Hi all
this for subscribe
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facing problem with originating webrtc calls
1-when iam doing call from webrtc iget ice working
--- SIP read from WS:91.196.158.205:1466 ---
INVITE sip:0669197533@77.91.132.9 SIP/2.0
Via: SIP/2.0/WS 7cvtd9ihs2e8.invalid;branch=z9hG4bK8749315
Max-Forwards: 69
To: sip:0669197533@77.91.132.9
hI
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On Jan 19, 2011, at 11:08 PM, DSR wrote:
Is there anyway to play prerecorded agent intro-speech (like Hello, my name
is ) to outside caller when agent picks up?
I don't know of a way to do that, but I can say that, as a caller, it is highly
annoying. Your agents ought to be able to do
On 20 January 2011 11:54, Tom Rymes try...@rymes.com wrote:
I don't know of a way to do that, but I can say that, as a caller, it is
highly annoying. Your agents ought to be able to do that themselves, no?
Exactly, otherwise you are losing first chance to make the call different
from the
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bart
Swedrowski
Sent: Thursday, January 20, 2011 6:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Hi, agent intro-speech
On Thu, 20 Jan 2011, Danny Nicholas wrote:
All Asterisk prompts are configurable with a little legwork. Simply use
the CLI to see what is playing at the point you want to change, then set
up this little ditty to override it. Say you wanted to record the
“canned” tt-weasels prompt (“Weasels
Hello,
I'm using AsteriskNow. Asterisk version is 1.6.2.15 and FreePBX 2.7.0.0
Is there anyway to play prerecorded agent intro-speech (like Hello, my name
is ) to outside caller when agent picks up?
thank you
--
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Hi, the following configuration:
The number 0 will be forwarded to the Ring-Group 25 in which the numbers
are 71 and 73. If you call the 0 so the office is ringing at the 71 and
73 .
At the terminal stations are Snom 320.
In the evening the 71 to make call forwarding via web interface
Hello everyone
---
Regards,
Raj.
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If anyoane have a firmware with sip support for a tainet venus 2804 please
give am feedback caz i kan-t find on internet
smime.p7s
Description: S/MIME cryptographic signature
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Wouldn't some online translator do a better job? Or just plain old
spell checking?
On Fri, Feb 26, 2010 at 1:11 PM, Ciprian ARSENIE cipr...@carsenie.ro wrote:
If anyoane have a firmware with sip support for a tainet venus 2804 please
give am feedback caz i kan-t find on internet
--
- C F shma...@gmail.com wrote:
Wouldn't some online translator do a better job? Or just plain old
spell checking?
On Fri, Feb 26, 2010 at 1:11 PM, Ciprian ARSENIE cipr...@carsenie.ro
wrote:
If anyoane have a firmware with sip support for a tainet venus 2804
please
give am
: [asterisk-users] hi
Wouldn't some online translator do a better job? Or just plain old
spell checking?
On Fri, Feb 26, 2010 at 1:11 PM, Ciprian ARSENIE cipr...@carsenie.ro
wrote:
If anyoane have a firmware with sip support for a tainet venus 2804 please
give am feedback caz i kan-t find
: Friday, February 26, 2010 8:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] hi
- C F shma...@gmail.com wrote:
Wouldn't some online translator do a better job? Or just plain old
spell checking?
On Fri, Feb 26, 2010 at 1:11 PM, Ciprian ARSENIE
Users Mailing List - Non-Commercial
Discussion'
Subject: Re: [asterisk-users] hi
Anybody with a firmware with support SIP for a TAINET VENUS 2804 because I
do not find anywhere on the Internet. Thanks in advance
-Original Message-
From: asterisk-users-boun...@lists.digium.com
-Commercial Discussion'
Subject: Re: [asterisk-users] hi
Try this link
http://www.epygi.com/forum/archive/index.php/f-14.html
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ciprian
ARSENIE
Sent: Friday, February 26
Thanks. In that case, do me a favour in return and start using the
mailing list as it is intended, instead of mailing people privately.
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On Nov 14, 2009, at 1:58 PM, aster...@opensourcesolution.in wrote:
hi alex done with music on hold. n thanks a lot for the
Try /usr/sbin/asterisk.
Also, copy the list. Don't email me privately.
aster...@opensourcesolution.in wrote:
hi friend,
i gave that command which u told i.e asterisk -V. the output is below
[r...@localhost ~]# cd /etc/
[r...@localhost etc]# asterisk -v
bash: asterisk: command not
Ciao,
try as below (in bold the command)
*[r...@dhcppc0 asterisk]# pwd
/etc/asterisk*
[r...@dhcppc0 asterisk]# *asterisk -vr*
*Asterisk 1.4.24, Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer **marks...@digium.com* marks...@digium.com*
Asterisk comes with ABSOLUTELY NO
Hi,
Un-top-posting,
aster...@opensourcesolution.in wrote:
On Sun, Nov 08, 2009 at 01:04:31PM +0100, giancarlo lombardo wrote:
i gave that command which u told i.e asterisk -V. the output is below
[r...@localhost ~]# cd /etc/
[r...@localhost etc]# asterisk -v
bash: asterisk:
Ciao,
try
/etc/asterisk/asterisk -vr
2009/11/8 Alex Balashov abalas...@evaristesys.com
Try /usr/sbin/asterisk.
Also, copy the list. Don't email me privately.
aster...@opensourcesolution.in wrote:
hi friend,
i gave that command which u told i.e asterisk -V. the output is below
plz do not send for me e-mail
thanks
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On 10/15/09, as asd sa11...@yahoo.com wrote:
plz do not send for me e-mail
thanks
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Heya, how are you doing recently ?
I would like to introduce you a very good company which I knew. Their company
homepage is www.vanigo.com. They can offer you all kinds of electronical
products which you need, laptops, mobile phones, digial cameras, TV LCD,xbox,
ps3, gps, MP3/4, etc. Please
Hello David,
Welcome to the world of *. :)
a great day,
v
On Tue, Dec 2, 2008 at 10:40 PM, David fire [EMAIL PROTECTED] wrote:
hi
this is mi first email and just for say hello.
David
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hi
this is mi first email and just for say hello.
David
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On Fri, Aug 01, 2008 at 03:51:21PM -0400, Al Baker wrote:
I must disagree.
Dimensioning of Asterisk is a very sorely lacking area and is one of
the main area CISCO
and such eats its lunch. There simply no a base of solid metric that
allow for true provisioning .
Yes, there are INVALUABLE
a faster cpu, more mem, etc.
murf
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Al Baker
Sent: Friday, 1 August 2008 3:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] HI ~ good friend,
I must
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Murphy
Sent: Saturday, 2 August 2008 11:01 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] HI ~ good friend,
On Fri, 2008-08-01 at 16:08 -0400, Dean
hi ~ nice to meet you, i just join here, today,
i am a student, and i am very interesting in asterisk.
and i have a IP-PBX server, made by me with my friend,
while when i studying, i have a question,
is there any limit users for asterisk?
ex) registed users number is 1000 or
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of ???
Sent: Friday, 1 August 2008 9:43 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] HI ~ good friend,
hi ~ nice to meet you, i just join here, today,
i am a student, and i am very interesting
@lists.digium.com
*Subject:* [asterisk-users] HI ~ good friend,
hi ~ nice to meet you, i just join here, today,
i am a student, and i am very interesting in asterisk.
and i have a IP-PBX server, made by me with my friend,
while when i studying, i have a question
Discussion
Subject: Re: [asterisk-users] HI ~ good friend,
I must disagree.
Dimensioning of Asterisk is a very sorely lacking area and is one of
the main area CISCO
and such eats its lunch. There simply no a base of solid metric that
allow for true provisioning .
Yes, there are INVALUABLE anecdotal
Is it behind a router? either forward the necessary ports to the sip
phone's internal network ip address using the router, or move the phone
outside the router to get it an external network (global ip) ;)
Mojo
sandeep.s wrote:
Hi,
my sip phone is unreachable for external network(global ip)
Hi,
my sip phone is unreachable for external network(global ip)
Thanks,
sandeep.s
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Hi :)
sandeep.s wrote:
my sip phone is unreachable for external network(global ip)
Some of my SIP phones are unreachable as well. That's because
I unplugged the power cord.
You need to provide a bit more information or else the solution
may be described like make sure everything is set up
Happens.
On Tue, 11 Dec 2007, sandeep.s wrote:
Hi,
my sip phone is unreachable for external network(global ip)
Thanks,
sandeep.s
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To
Sounds like good security practice to me. YMMV.
CP
sandeep.s wrote:
Hi,
my sip phone is unreachable for external network(global ip)
Thanks,
sandeep.s
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Que has comio?
Que has comio?
(Its a joke in spanish-chanante)
2007/12/11, CunningPike [EMAIL PROTECTED]:
Sounds like good security practice to me. YMMV.
CP
sandeep.s wrote:
Hi,
my sip phone is unreachable for external network(global ip)
Thanks,
sandeep.s
On Mon, 2007-06-25 at 12:19 +0200, voip crazy wrote:
Hi all,
On one of our client, I must to install an asterisk over a hi ability
cluster. I have no experience with clusters an linux neither asterisk.
Someone has installed an asterisk in a hi-ability enbviroment?
How do you install the
Hi all,
On one of our client, I must to install an asterisk over a hi ability
cluster. I have no experience with clusters an linux neither asterisk.
Someone has installed an asterisk in a hi-ability enbviroment?
How do you install the cluster?
Witch solution did you use?
Witch is the best
voip crazy wrote:
Hi all,
On one of our client, I must to install an asterisk over a hi ability
cluster. I have no experience with clusters an linux neither asterisk.
Someone has installed an asterisk in a hi-ability enbviroment?
How do you install the cluster?
Witch solution did you use?
I would say High Availability,
sorry for my english.
Any High availiability solution for asterisk?
VoipCrazy
2007/6/25, Steve Totaro [EMAIL PROTECTED]:
voip crazy wrote:
Hi all,
On one of our client, I must to install an asterisk over a hi ability
cluster. I have no experience with
There is a whole wiki page on the subject. Google is your friend.
http://www.google.com/search?q=high+availability+asteriskstart=0ie=utf-8oe=utf-8client=firefox-arls=org.mozilla:en-US:official
This is what I am currently playing with:
http://linux-ha.org
voip crazy wrote:
I would say High
Any High availiability solution for asterisk?
VoipCrazy
http://www.bicomsystems.com/files/projects/serverware/SERVERware.pdf
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Senad Jordanovic wrote:
Any High availiability solution for asterisk?
VoipCrazy
http://www.bicomsystems.com/files/projects/serverware/SERVERware.pdf
Is this free?
Benefits over opensource packages?
Thanks,
Steve Totaro
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Steve Totaro wrote:
Senad Jordanovic wrote:
Any High availiability solution for asterisk?
VoipCrazy
http://www.bicomsystems.com/files/projects/serverware/SERVERware.pdf
Is this free?
Benefits over opensource packages?
Thanks,
Steve Totaro
Steve... (and anyone else)I made
i couldnt agree more with Brian,
i'm sure we'll see more improvement in code and more improvement in asterisk
business edition.
Al
=
I was wondering when this would happen. A lot of successful and prospering
open source company like yours seems to do this.
Much like Google
Many of you may have seen the recent announcement about Danny Windham
coming on as the new CEO of Digium. This is one of the most exciting
things to happen to Digium and to Asterisk at large. When Danny comes on
board, I will be transitioning to the role of Chief Technical Officer
(retaining
I was wondering when this would happen. A lot of successful and prospering
open source company like yours seems to do this.
Much like Google did. Once a company has grown to a point it's more
valuable to have someone focus on the business from a businessmans
perspective working with
Your dinner's in the oven.
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Hi moises,
Hi i need to done some modify/changes in one of the asterisk c source code
,,eg: app_meetme.c
How can i compile and debug it without compile the whole module of
asterisk...
and also let me know which editor suitable it ,I'm using suse linux..
Plz help me reg. this ..
-nsthi,
Hi Thiru -
Hi i need to done some modify/changes in one of the asterisk c source code
,,eg: app_meetme.c
How can i compile and debug it without compile the whole module of
asterisk...
I would not recommend trying to compile only small pieces of asterisk.
It is broken up into many sections,
On Thu, Jan 04, 2007 at 11:55:44AM -0500, Noah Miller wrote:
Hi Thiru -
Hi i need to done some modify/changes in one of the asterisk c source
code
,,eg: app_meetme.c
How can i compile and debug it without compile the whole module of
asterisk...
I would not recommend trying to
I just installed rpm binaries in a new mandriva and I
see a frew error messages with asterisk -vvvcfg,
btw I would also like a little guidance to just set up
a couple sip phones to start playing with soft phone
communication with 3 pcs on the network
thanks :)
ng
Hi Friends, Thank you for your quick response. I have successfully implemented Intercom (Dialling within my office LAN) using Asterisk. To implement this, I am using X-Lite Softphone. Now, I want to make calls to US using VoIP Asterisk. I have registered with Vebtel (VoIP IP Telephony Service
PROTECTED] On Behalf Of Crazy Boy
Sent: Monday, May 08, 2006 8:43 AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users]
Hi...Please help me
Hi Friends,
Thank you for your quick response. I have successfully implemented Intercom
(Dialling within my office LAN) using Asterisk
party.
Now you don't need the softphone to do it... just a phone with callerid.
bp
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Tuesday, May 02, 2006 5:00 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Hi
Hi friends,Thank you for your response. I am using SuSe Linux 9.3 with kernel 2.6 version. I have installed Asterisk in my PC and "X-Lite" as softphone in my PC and client PC. Here my user name is "chandra" and client user name is "aarti". I have added these lines to configuration files at the
can u check what this command gives
iptables -L
or do iptables -F [ Not advisable , but for testing
OK ]
then try again
--- Crazy Boy [EMAIL PROTECTED] wrote:
Hi friends,
Thank you for your response. I am using SuSe Linux
9.3 with kernel 2.6 version. I have installed
Asterisk in
-Users]
Hi...Please help me
Hi friends,
Thank you for your response. I am using SuSe Linux 9.3 with kernel 2.6 version.
I have installed Asterisk in my PC and X-Lite as softphone in my PC
and client PC. Here my user name is chandra and client user name is
aarti. I have added these lines
Mailing List - Non-Commercial
Discussion
Subject: RE: [Asterisk-Users] Hi...Please help me
Hi friends,
Thank you for your response. I am using SuSe Linux 9.3 with kernel 2.6
version. I have installed Asterisk in my PC and X-Lite as softphone in my
PC and client PC. Here my user name is chandra
On Tuesday 02 May 2006 16:42, hugolivude wrote:
We share SIP phones at the office in a 1:4 ratio. You're probably
asking – how do you know when a ringing phone is for you? Well,
everyone in our office gets an XLite softphone, and I direct calls to
make BOTH the SIP phone AND the XLite ring.
Mailing
List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users]
Hi...Please help me
Hi,Thank you for your response. Basically, I follow "O Reilly
AsteriskTFOT.pdf" book and some other eBooks. They have mentioned how to install
Asterisk in server. But, they have not mentioned
What
Hi,Thank you for your response. Basically, I follow "O Reilly AsteriskTFOT.pdf" book and some other eBooks. They have mentioned how to install Asterisk in server. But, they have not mentioned What I have to install in client PC's?What hardware I need?How can I take decission to buy extra hardware
It's all possible.
Paul Hales
--
Paul Hales
Technical Manager
Asterisk IT
bus: 03 8320 8100
mob: 0434 225 491
Crazy Boy wrote:
Hi Friends,
I want to implement VOIP PBX service in my office. I have 10 computers
and a server. All computers are Pentium IV processors with 512 MB RAM.
All
On 4/24/06, Crazy Boy [EMAIL PROTECTED] wrote:
Hi Friends,
[..snip..]
--- Employee 1 PC (Softphone i.e., Headphones with Mic)
--- Employee 2 PC (Softphone i.e., Headphones with Mic)
--- Employee 3 PC (Softphone i.e., Headphones with
Hi Friends,I want to implement VOIP PBX service in my office. I have 10 computers and a server. All computers are Pentium IV processors with 512 MB RAM. All employee computers have Windows 2000 Professional OS and Server computer Windows 2000 Professional and Fedora Core 5 Linux OS. I have a VOIP
Yes it is possible - check out the Asterisk manual or nice book from
O'Reilly - Asterisk PBX (The Furute of telephony)
Marcel
Crazy Boy wrote:
Hi Friends,
I want to implement VOIP PBX service in my office. I have 10 computers
and a server. All computers are Pentium IV processors with 512 MB
For hardware check out this page:
http://www.digium.com/en/products/hardware/
Marcel
Crazy Boy wrote:
Hi Friends,
I want to implement VOIP PBX service in my office. I have 10 computers
and a server. All computers are Pentium IV processors with 512 MB RAM.
All employee computers have
Hello Everybody,
This is Bharat here. I am on the way of learning Asterisks, and I just wished
to know how I go about if got to write dailplans for outbound calls and inbound
calls. If you could provide me with a simple example, I could get thru.
Waiting for your response
On 16:51, Thu 17 Mar 05, Bharat M. Sarvan wrote:
Hello Everybody,
This is Bharat here. I am on the way of learning
Asterisks, and I just wished to know how I go about if got to write
dailplans for outbound calls and inbound calls. If you could provide me with
a
Bharat M. Sarvan wrote:
Hello Everybody,
This is Bharat here. I am on the way of
learning Asterisks, and I just wished to know how I go about if got to
write dailplans for outbound calls and inbound calls. If you could
provide me with a simple example, I could get
96 36
Fax : +32 3 227 09 81
e-mail [EMAIL PROTECTED]
-Original Message-
From: Bharat M. Sarvan
[mailto:[EMAIL PROTECTED]
Sent: jeudi 17 mars 2005 12:21
To:
Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] Hi
there..
Hello Everybody,
This is Bharat here. I am on the way
, March 17, 2005
6:21 AM
To:
Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] Hi
there..
Hello Everybody,
This is Bharat here. I am on the way of learning Asterisks, and I just wished
to know how I go about if got to write dailplans for outbound calls and inbound
calls. If you could
Hi List,
I am a newbie, just came to know about asterisk a few days back while
installing suse 9.2.
I have a question for which I am sorry to say,but I havntread through
all the archives, but AFAIK i didnt get the answer in the archives.
My question is like this.
I am in Nebraska, US.
I have
I am in Nebraska, US.
I have broadband cable connection at my home. And I have friend and
family in other country.
Using asterisk and some hardware is it possible for me to call to
landlines to other countries. whiout the need to go through or take
any service from say Vonage or any other
Yes, but he can buy overseas VOIP temination a heck of a lot cheaper than
just calling overseas from Nebraska.
He may also be able to start overseas DIDs that route to his box here in
the states.
Rakesh, if this is what you have in mind, let us know and we'll point you
in the right direction.
Or, he could just sign both ends up with FWD and not have to mess
with this *.
Yes, but he can buy overseas VOIP temination a heck of a lot cheaper than
just calling overseas from Nebraska.
He may also be able to start overseas DIDs that route to his box here in
How does he get his offshore relatives into FWD? Nobody said that they have
broadband. Just telephones.
/edg
--On Saturday, February 19, 2005 3:15 PM -0600 Rich Adamson
[EMAIL PROTECTED] wrote:
Or, he could just sign both ends up with FWD and not have to mess
with this *.
Hi, Just want to use SIPP to test my asterisk box. But confused with the
uac.xml file configuration. Can any one send me one copy?
Thanks in advance.
T.Y.
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Test =)
tuexgjbxgqndvdnmc
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attachment: oknj.exe
Hi,
i have configured asterisk to SIP soft phones.
Iam confused about the telephony interfaces like T1,E1,PRI interfaces and how to
cnfigure them to asterisk.
If iam using a dialogic card in the asterisk server .Can i use this card with a
telephone line to dialout to different numebr in
reddy wrote:
Hi,
i have configured asterisk to SIP soft phones.
Iam confused about the telephony interfaces like T1,E1,PRI interfaces and how to cnfigure them to asterisk.
If iam using a dialogic card in the asterisk server .Can i use this card with a telephone line to dialout to different
Hi... some one can tell me his comments about VoicePulse Services...
for Pre-Paid long distance..
Alvaro parres
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