Displaying 20 results from an estimated 6000 matches similar to: "Interview with Mark Spencer"
2004 Dec 10
1
IAXPeers for Windows Beta released
Hi,
I've just done up a quick proggy to show me the status of my IAX peers
from my windows box. It plugs into the simple manager proxy.
You can see more information (including a screenshot) at:
http://www.sineapps.com/news.php?rssid=384
You can download it directly from:
http://www.sineapps.com/down/IAXPeers.zip
Could you please have a look and let me know your thoughts.
--
Cheers,
2005 Jan 17
3
On Hold music
This may sound kind of crazy and I maybe missing something. But are you
placing the call on hold so you can hear the hold music. This may not
be the case but you may have to place the call on hold to here the
music.
Also you mentioned sound, you do not need a sound card in the asterisk
box to use this hold music feature.
Hope this helps.
-----Original Message-----
From:
2004 Sep 22
1
News From Astricon
We've got some replies to questions online about Astricon and we now
have a mirror available at:
http://astricon.voctel.com/news.php
If anyone has any comments about Astricon, please forward them to me
and I will put them up on the site so that all the people who didn't
go can read them.
Cheers,
Matt Riddell
http://www.sineapps.com/news.php (Daily Asterisk News - html)
2006 Jan 18
1
I see Asterisk 1.2.2 into the ftp or was a vision?
Someone can confirm the new release is out?
Haven's seen any post about it!!!!!
--
Adri? Vidal
adriavidal@gmail.com
2005 Aug 13
1
New Beta IAX Statistics Program
Hot off the wire:
http://www.sineapps.com/news.php?rssid=927
Hi, we have put together a small application for Windows to allow you to
check IAX network statistics.
Basically all you need is the .Net framework and the
user/pass/host/extension/context details.
There is one parameter available when you start. This is dial string. It
is made up as follows:
user:pass@host/extension@context
You
2005 Sep 20
6
iax2 trunking wackyness
Hi
I was doing some bandwidth testing, and my incomming usage is
36% more than my outgoing bandwidth.
The setup is IAX2 trunking using GSM codec.
Is there any obvious reason I am overlooking to figure out why
there is such a big difference between the two.?
I am using CVS-head September 3rd, maybe there is a version
skew?
Any suggestions will be appreciated.
Thanks
Clive
2005 Jul 17
2
HOW TO make xten eyebeam incoming video start before you send yours
Sorry, I don't remember who was asking about this, but it seems that if
you record a video message that contains the send video start, it will
actually fire up the remote receive window.
I.E. Previously I was using the recording section of voicemail to create
my video IVR's. This meant that when I arrived at the section to record
the message, I had already clicked send video in Xten.
2005 Sep 21
7
add 0 (zero) to incoming callerID - how?
I have an asterisk box and SIP / IAX2 phones.
To call out, users have to add 0 (zero) before a real telephone number.
That means, that if they want to call someone that has a number 123456,
they have to call 0-123456.
Simple, right?
This has a serious drawback though - when someone calls us from the
number 123456, we see the callerID 123456, and we're unable to use the
callback/redial
2005 May 16
4
Web Client with IAX2 and ilbc
Guys.
Maybe this is asking for a lot :) but is there any web client that can use
IAX2 and ilbc?
This is for a "call us" web idea.... Any leads?
2004 Sep 17
2
Re: Asterisk-Users Digest, Vol 2, Issue 163
Hi Matt,
I have verified with ztmonitor the audio level and it was too low, then
with this the fax machine report "Not Response". I modified the audio level
in zapata.conf and after that the fax machine report "Commnunication Error".
Do you an idea what could be ?
Thanks,
Angel.
> Message: 3
> Date: Sat, 18 Sep 2004 00:48:23 +1200
> From:
2005 May 25
4
SER Help
Hi,
I'm looking for a tutorial or installation guide for
SER to be used with asterisk to solve the remote SIP
agent problem. All the documents available are for
large scale installation.
Any help is highly appreciated.
Regards.
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2004 Sep 21
12
Astricon pictures
Hey,
I am here at Astricon and about to go down to registration. Is there any
interest in pictures if I take my digital camera? I am sure that someone
is already doing this. (Probably someone official). I would take
pictures of each day and upload them to my website if anyone is
interested. Let me know!
--
Kristian Kielhofner
2005 Mar 15
2
Wiki down: Is there another source for documentation?
As the title suggests, I was wondering if there was another source of
documentation for Asterisk.
Related: If one wanted to contribute to documentation, who would one
contact?
Thanks!
Sean
2005 Sep 30
2
Why does the s extension not work in my extensions.conf file
Hello
In my extensions.conf file:
[frompstnisdn]
exten => s,1,Dial(SIP/200&SIP/202,20)
exten => s,2,Voicemail(su200)
exten => s,3,Hangup
I use the s, start, extension to handle incoming calls.
In my zapata.conf:
context=frompstnisdn
This works ok on another asterisk box I setup. But on incoming calls I get:
-- Extension '787367' in context 'frompstnisdn'
2006 Jan 11
3
video development
Hi Fran, you could do it using Adobe/Macromedia Flash Media Server 2,
but I guess that's not the answer you are looking for.
If you manage to do this and release it under GPL I'll kick in $50 for a
bounty.
Regards,
Dean Collins
dean@collins.net.pr
+1-212-203-4357
+61-2-9016-5642 (Sydney in-dial).
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
2005 Sep 30
2
analog phone/door buzzer going through a Sipura SPA2000 ATA dials really slowly
Hello
We have setup a doorbell which has an inbuilt analog phone which is
connected to our Asterisk via a SPA2000 ATA. The problem we are getting is
that when a caller presses the buzzer it is taking two or more minutes to
finally call the reception phone.
In the SPA2000 I have set dtmfmode to be inband.
I notice that with the asterisk you dial a number and then it waits for a
timeout
2005 Jan 13
3
High delay with diax099f + Asterisk
Hi all!
Somebody knows something to do with a high delay using Asterisk + DIAX!?
When I used IAXComm(Linux) in both sides(peer and me) no problems.
Whan I used DIAX099f(WinXP) in both sides(peer and me) I have a delay in the
voice coming from the person that I called. I don't have delay in my voice
to the peer phone.
CODEC: u-law (I tried with all available codecs)
Thanks for your help!
2005 May 25
5
Asterisk Crashing; Not getting Core dumps
This is frustrating. Asterisk has crashed now twice today and neither crash
has produced a core file. My ulimit is unlimited.
I'm using safe_asterisk so asterisk is restarting immediatly, but how the
hell am I suposed to find out wtf happened with no core file? Debug log
doesn't say anything either.
AGRHHHHHHHH
-Matthew
--
2006 May 26
4
mpg123 or asterisk
should I use mpg123 with asterisk 1.2.7 or should i use the native
player asterisk has?
the target machine will receive heavy load.
also, has anyone succedded in compiling mpg123 in a dual core pentium
with centos 4.3 ?
--
-------------------------------------------
Erick Perez
Linux User 376588
http://counter.li.org/ (Get counted!!!)
Panama, Republic of Panama
2005 Aug 28
2
Need quote for Asterisk and billing remote install
Please send me a quote for remote installation of
Asterisk, GUI administration, and billing for calling
card, caller ID based prepaid, and postpaid.
Off list please.
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