I am wondering if its possible to have sometime like this:
exten 100 = Dial (g/08039269311)
where g would be a group of SIP extensions and i would be parsing/hard coding
the PSTN numbers into it, so when i dial extension 100, it passes the call to a
group of SIP service provider extensions.
i
need to have a conference with a group of sip phones?
best
On Tue, Feb 15, 2011 at 3:13 PM, ayodele abejide ayodeleabej...@hotmail.com
wrote:
I am wondering if its possible to have sometime like this:
exten 100 = Dial (g/08039269311)
where g would be a group of SIP extensions and i would
I am having problems playing files with the playback command, also with the
Dial (A()) option this is the output from console:
[Feb 12 01:58:41] WARNING[1569]: file.c:650 ast_openstream_full: File
home/abejide/Desktop/a.wav does not exist in any format[Feb 12 01:58:41]
WARNING[1569]:
)
+2348039269311
Before long, paying for a phone call will be as alien as paying for email
Date: Fri, 11 Feb 2011 18:12:19 -0800
From: asterisk@sedwards.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] On-Hold Music
On Sat, 12 Feb 2011, ayodele abejide wrote:
I am having
Dear Asterisk-Users,
I installed festival and while trying to connect it to asterisk it comes up
with:
serverMon Nov 8 18:38:51 2010 : Festival server started on port
1314client(1) Mon Nov 8 18:38:51 2010 : accepted from
localhost.localdomainclient(1) Mon Nov 8 18:38:51 2010 :
It is 1.6.2.13
ABEJIDE, Ayodele A. (CCNA)
+2348039269311
From: asannu...@gmail.com
To: asterisk-users@lists.digium.com
Date: Tue, 9 Nov 2010 07:38:44 -0500
Subject: Re: [asterisk-users] Festival
Hi,
wich version of Asterisk?
If is 1.6.2.13, there is a open issue becouse not
On my own version of sox (14.3.0), says -w option is not allowed
ABEJIDE, Ayodele A. (CCNA)
+2348039269311
Date: Mon, 1 Nov 2010 13:39:43 -0700
From: asterisk@sedwards.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Music On Hold Help
Un-top-posting...
On
Dear Asterisk-Users,
I have this Asterisk Box I run in my house, I need to terminate and originate
remote calls through the box via internet (SIP), the problem is in Nigeria most
ISPs would not provide you with Public Addresses, all they provide is dynamic
Natted addresses which change each
with dynamic IPs.
Regards,
Jonathan
On Tue, Oct 26, 2010 at 12:40 PM, ayodele abejide ayodeleabej...@hotmail.com
wrote:
Dear Asterisk-Users,
I have this Asterisk Box I run in my house, I need to terminate and originate
remote calls through the box via internet (SIP), the problem is in Nigeria
] Mobile Phones and Asterisk
Try http://www.dyndns.com/ that should solve your problem with dynamic IPs.
Regards,
Jonathan
On Tue, Oct 26, 2010 at 12:40 PM, ayodele abejide ayodeleabej...@hotmail.com
wrote:
Dear Asterisk-Users,
I have this Asterisk Box I run in my house, I need to terminate
thanks, i would try all the options out. I am very grateful
_
Hotmail: Trusted email with Microsoft’s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969--
I have asterisk running at home, a friend would be traveling out of the
country and I want him to be able to put a call through from his remote
location, I am wondering how I would configure the X-lite client on his pc so
he would be able to call through assuming my public address is A.B.C.D
I am in Nigeria and I am wondering if I could get a free SIP/DID provider with
regards to my region
Thanks
_
Hotmail: Trusted email with Microsoft’s powerful SPAM protection.
Hi everyone,
I want to Implement asterisk on my company's intranet comprising of 100 local
area networks I don't know how to configure my dialplan and my sip.conf files
Thanks
Date: Sun, 30 May 2010 10:06:12 -0400
From: vene...@gmail.com
To: asterisk-users@lists.digium.com
Subject:
hi,
I did not install sounds during the installation of Asterisk, was wondering if
there is any means through which I can get to install sounds without having to
do a complete re-install of asterisk
Hello group,
I have asterisk running on my ubuntu machine, and I have a
peer to peer network with an XP machine, both of the running x-lite client, I
try
calling either of the soft phone from the other and the response I get is on my
asterisk console is as below:
[May 19 19:31:18]
I am having serious problems connecting my client software to asterisk, i tried
x-lite would not connect, and i tried with twinkle too, it wouldnt, i cannot
get to call myself, i am not on a network, just trying all this out locally,
can i not get to connect without been on a network?
I am a newbie to asterisk, I have a complete installation of
asterisk running on my ubuntu machine and I have x-lite installed also, I would
like to know if I can call myself on the same machine, because whenever I try
to call myself I get an engaged tone. My configuration file settigns are as
I tried port 5061 for my softphone, but the same problem occurs
From: cur...@telecomabmex.com
To: asterisk-users@lists.digium.com
Date: Mon, 22 Mar 2010 17:54:27 -0600
Subject: Re: [asterisk-users] Can I call myself on the same machine
On Mon, 2010-03-22 at 23:40 +, ayodele abejide
I am a complete newbie, completed editing the extensions.conf file, having
problem reloading my diaplan via asterisk console, tried to reload it with
diaplan reload command, but it says command does not exist.
Please help
ayodele abejide wrote:
I am a complete newbie, completed editing the extensions.conf file, having
problem reloading my diaplan via asterisk console, tried to reload it with
diaplan reload command, but it says command does not exist.
Looks like you're missing a letter: dialplan reload
reason you have a 1.2 or older asterisk install, you'll need
to use extensions reload (I think, I don't have a 1.2 box in front of me to
confirm the exact command).
Thanks,--Warren Selby
On Mar 10, 2010, at 6:55 PM, ayodele abejide ayodeleabej...@hotmail.com wrote:
A letter, I spelt it right
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