charset="us-ascii"

1. a clause in iphone Developpers agreement that forbid applications runnin= g in background, 2. lack of sip clients. Now it seems skype is available on iphones. Has someone tried it ? Along new skype capabilities in Asterisk, could it be used to hook iphones = to Asterisk for both inbound and outbound calls ? Regards --_000_EC80F07C30CE3E46B2AD6B4407BE086F0C2AAD0242cworksmailcwo_

and Asterisk is plugging in pseudo ID. Is that correct? It seems to me that Asterisk should simply say "no caller ID" or "No ID" or something besides "Asterisk". In any case, we are trying to filter them with little success. When we do a LEN(CALLERID(num) we get "13", when we expect "10" The call pattern is 1 call followed by a

adding gsm or just comment out the disallow and the 2 allows. (your recipient is using a codec that isn't ulaw or alaw). _____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel freiha Sent: Wednesday, January 28, 2009 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users]

I got this notion monitor-format = wav49 wav49 presents much louder than regular wav and gsm in my experience -- _____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Lenz Emilitri Sent: Friday, January 22, 2010 8:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users]

used Kamalio to "supplement" the features that Asterisk either doesn't provide or doesn't provide in as nice a form as the OP desired - can't really speak beyond this as I am not one of them. ------=_NextPart_000_010C_01CB6EAA.3AC2C610 Content-Type: text/html; charset="us-ascii" Content-Transfer-Encoding: quoted-printable <html

Moh show files This will show you if your class is set up correctly. ------=_NextPart_000_016C_01CBF83B.306A1A90 Content-Type: text/html; charset="US-ASCII" Content-Transfer-Encoding: quoted-printable <html xmlns:v=3D"urn:schemas-microsoft-com:vml" = xmlns:o=3D"urn:schemas-microsoft-com:office:office" =

adding gsm or just comment out the disallow and the 2 allows. (your recipient is using a codec that isn't ulaw or alaw). _____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel freiha Sent: Wednesday, January 28, 2009 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

adding gsm or just comment out the disallow and the 2 allows. (your recipient is using a codec that isn't ulaw or alaw). _____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel freiha Sent: Wednesday, January 28, 2009 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

major undertaking. But since you are using an AGI to control the Queue command instead of using it from the dialplan, you have more control over this problem than you realize. For simplicity of illustration, let's say your AGI simply wants to take a call and send it to the next agent in the queue. Your Agents are Agent007, AgentQ and AgentM. Because you did the Polycom transfer from

adding gsm or just comment out the disallow and the 2 allows. (your recipient is using a codec that isn't ulaw or alaw). _____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel freiha Sent: Wednesday, January 28, 2009 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

adding gsm or just comment out the disallow and the 2 allows. (your recipient is using a codec that isn't ulaw or alaw). _____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel freiha Sent: Wednesday, January 28, 2009 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

adding gsm or just comment out the disallow and the 2 allows. (your recipient is using a codec that isn't ulaw or alaw). _____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel freiha Sent: Wednesday, January 28, 2009 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

adding gsm or just comment out the disallow and the 2 allows. (your recipient is using a codec that isn't ulaw or alaw). _____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel freiha Sent: Wednesday, January 28, 2009 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

adding gsm or just comment out the disallow and the 2 allows. (your = recipient is using a codec that isn't ulaw or alaw). =20 _____ =20 From: asterisk-users-bounces at lists.digium.com = [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel = freiha Sent: Wednesday, January 28, 2009 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: