Hello Stefan Gofferje,
Am 2012-07-25 22:35:28, hacktest Du folgendes herunter:
can anybody recommend a priceworthy SIP/GSM-gateway that's known to work
flawlessly with asterisk?
Normaly, an USB Stick Huawei K3765-HV would be enough... However, under
Linux I had problems with the stability and
Hi Herve,
Asterisk is legal in India and using it for Fax shouldn't create any issues
as far as legality is concerned.
Look at following link to get some idea on VoIP regulation in India.
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_5_1/ccmfeat/fslopar.html#wp1114625
--Satish Barot
cjwstudios wrote:
Does anyone have any application examples for Confbridge in Asterisk
10?
I'm looking for such examples as well. But just a note, meetme is still
available if you're compiling from source, you just have to enable it.
Doug
--
Ben Franklin quote:
Those who would give up
Hi
I'm thinking about deploying TTS onto our asterisk servers and was just
wondering which ones people use and like...
Thanks
Ish
--
Ishfaq Malik i...@pack-net.co.uk
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w:
Hi/
I am newbe in asterisk.
I try to setup callback with Disa on my home server
Anybody help me, pls
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory
There are lot of TTS it's depends on you which one you like,
flite
festival
google
swift
main things of TTS is it's Voice accent.
On Thu, Jul 26, 2012 at 3:38 PM, Ishfaq Malik i...@pack-net.co.uk wrote:
Hi
I'm thinking about deploying TTS onto our asterisk servers and was just
wondering
On 26/7/12 11:08 am, Ishfaq Malik wrote:
I'm thinking about deploying TTS onto our asterisk servers and was just
wondering which ones people use and like...
We've tried Festival, Cepstral and Ivona.
Ivona was by far and away the best.
If you need free (or very low cost) then your only real
On Thu, 2012-07-26 at 11:40 +0100, Chris Bagnall wrote:
On 26/7/12 11:08 am, Ishfaq Malik wrote:
I'm thinking about deploying TTS onto our asterisk servers and was just
wondering which ones people use and like...
We've tried Festival, Cepstral and Ivona.
Ivona was by far and away the
UK English is exactly what we're after. Did you try flite at all?
No, I wasn't aware of flite when we ran these tests.
Kind regards,
Chris
--
This email is made from 100% recycled electrons
--
_
-- Bandwidth and Colocation
On 26-07-12 12:40, Chris Bagnall wrote:
On 26/7/12 11:08 am, Ishfaq Malik wrote:
I'm thinking about deploying TTS onto our asterisk servers and was just
wondering which ones people use and like...
We've tried Festival, Cepstral and Ivona.
I didn't know about Ivona so thanks for mentioning
Dear all,
I know the topic comes back like boomerang, but I did not find a nice
solution.
Does someone has/knows how to achieve call back on busy otherwise called
camping?
If one is calling the extension and it is busy, then caller should get
something like Press 5 to request call back and
Hello,
is there a way to order call queue members in the database table?
When defining the table for realtime queue_members, I notice there is no
ID-column.
Can I add an ID-column, or will this fail realtime Queues ?
Kind regards,
Jonas.
--
Thank you again Mitul.
Ok, the we ill use EM.
Regards
Jorge Mendoza
- Original Message -
From: Mitul Limbani mi...@enterux.in
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, 25 July, 2012 11:35:08 PM
Subject: Re:
The inherent problem with this is that it either requires a brute force
solution or a queue and call solution. The brute Force solution would
be something like this:
[callee-is-busy]
Exten = s,1,playback(callbackmsg)
Exten = s,n,wait(3)
Exten = s,n,playback(vm-goodbye)
Exten = s,n,hangup()
Try google-ing asterisk disa - help is offered more readily to those who
have tried first.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Sent: Thursday, July 26, 2012 5:37 AM
To:
I know the topic comes back like boomerang , but I did not find a
nice solution.
Does someone has/knows how to achieve call back on busy otherwise
called camping?
If one is calling the extension and it is busy, then caller should
get something like Press 5 to request call back and after the
Hi Doug,
I did find the following on voip-info.
http://www.voip-info.org/wiki/view/Asterisk+cmd+ConfBridge
It's somewhat rudimentary but it does work.
Thanks,
C
On Thu, Jul 26, 2012 at 2:36 AM, Doug Lytle supp...@drdos.info wrote:
cjwstudios wrote:
Does anyone have any application examples
Hi All,
I have an small issue, which is not creating any problem on working syatem
but not sure about the problem that is why eager to know about it. I had
installed Asterisk realtime with Asterisk 1.4.41. Every thing is working
good but getting warning at Asterisk CLI.
[Jul 26 21:17:36]
can you post your sip.conf for Exten. 1000?
it does not seem like you have
[1000]
mailbox=1000@default
Thanks,
-motty
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati
Sent: Thursday, July 26, 2012 10:35
My sip.conf don't have any entry related to sip pees. I have everything
into database.
for more details please check below url, which have good example of
asterisk realtime
http://bahjons.com/stuff/asterisk-realtime-installation-guide
On Thu, Jul 26, 2012 at 11:14 PM, motty.cruz
Apologies for the multiple sends -- I'd been having some outbound SMTP issues,
and thought the first one had fallen into the ether. Turned out, it was the
upstream host that was the issue. Once kicked, lo!
-Ken
On Wed, 25 Jul 2012 14:24:50 -0400 Ken D'Ambrosio k...@jots.org wrote
Hi, all.
On Wed, 2012-07-25 at 16:05 -0400, Ken D'Ambrosio wrote:
Hi, all. I see that, with Asterisk 10, there've been some additions with an
eye toward conferencing, and, apparently, hooks for video conferencing.
Googling like crazy, however, has given me little to go on. I've been tasked
with
Hello guys,
Because I'm using AsteriskNOW and the FREEPBX was automatically installed the
/etc/asterisk changed a little bit, so after read some .conf files I made a
little modification on sip_general_custom.conf inserting the following lines:
videosupport=yes
allow=h263
and then video call
On Tue, Jul 24, 2012 at 3:54 PM, Alejandro Imass a...@p2ee.org wrote:
On Tue, Jun 12, 2012 at 4:04 PM, Alejandro Imass a...@p2ee.org wrote:
we upgraded to 1.8.13.1 and we have much the same problem although after
the upgrade I don't seem to find any cases where the qualify value is
OK (xx ms)
Greetings-
I've got a handful of Polycom IP 550 handsets connected to an Asterisk 1.8.12.0
system. Everything is running smoothly with few problems. However, I have an
issue that maybe someone could shed light on...
Many of the phones have 'buddy watch' enabled for the other phones, basically
Question 1 - I think asterisk only supports a limited set of statuses
Question 2 - you could reset the phone and re-provision it or possibly just
tweak the config file and update it. I have 501's so the 550 is just a WAG.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
On 07/26/2012 03:32 PM, Danny Nicholas wrote:
Question 1 - I think asterisk only supports a limited set of statuses
Asterisk does not *receive* presence updates from Polycom phones (or
really, non-Digium phones) at all. Instead, the presence (status)
updates you are seeing appear on your
On 07/26/2012 04:28 PM, Tim Nelson wrote:
Greetings-
I've got a handful of Polycom IP 550 handsets connected to an Asterisk 1.8.12.0
system. Everything is running smoothly with few problems. However, I have an
issue that maybe someone could shed light on...
Many of the phones have 'buddy
Hello friends,
I am trying to deeply integrate asterisk cdr with voipmonitor cdr. I can
access the caller Call ID (fbasename field in voipmonitor cdr) looking at
the SIPCALLID variable in asterisk, but how can I access from within
asterisk the Call ID of the second leg of the call (the one
On 27/07/2012, at 8:16 AM, Alejandro Imass a...@p2ee.org wrote:
On Tue, Jul 24, 2012 at 3:54 PM, Alejandro Imass a...@p2ee.org wrote:
On Tue, Jun 12, 2012 at 4:04 PM, Alejandro Imass a...@p2ee.org wrote:
we upgraded to 1.8.13.1 and we have much the same problem although after
the upgrade
Hi, one of our asterisk servers recently crashed. Any direction as to how I
can provide helpful information about this issue would be appreciated:
asterisk*CLI core show version
Asterisk 1.8.13.0 built by root @ asterisk on a i686 running Linux on
2012-06-10 22:22:13 UTC
Jul 26 16:39:39 asterisk
On 7/26/2012 6:21 PM, Brandon B. wrote:
Hi, one of our asterisk servers recently crashed. Any direction as to
how I can provide helpful information about this issue would be
appreciated:
If the seg fault happens again, make sure to get a non-optimized
backtrace per the instructions you linked,
On 27/07/2012, at 3:42 AM, Richard Mudgett rmudg...@digium.com wrote:
I know the topic comes back like boomerang , but I did not find a
nice solution.
Does someone has/knows how to achieve call back on busy otherwise
called camping?
If one is calling the extension and it is busy, then
I've posted my problem with ReceiveFax() a long time ago.
Majority of the incoming faxes still end up with a T2 timeout or
hangup (fax session hangup) errors.
Our Setup:
- we're using the Digium Free Fax module for Asterisk, all settings are default
- incoming/outgoing faxes go through an E1 line
25.07.2012 22:24, Ken D'Ambrosio пишет:
Hi, all. I'm 99% sure that Asterisk technically *supports*
videoconferencing
well, confbridge supports sort of videoconferences , but our users
refused to use them because asterisk switches video in the middle of
stream and this leads to broken
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