Hi Matt,
Can an Asterisk server hold logins for multiple Japper accounts on a
remote Jabber server, and carry multiple Jabber calls simultaneously the
way it can carry multiple SIP (or IAX, or ZAP, etc) calls? If so, is
each of those Jabber calls as lightweight as, say, each SIP call?
Hi all,
I read on serveral pages that chan_spy is not part of asterisk anymore
as on http://www.voip-info.org/wiki-Asterisk+cmd+ChanSpy on the bottom
of this page.
I have a testing server with debian-testing and debian packages for
asterisk installed. In the modules directory
Hi,
in december last year I posted the following problem:
QUOTE
When I dial the number of our client, located in another town, I get a
connection to the asterisk server, I can talk to my client or listen to
his mailbox.
If someone in the town of this client calls him, he gets the ISDN error
Hello Olivier,
from what I've understood Asterisk (or its plugin) should act as MRCP
client.
I copy and paste from RFC 4463 (hope it doesn't brake through email
exchange):
||
||--|| |--|
|| Application
On 1/14/08, Jan-Hendrik Palic [EMAIL PROTECTED] wrote:
Hi all,
I read on serveral pages that chan_spy is not part of asterisk anymore
as on http://www.voip-info.org/wiki-Asterisk+cmd+ChanSpy on the bottom
of this page.
I have a testing server with debian-testing and debian packages for
2008/1/14, Cavalera Claudio Luigi [EMAIL PROTECTED]:
Hello Olivier,
from what I've understood Asterisk (or its plugin) should act as MRCP
client.
I agree with you as Asterisk's TTS or ASR plugins (Lumenvox ASR module for
instance, see http://www.lumenvox.com/products/speech_engine/), should
Hi,
Using Asterisk 1.4.17. I'm calling a PHP script through AGI. No matter
what my script returns (0 or -1), AGISTATUS always appears to be 0 =
SUCCESS.
I was wanting my script to be able to return a value to the dialplan and
then test AGISTATUS but it looks like I'm going down the wrong path.
Hi,
I use asterisk 1.4.11 version for making outbound calls. Running it on
linux(fedora core 7) machine. Recently purchased the g729 codec, got it
registered with my asterisk box. I have two queries for you to help me.
1. How do i know when an outbound call is placed that it makes use of the
Hi List;
With new technolgy, alot of mobiles now support Video
Call, so what is the possibility to have Asterisk
supporting Video so it support Video call at theie
Phones?
Regards
Bilal
Looking for last
On Jan 14, 2008 7:04 AM, Naveen Palani [EMAIL PROTECTED] wrote:
Hi,
I use asterisk 1.4.11 version for making outbound calls. Running it on
linux(fedora core 7) machine. Recently purchased the g729 codec, got it
registered with my asterisk box. I have two queries for you to help me.
1. How
Naveen Palani wrote:
Hi,
I use asterisk 1.4.11 version for making outbound calls. Running it on
linux(fedora core 7) machine. Recently purchased the g729 codec, got it
registered with my asterisk box. I have two queries for you to help me.
1. How do i know when an outbound call is
Hi,
Atis Lezdins schrieb:
Well, it is bundled in asterisk. If you pay more attention, it shows:
* As of october 19 2004, ChanSpy is not included in the standard
Asterisk distribution or the development CVS tree.
* As of March 22 2005 this bug was reopened and can be viewed at above
Hi List;
Is there an Digium cards support GSM SIM cards so we
can fix an SIM card to be used for calls within
mobiles as it is less rate?
Or I have to use an FXS to SIM adaptor? If yes, then
anyone advise a models and prices?
Regards
Bilal
Or I have to use an FXS to SIM adaptor? If yes, then
anyone advise a models and prices?
I'll leave others to suggest models and prices, but in my experience, you're
better with a SIP-GSM gateway and cutting out the analogue loop altogether.
They're a bit more expensive, but it'll save you
On Mon, Jan 14, 2008 at 08:20:11AM -0500, Steve Totaro wrote:
Check chan_mobile for bluetooth enabled phones. This is an extremely cool
app. The phone can be used as an FXO and/or an extension. You can also use
bluetooth headsets as FXSs.
AFAIR, the main limitation of it was that you could
On Jan 14, 2008 7:42 AM, bilal ghayyad [EMAIL PROTECTED] wrote:
Hi List;
Is there an Digium cards support GSM SIM cards so we
can fix an SIM card to be used for calls within
mobiles as it is less rate?
Or I have to use an FXS to SIM adaptor? If yes, then
anyone advise a models and prices?
2008/1/9, Johansson Olle E [EMAIL PROTECTED]:
9 jan 2008 kl. 02.48 skrev Raj Jain:
This issue of phone vendors not supporting OPTIONS according to RFC
3261
often comes up on this list. Like Kevin Fleming said, an OPTIONS
request is
supposed to be responded in the same way as an
14 jan 2008 kl. 09.46 skrev Philippe Sultan:
Hi Matt,
Can an Asterisk server hold logins for multiple Japper
accounts on a
remote Jabber server, and carry multiple Jabber calls
simultaneously the
way it can carry multiple SIP (or IAX, or ZAP, etc) calls? If so, is
each of
On Mon, 14 Jan 2008, bilal ghayyad wrote:
Hi List;
Is there an Digium cards support GSM SIM cards so we
can fix an SIM card to be used for calls within
mobiles as it is less rate?
Or I have to use an FXS to SIM adaptor? If yes, then
anyone advise a models and prices?
A few solutions - If
On Mon, Jan 14, 2008 at 08:20:11AM -0500, Steve Totaro wrote:
On Jan 14, 2008 7:42 AM, bilal ghayyad [EMAIL PROTECTED] wrote:
Is there an Digium cards support GSM SIM cards so we
can fix an SIM card to be used for calls within
mobiles as it is less rate?
Or I have to
This possibly isn't 100% asterisk related, but I'd like some
opinions/feedback...
A customer wanted different ring-tones to differentiate external and
internal calls. No biggie once I'd worked out that details - they have
100% GXP2000 phones, so adding in the relevant SIP header and altering
On Mon, Jan 14, 2008 at 01:25:13PM -, Chris Bagnall wrote:
Or I have to use an FXS to SIM adaptor? If yes, then
anyone advise a models and prices?
I'll leave others to suggest models and prices, but in my experience,
you're better with a SIP-GSM gateway and cutting out the analogue
Julian Lyndon-Smith wrote:
Lee Jenkins wrote:
Julian Lyndon-Smith wrote:
astmanproxy does this already, I think ..
Julian.
Of course ;) AstManProxy is a great product from what I had read up on it.
One thing is that it requires (if I'm not mistaken) an mysql installation
which
is too
I have this in my extension.conf:
[incoming_28345474]
; 8862100 is the hotline number of the Welltech 3804
;
exten = 8862100,1,NoOp(${CALLERID(num)})
exten = 8862100,2,Wait(1)
exten = 8862100,3,Set(CALLERID(num)=${CALLERID(num)})
include = fax2emailstart
[fax2emailstart]
exten =
On Sat, 12 Jan 2008, Antonios Tsakiridis wrote:
Hello everyone,
I'm using very simple setup to make and receive external ISDN calls through a
softphone (x-lite version 3.0 - Win32) via an asterisk box.
Hardware setup:
- Dialogic Diva BRI (lspci yields: Network controller: Eicon
Technology
With the meetme r option, I know it can record a conference in wav
format. As this format eats up space like none other, is it possible to
have it record in gsm or wav/gsm format to keep it down in size?
Rob
___
-- Bandwidth and Colocation Provided by
thanks for the reply
I'm already on 1.4.7.1
regards
Robb
Ed Nunez wrote:
I had the same issue and updated my Zaptel drivers to version 1.4.17 and
it's rebooting fine now.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of robert
boardman
Sent:
Hello,
sorry for beeing off-topic here. But can anyone confirm that
there is a problem reverse resolving lists.digium.com (216.207.245.17) ?
Because of this problem, my mail server will not accept mails from
lists.digium.com (it is configured to accept valid DNS only).
Armin
Hello List,
I'm trying to help a family from the Dominican Republic and to do so need a
temporary DID from DR.
The short story is that there is a 2 year old here with a serious heart defect
from a remote area of Dominican Republic near the Haitian border. He was
referred to Gift of Life D.R.
On Jan 14, 2008 8:28 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Mon, Jan 14, 2008 at 08:20:11AM -0500, Steve Totaro wrote:
Check chan_mobile for bluetooth enabled phones. This is an extremely
cool
app. The phone can be used as an FXO and/or an extension. You can also
use
bluetooth
I am trying to record a call into a stereo mp3 in Asterisk 1.4, but I can't
seem to get it to work correct. Could someone point me to what I need to do?
I have attached what I believe are the relevant parts.
[globals]
; script to be executed when monitoring has been finished
Hello
Same problem here. That could explain why I'm asked for a subscription
confirmation every day due to excessive bounces only for this list.
Gaetan
;; Got SERVFAIL reply from 195.238.2.21, trying next server
;; Got SERVFAIL reply from 195.238.2.21, trying next server
Server:
On Mon, Jan 14, 2008 at 01:43:54PM +, Gordon Henderson wrote:
I'd presonally avoid any sort of bluetooth solution in anything resembling
a commercial environment. Give me a bit of wire, anyday!
You've asked for it:
http://www.celliax.org/
P.S: you may have to solder the wire yourself
On Jan 14, 2008 8:43 AM, Gordon Henderson [EMAIL PROTECTED]
wrote:
On Mon, 14 Jan 2008, bilal ghayyad wrote:
Hi List;
Is there an Digium cards support GSM SIM cards so we
can fix an SIM card to be used for calls within
mobiles as it is less rate?
Or I have to use an FXS to SIM
Yes I had to whitelist their mail servers because their reverse DNS
disappeared sometime last week...
Armin Schindler wrote:
Hello,
sorry for beeing off-topic here. But can anyone confirm that
there is a problem reverse resolving lists.digium.com (216.207.245.17) ?
Because of this problem,
Boa tarde
Estou procurando por algum aplicação que faça a checagem de voicemail, e caso
exista uma nova mensagem, ele disque para o ramal que recebeu a mensagem.
Alguém já viu algo parecido?
Achei um tal de MWI mas ele aparentemente não funciona para caixas de mensagens
individuais.
Obrigado
You might take a few ideas from this combine.sh script which works for
me. It uses the combine_wave program from
http://panteltje.com/panteltje/dvd/combine_wave-0.3.tgz and the lame
program to convert to mp3.
It converts the entire directory /var/spool/asterisk/monitor/*-in.wav
files to mp3
On Mon, 14 Jan 2008, Gaëtan Minet wrote:
Hello
Same problem here. That could explain why I'm asked for a subscription
confirmation every day due to excessive bounces only for this list.
That is exactly the problem I have. I did a whitelisting for the moment too.
It seems that the lower
Stefan Guenther wrote:
QUOTE
When I dial the number of our client, located in another town, I get a
connection to the asterisk server, I can talk to my client or listen to
his mailbox.
If someone in the town of this client calls him, he gets the ISDN error
service not available.
With
On Mon, 2008-01-14 at 08:35 -0500, David Boyd wrote:
Maybe Digium doesn't care if they lose community support as they have been
successful in bringing in investment that will carry them forward as they
deploy more commercial product
Absolutely not. Digium cares *very deeply* about the
Hi all
Someone knows how can I do to send any notify to user, when he received a new
message in your mailbox on voicemail?
Thanks for any help
--
Gilberto Nunes
Itajaí - SC
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
On 1/14/08, Gilberto Nunes [EMAIL PROTECTED] wrote:
Boa tarde
Estou procurando por algum aplicação que faça a checagem de voicemail, e
caso
exista uma nova mensagem, ele disque para o ramal que recebeu a mensagem.
Alguém já viu algo parecido?
Achei um tal de MWI mas ele aparentemente não
A Monday 14 January 2008 16:01:27, Shane D escreveu:
Yes, i can!
In fact, I really do! :-)
Sorry!
thanks
On 1/14/08, Gilberto Nunes [EMAIL PROTECTED] wrote:
Boa tarde
Estou procurando por algum aplicação que faça a checagem de voicemail, e
caso
exista uma nova mensagem, ele disque
A Monday 14 January 2008 14:51:43, Gilberto Nunes escreveu:
Hi all
Someone knows how can I do to send any notify to user, when he received a new
message in your mailbox on voicemail?
Thanks for any help
--
Gilberto Nunes
Itajaí - SC
___
--
The user will receive email notification if you have configured the
user's email address in /etc/asterisk/voicemail.conf .
See: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+voicemail.conf
Also check the externnotify option which lets you run an external
script when new voicemail
Armin Schindler wrote:
sorry for beeing off-topic here. But can anyone confirm that
there is a problem reverse resolving lists.digium.com (216.207.245.17) ?
Our IT department reports that this has been corrected.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - The
A Monday 14 January 2008 16:25:15, Steve Johnson escreveu:
Yeah! I'm just do this right now!
But I want more!
How can I create some extension to call to user, and pass the information about
new voicemail message?
Thanks
The user will receive email notification if you have configured the
On Mon, 14 Jan 2008, Kevin P. Fleming wrote:
Armin Schindler wrote:
sorry for beeing off-topic here. But can anyone confirm that
there is a problem reverse resolving lists.digium.com (216.207.245.17) ?
Our IT department reports that this has been corrected.
Sorry, but I cannot confirm that.
Hi
The authoritative apid.com nameservers still don't reply correctly so
I don't think something better will propagate soon.
(and there is no ns record for the 3-63.245.207.216.in-addr.arpa
subzone in which 17.245.207.216.in-addr.arpa is cname'd)
Gaetan
On 14/01/2008, at 19:34, Kevin
On Mon, 14 Jan 2008, Steve Totaro wrote:
On Jan 14, 2008 8:43 AM, Gordon Henderson [EMAIL PROTECTED]
wrote:
On Mon, 14 Jan 2008, bilal ghayyad wrote:
Hi List;
Is there an Digium cards support GSM SIM cards so we
can fix an SIM card to be used for calls within
mobiles as it is less rate?
Does what I have in the dialplan look right or am I way off base with being
able to use that script?
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: Steve Johnson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; Asterisk Users Mailing List
2008/1/14 Gilberto Nunes [EMAIL PROTECTED]:
A Monday 14 January 2008 16:01:27, Shane D escreveu:
Yes, i can!
In fact, I really do! :-)
Sorry!
thanks
Sorry this is an English-only list. Have you tried asteriskbrasil.org?
--
Kristian Kielhofner
Here's what I would suggest. You should insert some NoOp() statements
and watch the CLI as you dial your 555 extension so that you can see
whether it's working or not.
Your example (which you mentioned you want to run under Asterisk 1.4):
[test]
exten =
The vast majority of what I've done with Asterisk has been with the
Grandstream GXP-2000's. These phones work great for us for everything
*except* speaker quality is quite poor and appears to be half-duplex.
So now that we've bought and are using 40 GXP-2000's we're doing some
testing on other
Having just gotten into this today, here's what I got for Grandstream,
Linksys Polycom phones to all work:
exten = _7XX,1,SIPAddHeader(Call-Info: sip:\;answer-after=0)
;this works for Linksys Grandstream
exten = _7XX,n,SIPAddHeader(Alert-Info: Ring Answer) ;this
works for
I think what you need is a GSM Gateway
You can find a cheap one at cyber-telecom.net
The model you should be looking for is CT-G1000 or 2000
Plug the SIM in there and it will give you a RJ11 telephone port where you
plug into something like X100P then you are ok to go..
Sam
-Original
Paul Hales wrote:
Maybe ADDQUEUEMEMBER(queue1|${CALLERID(num)) is closer to what you are
looking foror ADDQUEUEMEMBER(queue1) - without the pipe...
the construction
exten = 6662,1,ADDQUEUEMEMBER(queue1|SIP/${ID${CALLERID(num)[EMAIL PROTECTED])
was okay for my szenario but the @local was
Hey all, when you guys have requests from clients to block their CID
from showing through, what are others doing? I had a coworker throw in
some Name Here0 garbage which none my carriers like. I don't want to
do Private12345678910 so any suggestions.
--
Asterisk 1.2.24 seems to crash repeatedly under any substantial call load
(and sometimes without a substantial call load - just one SIP leg is
enough to do it) when using the G.729 pre-compiled binaries from:
http://asterisk.hosting.lv/
As per:
Guys, anybody has a 1.2.x compatible app_voicemail patched for Spanish
prompts that can handle for example, instead of saying trabajo mensjes
would say mensajes de trabajo o mensajes trabajo (inverse)? Also can
handle singular and plural (mensaje vs. mensajes)?
Anton
On Jan 14, 2008 5:09 PM, Alex Balashov [EMAIL PROTECTED] wrote:
Asterisk 1.2.24 seems to crash repeatedly under any substantial call load
(and sometimes without a substantial call load - just one SIP leg is
enough to do it) when using the G.729 pre-compiled binaries from:
On Mon, 14 Jan 2008, Steve Totaro wrote:
I would suggest building it yourself (
http://www.readytechnology.co.uk/open/ipp-codecs/doc-svn6.txt). It is not
that difficult and ensures that it should be compatible with your
machine. Just a little work.
That was what I initially tried to do,
Hi All,
We updated with Asterisk 1.4.17 but it seems unstable. 3, 4 times in one day it
stop to response to the SIP Clinets so they cannot make call or register. But
safe_asterisk not restarting it back because asterisk running without any
response to the sip clients.
When we try to do 'core
On Jan 14, 2008 5:55 PM, Alex Balashov [EMAIL PROTECTED] wrote:
On Mon, 14 Jan 2008, Steve Totaro wrote:
I would suggest building it yourself (
http://www.readytechnology.co.uk/open/ipp-codecs/doc-svn6.txt). It is
not
that difficult and ensures that it should be compatible with your
The 'setcallerpres' application is the one to use...
PaulH
On Mon, 2008-01-14 at 17:18 -0500, J. Oquendo wrote:
Hey all, when you guys have requests from clients to block their CID
from showing through, what are others doing? I had a coworker throw in
some Name Here0 garbage which none my
On Jan 14, 2008 5:51 PM, Steve Totaro [EMAIL PROTECTED] wrote:
Either that or pay for the legal licensing of G729 and get support through
the appropriate channels. Using the code for anything other than learning
purposes is illegal, not to mention that licensing is quite inexpensive.
Using
The language support is supposed to be there I know I've played with
it and there are at least SOME grammatical changes (don't recall which
right now)
But if further language support is needed you should file a bugreport.
On Jan 14, 2008 5:04 PM, Anton Krall [EMAIL PROTECTED] wrote:
Guys,
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Brian Hutchinson wrote:
| Hi,
|
| Using Asterisk 1.4.17. I'm calling a PHP script through AGI. No matter
| what my script returns (0 or -1), AGISTATUS always appears to be 0 =
| SUCCESS.
|
| I was wanting my script to be able to return a value to
The Digium svn servers are down, and will likely be down for the rest of the
evening, as I perform some system maintenance. I apologize for any
inconvenience that this may cause.
--
Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc.
On Jan 14, 2008 6:23 PM, Abdul [EMAIL PROTECTED] wrote:
Hi All,
We updated with Asterisk 1.4.17 but it seems unstable. 3, 4 times in one
day it stop to response to the SIP Clinets so they cannot make call or
register. But safe_asterisk not restarting it back because asterisk running
without
The Asterisk.org development team has released Zaptel versions 1.2.23 and 1.4.8.
These releases contain a number of bug fixes as well as new features, including:
* New and greatly improved fxotune utility
-
http://lists.digium.com/pipermail/asterisk-users/2008-January/203778.html
*
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
bilal ghayyad wrote:
| Hi List;
|
| With new technolgy, alot of mobiles now support Video
| Call, so what is the possibility to have Asterisk
| supporting Video so it support Video call at theie
| Phones?
Have a look at sip.fontventa.com as well as
Hi David,
Thank you for suggestion. It seems to work well. So asterisk does inband
dtmf to SIP INFO dtmf conversion well. I am curious to know why there is no
consistency with 2833 to INFO DTMF conversion. Is it a known issue with
asterisk?
Regards,
Mayur
_
From: dave cantera
Im looking at app_voicemail (remember, this is on 1.2.x) and there seems
to be some syntax changes for Spanish but doesn't seem to have all
that's required... Ill file a bug report on mantis.
AK
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
On Jan 14, 2008 6:54 PM, Andrew Joakimsen [EMAIL PROTECTED] wrote:
On Jan 14, 2008 5:51 PM, Steve Totaro [EMAIL PROTECTED]
wrote:
Either that or pay for the legal licensing of G729 and get support
through
the appropriate channels. Using the code for anything other than
learning
On Mon, Jan 14, 2008 at 07:50:22PM -0500, Steve Totaro wrote:
On Jan 14, 2008 6:54 PM, Andrew Joakimsen [EMAIL PROTECTED] wrote:
I wonder how many Chinese VoIP phones with G729 G723 codecs have
actually licensed the codec?
Probably none.
Well, they sell in the US and in other countries.
On Tue, 15 Jan 2008, Matt Riddell wrote:
Brian Hutchinson wrote:
|
| Using Asterisk 1.4.17. I'm calling a PHP script through AGI. No matter
| what my script returns (0 or -1), AGISTATUS always appears to be 0 =
| SUCCESS.
Why don't you just set a variable from the AGI and then test for
I'm running 1.4.17. I've been running that version plus an addition of
Unicall MFC/R2 and the only time I have seen it die is right away on startup
due to something in one of the .conf files not being right. It has not died
during normal operation. I'm running two TE420B cards on a large Dell
I'm trying to get call parking to work, but I've run out of things to try.
I can place a call between two internal extensions, then on one
extension transfer the call to extension 700, and the call gets parked
on 701 but I don't hear the extension number when I do the transfer. I
can hangup and
On Jan 14, 2008 7:50 PM, Steve Totaro [EMAIL PROTECTED] wrote:
I would argue that it is illegal. The main definition of illegal is 1.
against law: contravening a specific law, especially a criminal law.
http://encarta.msn.com/dictionary_/illegal.html
Illegal means that something violates
Why don't you just set a variable from the AGI and then test for it in
That is what I ended up doing and that worked. Just thought I'd post to the
list since from what I read it sounds like the script return value should be
reflected in AGISTATUS and it wasn't. Didn't know if it was a bug
No features are being added for 1.2 so I'd check to see if 1.4 has the
changes you need before filing a bugreport.
On Jan 14, 2008 7:47 PM, Anton Krall [EMAIL PROTECTED] wrote:
Im looking at app_voicemail (remember, this is on 1.2.x) and there seems
to be some syntax changes for Spanish but
1.4.17.
Rob wrote:
I'm trying to get call parking to work, but I've run out of things to try.
I can place a call between two internal extensions, then on one
extension transfer the call to extension 700, and the call gets parked
on 701 but I don't hear the extension number when I do the
Hello,
following the description in the wiki
(http://www.voip-info.org/wiki/view/Asterisk+phone+snom)
I have set up a number of SNOM phones to monitor extensions with hints.
The lights on the phones flash when a call on another phone comes in.
According to the article in the wiki I need the
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