On Sun, Aug 1, 2010 at 10:34 PM, Janu Mukherjee janu.mu...@gmail.com wrote:
Hi all,
I have the following problem. I want to
Call -- Asterisk AGI Answer -- Create File - Copy File Asterisk
-- Play File -- Finish Call
For now we are using sshfs to map the directories. I now want to achieve
Well, actually we are in contact with quite a number of call-centers that
use the free version - a lot of times it's embedded call centers, like
internal help-desks and such. One of the nicest things of * is that you
would not buy an ACD module for a traditional pbx to support just a couple
of
On Monday 02 Aug 2010, Janu Mukherjee wrote:
Hi all,
I have the following problem. I want to
Call -- Asterisk AGI Answer -- Create File - Copy File Asterisk
-- Play File -- Finish Call
For now we are using sshfs to map the directories. I now want to achieve
this using samba server. I am
Hi,
Is there any Free software that can connect to an Asterisk Server and
Do video Conferencing? or atleast one to one video chat?
thanks
--Siju
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New
If you're only running a 2 agent call center, you could also take a look at http://www.orderlyq.com/asteriskcallcenterstatistics.html - its also free for 2 agents
Rob
On Sat, 31 Jul 2010 15:31:56 -0400, bruce bruce bruceb...@gmail.com wrote:
2 users. So, it's probably never used as a free
Of course i use Wireshark and i see T.38 traffic but it isn't clear to me why
the fax fails every time. I would like to know if there are T.38 tools/plugins
that analyze a .pcap file more thorough.
In my setup the analogue fax device and the ATA are near each other at the
customer site. The
No, from sources version 0.64 it's working fine.
On Sat, Jul 31, 2010 at 10:59 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Fri, Jul 30, 2010 at 07:15:00AM -0400, Fred Posner wrote:
On Jul 30, 2010, at 5:04 AM, Andraž wrote:
Ok, problem is another, when I run configure, it write
Hello list,
Grandstream GXP2010 phone calling to Snom 320, Asterisk in the middle.
Grandstream allows for 8 different codec specifications. I have defined
them as 4 x G726 4 x alaw.
Snom allow for 7 different codec specifications. I have defined them as
3 x G726 4 x G729.
The SIP peers
Hello,
I would like to know whether there is a way to associate a TV media server
with Asterisk. Is it possible to access TV Chanels in the Telephone Sets.
Anybody have any tips or documents related to this please let me know.
Thanks
--
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Janu Mukherjee
Subject: [asterisk-users] how to place a call on hold and play music on
holdusing agi
I want to originate a call using asterisk agi. I could this. I now want to
place this
Hi All,
Is there a way to change the mappings of disconnect reasons to certain SIP
messages? E.G. I need to change the mapping for SIP 402 “Payment Required” from
16 (normal termination) like it is in 1.4.24 to 21 (call rejected) as defined
in RFC 3398. For me this is a big issue because my
hello,
i just subscribed to this list, i discovered asterisk and i would like
to try it at home on my personal pc.
the computer is a p4 at 3 ghz with 2 gb ram and 80 gb hdd, a 1 Mbit
guarranted connection and runs a gentoo linux.
i search about digium products but i can't find them in my area
On Mon, 2 Aug 2010, Daniel Petre wrote:
hello,
i just subscribed to this list, i discovered asterisk and i would like
to try it at home on my personal pc.
the computer is a p4 at 3 ghz with 2 gb ram and 80 gb hdd, a 1 Mbit
guarranted connection and runs a gentoo linux.
i search about
Hi Daniel,
have a look at this page, maybe it will help you find a reseller:
http://www.voip-info.org/wiki/view/Asterisk+Consultants+Romania .
Best Regards,
Alex
-Messaggio originale-
Da: asterisk-users-boun...@lists.digium.com
On 08/02/2010 02:34 AM, Siju George wrote:
Hi,
Is there any Free software that can connect to an Asterisk Server and
Do video Conferencing? or atleast one to one video chat?
One to one video chat is already supported by Asterisk, using SIP or
H.323 video phones.
--
Kevin P. Fleming
Digium,
On Mon, Aug 2, 2010 at 5:37 AM, Tino t...@sparksupport.com wrote:
Hello,
I would like to know whether there is a way to associate a TV media server
with Asterisk. Is it possible to access TV Chanels in the Telephone Sets.
Anybody have any tips or documents related to this please let me
Sorry, I am a newbie to this concept. Can you please briefly explain how it
is possible to watch TV channels using a video phone by just dialing a
number. Is there any website links that you can share with me on this
subject ? . Thanks for your interest in this matter.
On Mon, Aug 2, 2010 at
On Mon, Aug 2, 2010 at 8:37 AM, Tino t...@sparksupport.com wrote:
Anybody have any tips or documents related to this please let me know.
http://www.youtube.com/watch?v=3h6-PSpD-Oc
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
Hi all,
I am just trying to implement DUNDi-Routing like described here
http://www.voip-info.org/wiki/view/DUNDi+Enterprise+Configuration+SIP+with+no+passwords
and have a most probably stupid question:
My config is exactly like described except that instead of
exten =
Hi all,
Can some one suggest me an IAX client for Linux and Windows?
I used KIAX once, but know it seems complicated to have it working on Ubuntu.
Thanks.
Ronaldo.
--
_
-- Bandwidth and Colocation Provided by
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
unsero...@aol.com
Subject: [asterisk-users] Stupid Macro question
Hi all,
I have
exten = _X.,1,Macro(dundi-priv,${EXTEN})
exten = _X.,2,DIAL(CAPI/contr1/${EXTEN})
Now my
On 02/08/10 17:35, Ronaldo Zacarias Afonso wrote:
Hi all,
Can some one suggest me an IAX client for Linux and Windows?
I used KIAX once, but know it seems complicated to have it working on Ubuntu.
This one is great on Ubuntu/Linux. http://www.sflphone.org/
Unfortunately I know not about
Is a mail server built in in asterisk now
Like in elastix
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
Hi all,
I have
exten =_X.,1,Macro(dundi-priv,${EXTEN})
exten = _X.,2,DIAL(CAPI/contr1/${EXTEN})
Now my problem is, thatafter hanging up a call, the call is instantly
re-established using theh-extension which is almost a loop.
I am sure this is astupid question, but what am I doing
I use http://www.voixphone.com/
On Mon, Aug 2, 2010 at 9:41 PM, Alan Lord (News) alansli...@gmail.comwrote:
On 02/08/10 17:35, Ronaldo Zacarias Afonso wrote:
Hi all,
Can some one suggest me an IAX client for Linux and Windows?
I used KIAX once, but know it seems complicated to have it
On Mon, Aug 2, 2010 at 9:56 AM, mattias m...@mjw.se wrote:
Is a mail server built in in asterisk now
Like in elastix
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for
Hi Everyone,
Sorry, if it's not directly related to Asterisk. Some of people on this list
might have PBX deployed for their clients. What software do you use to
invoice them so the invoice looks like a proper telecom invoice maybe?
Prefer:
-opensource with Windows binary available.
-able to
On Mon, 2010-08-02 at 14:26 -0400, bruce bruce wrote:
Hi Everyone,
Sorry, if it's not directly related to Asterisk. Some of people on
this list might have PBX deployed for their clients. What software do
you use to invoice them so the invoice looks like a proper telecom
invoice maybe?
On Mon, 2 Aug 2010, Ronaldo Zacarias Afonso wrote:
Hi all,
Can some one suggest me an IAX client for Linux and Windows?
I used KIAX once, but know it seems complicated to have it working on Ubuntu.
Thanks.
www.Zoiper.com
Gordon
--
On Mon, 2 Aug 2010, bruce bruce wrote:
Hi Everyone,
Sorry, if it's not directly related to Asterisk. Some of people on this list
might have PBX deployed for their clients. What software do you use to
invoice them so the invoice looks like a proper telecom invoice maybe?
Prefer:
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alejandro
Imass
Subject: [asterisk-users] FAX Options
Is FAXing with Asterisk a practical option ? Or is it better just to
use a plain fax connected to an FXS and just switch with
Hi,
Is FAXing with Asterisk a practical option ? Or is it better just to
use a plain fax connected to an FXS and just switch with Asterisk. I
specifically wanted to know if there was any experience using just the
fax scanner to send faxes and receive them via asterisk and the to
e-mail. My idea
Anyone know where the sources for app_nv_faxdetect officially live? I
couldn't turn them up on a web search, just patched versions for 1.4, etc.
Thanks.
--
_
-- Bandwidth and Colocation Provided by
Thanks for your reply.
My configuration is correct. It works with ssh: many attacks have been
stopped. Also, the config has worked for asterisk one time: I have seen that
in the fail2ban.log file.
--
_
-- Bandwidth and
Is anyone aware of a GSM femtocell that will trunk back to a VoIP softswitch
such as Asterisk?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar
On Mon, Aug 2, 2010 at 3:36 PM, Matt mhop...@gmail.com wrote:
Is anyone aware of a GSM femtocell that will trunk back to a VoIP
softswitch such as Asterisk?
I have not, but I have had great luck with OpenBTS.
Thanks,
Steve T
--
Hi list,
I'm having a problem with CallerID names not showing up when calls come
in. I have dialplan code to store the callerid(name) away and it is
blank (null). However, the voicemail variable ${VM_CALLERID} has the
name field populated. For example, here is some of the dialplan code:
2.
On Mon, Aug 2, 2010 at 3:53 PM, Steve Totaro stot...@totarotechnologies.com
wrote:
On Mon, Aug 2, 2010 at 3:36 PM, Matt mhop...@gmail.com wrote:
Is anyone aware of a GSM femtocell that will trunk back to a VoIP
softswitch such as Asterisk?
I have not, but I have had great luck with
Hi Group,
short question. is it possible to use
#include asterisk/alaw.h instead of #include asterisk/ulaw.h
in app_meetme.c or is ulaw required in meetme?
thanx for the answer.
Daniel
--
_
-- Bandwidth and
On Mon, Aug 2, 2010 at 3:03 PM, Danny Nicholas da...@debsinc.com wrote:
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alejandro
Imass
Subject: [asterisk-users] FAX Options
[...]
TIA,
Alejandro Imass
IMO, as long as you're
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alejandro
Imass
Subject: Re: [asterisk-users] FAX Options
Could you recommend a good starting point? Like a faxing with Asterisk
how-to...
I would personally get the Free Fax for Asterisk.
On Mon, Aug 2, 2010 at 2:56 PM, Cassius Smith cass...@cassius.org wrote:
Any ideas?
THANKS
Cassius
Add a Wait(2) before your first Set statement. Sometimes callerid takes a
few seconds to arrive over the line, depending on your technology.
--
Thanks,
--Warren Selby
Thanks Warren. That fixed it.
I am using T1's and didn't think the spill would take that long.
Ciao,
Cassius
Add a Wait(2) before your first Set statement. Sometimes callerid
takes a
few seconds to arrive over the line, depending on your technology.
--
Un-top-posting...
On Mon, 2 Aug 2010, Cassius Smith wrote:
I'm having a problem with CallerID names not showing up when calls come
in.
On Mon, 2 Aug 2010, Warren Selby wrote:
Add a Wait(2) before your first Set statement. Sometimes callerid
takes a few seconds to arrive over the line,
On Mon, Aug 02, 2010 at 03:36:59PM -0400, Matt wrote:
Is anyone aware of a GSM femtocell that will trunk back to a VoIP
softswitch such as Asterisk?
Most people seem to be concentrating on 3G femtocells (there are various
companies making designs based on picoChip soft radios).
OpenBTS
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Alejandro Imass
Sent: Monday, August 02, 2010 9:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] FAX Options
On 30/06/10 1:53 AM, bruce bruce wrote:
Hi Everyone,
I am accustomed to PUTTY and it's very nice as in it allows many many
SSH profiles to be saved and allows tunneling etcbut it's not very
good when it comes to scrolling up and down, colors, text size, and
specially it doesn't give a
On 16/07/10 4:40 AM, Gilles wrote:
Hello
I'd like to write a script that would make it easier for people to
call in, listen to the IVR, and make an appointment (eg. When? ASAP?
A given day? - Morning? Afternon, etc.)
I assume I'm not the first one to try and write this type of IVR, so
On Mon, Aug 2, 2010 at 7:26 PM, Mark Scholten m...@streamservice.nl wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Alejandro Imass
Sent: Monday, August 02, 2010 9:00 PM
To: Asterisk Users Mailing
Hi all,
My latest Asterisk system is based on Debian squeeze with Asterisk
1.6.2.6-1 and SIP only. One of my favorite features that I had working
with Asterisk 1.4 is the PrivacyManager. However, this was not
straightforward, because anonymous SIP calls arrive with
${CALLERID(num)} =
I am using T1's and didn't think the spill would take that long.
PRI no, EM yes.
Some PRI take that long too because the telco sends the name in a followup
message, not in the initial call setup.
--
_
-- Bandwidth and
Hello.
I'm looking to buy a FXO card to do some testing with two phone lines I have at
home and was looking in ebay some and found some cheap ones but, the I've never
heard of the brand or manufacturer: chinaroby. They run for about $99 plus
shipping. Have any one used these? or please
Try removing the quotes in your n(true) priority.
Thanks,
--Warren Selby
On Aug 2, 2010, at 7:40 PM, Jaap Winius jwin...@umrk.nl wrote:
Hi all,
My latest Asterisk system is based on Debian squeeze with Asterisk
1.6.2.6-1 and SIP only. One of my favorite features that I had working
with
- Mark Scholten m...@streamservice.nl wrote:
Here we have the following setup, could you say if that is acceptable
for
you?
Outgoing fax:
Fax - Linksys pap2t (sip, no t38, for settings see
http://www.provu.co.uk/pdf/sipura/ip_faxing_sipura_linksys.pdf) -
asterisk
- sip trunk provider
On Mon, Aug 2, 2010 at 12:15 PM, mosbah abdelkader
mosbah.abdelka...@gmail.com wrote:
Thanks for your reply.
My configuration is correct. It works with ssh: many attacks have been
stopped. Also, the config has worked for asterisk one time: I have seen that
in the fail2ban.log file.
--
Quoting Warren Selby wcse...@selbytech.com:
Try removing the quotes in your n(true) priority.
From FAILED? That makes no difference: with or without the quotes,
the result is always 0, which leads in the Dial() rule being executed.
Actually, though, that's not even relevant, because before
Maybe good but the first look brought me to a Pay version. Doesn't satisfy
the opensource condition.
thanks,
On Mon, Aug 2, 2010 at 2:39 PM, Jeff LaCoursiere j...@sunfone.com wrote:
On Mon, 2010-08-02 at 14:26 -0400, bruce bruce wrote:
Hi Everyone,
Sorry, if it's not directly related
Sorry, I am not familiar with them.
Wondering if any full package system out there does the job.
Thanks
On Mon, Aug 2, 2010 at 2:55 PM, Gordon Henderson
gordon+aster...@drogon.netgordon%2baster...@drogon.net
wrote:
On Mon, 2 Aug 2010, bruce bruce wrote:
Hi Everyone,
Sorry, if it's not
You forgot to say for free
--Don
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bruce bruce
Sent: Monday, August 02, 2010 10:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bruce bruce
Sent: Tuesday, 3 August 2010 1:58 PM
To: j...@sunfone.com; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] What do you use for Invoicing?
hi,
I am using this card and IP phone about 6 months. There is no issues at all.
Installation procedures are same as Digium analog card.
Hope it helps,
Ashik
On Tue, Aug 3, 2010 at 6:28 AM, Landy Landy landysacco...@yahoo.com wrote:
Hello.
I'm looking to buy a FXO card to do some testing
61 matches
Mail list logo
