Displaying 20 results from an estimated 7000 matches similar to: "Web Client with IAX2 and ilbc"
2005 May 11
2
Sip or IAX2 eb Client
Is there any good IAX2 or SIP free web client? Im looking for something open
source or free preferably IAX2 for integrating into a web site... Any leads?
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
2009 Aug 18
3
IAX2 ActiveX Control
hello,
please any IAX2 ActiveX control that wrap libiax2 or libiaxclient?
i want to develope my softphone in delphi
thanks
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2005 Mar 28
3
call files run at certain times
Im checking the wiki for call files info and seems somebody has a wake up
script that runs call files at certain times.
Do you know if its possible to run a call file by using some other methods
different from cron jobs or at? The wiki mentions that it might be possible
to do this is you modify the files creation date and time by using touch
-f.. Is it true?
The wakeup script creates call files
2005 Jun 05
1
Voice Dtect
Guys, is there any way to detect voice when calling a zap channel? For
example, if you want to send out or playback a recorded message, you need to
wait for somebody to actually answer the phone before playing starts..
Anyway to detect this?
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:
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 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
--
2005 Aug 26
12
IAX2 Softphone Quality & Network Cards
We are in the process of an Asterisk call center deployment using IAX2
G711 ulaw softphones. Outbound sound quality is terrible.
This week we rebuilt the entire LAN with Cisco 2950-EI switches and have
employed QoS on the switches and router. Still sounds terrible.
What we are now finding is that the network card in the PC may be the
key to the problem. A Dell Optiplex P4 2.4GHz 512MB
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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2005 Jun 05
2
TDM400P Polarity reversal detection
Hi
Can TDM400P detect polarity reversal on FXO module?
We have C.O. lines that reverse polarity on Answer and release.
Thank You
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2005 Jan 11
1
Channel IAX2 Socket Read Error
I grabbed the latest sources from CVS yesteday and am having problems
compiling. * v1.0.3 was running previously without issue. I tried
checking out the older source but get the same make errors.
The box is running RH 9. I am getting the following errors. Any
thoughts on what is wrong?
gcc -shared -Xlinker -x -o chan_mgcp.so chan_mgcp.o
gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes