I'm curious if anyone knows of any possibility to use video VOIP client
(like Ekiga or Linphone or...) under Linux that could be operated by
touchscreen friendly GUI (bigger buttons, large keypad, etc...) ?
I like Ekiga, but GUI is small and cannot be operated via touchscreen... But
maybe
On Wed, Jan 14, 2009 at 7:47 AM, Jai Rangi jpra...@gmail.com wrote:
Alex,
I must say wow, great explanation. It was a wonderful reading.
Thanks to everyone who made this interesting reading!
You're all invited to argue about this tomorrow, Friday the 15Th of
January at 12 Noon EST on the VoIP
On Wed, 14 Jan 2009 16:09:02 +0100, philipp-chemn...@gmx.de wrote:
setting the caller ID works perfect. Detecting if a caller is or isn't
registered is the problem. I'm using sip.
wouldnt ChanIsAvail() or regexten/regcontext settings in sip.conf assist
in this ?
--
Regards,
Joshua Colp schrieb:
- Klaus Darilion klaus.mailingli...@pernau.at wrote:
Philipp Kempgen schrieb:
Klaus Darilion schrieb:
Is it somehow possible to evaluate the SIP response code inside
the dialplan?
No. Part of the reasoning is that Asterisk is meant to be a
multi- protocol PBX,
IIRC FaxGateway is intelligent and works in both directions.
What are the problems?
klaus
Alex Balashov schrieb:
Well, T.38 works over IP, not TDM...
James Lamanna wrote:
Hi,
I've been trying to use the FaxGateway application to send T.38 out
over Zaptel using asterisk but I don't seem
14 jan 2009 kl. 14.02 skrev Klaus Darilion:
Hi!
Is it somehow possible to evaluate the SIP response code inside the
dialplan?
I have an Asterisk server which forwards requests to various PSTN
gateways with SIP. If the Dial() attempt is not successful I want to
differ at least these 3
14 jan 2009 kl. 18.57 skrev Philipp Kempgen:
Klaus Darilion schrieb:
Philipp Kempgen schrieb:
Klaus Darilion schrieb:
Is it somehow possible to evaluate the SIP response code inside the
dialplan?
No.
Part of the reasoning is that Asterisk is meant to be a multi-
protocol PBX, not a SIP
Just google for your subject.
short: insmod just tries to load one module. modprobe checks
dependencies and loads needed kernel modules too.
klaus
Olivier schrieb:
hello,
Here (http://updates.xorcom.com/astribank/bristuff/1.4/INSTALL.html) you
can read :
cd qozap
modprobe zaptel
On Thursday, January 15, 2009, D Tucny wrote:
It's so much nicer to use packages, in the case of CentOS, RPMs...
that way everything installed is owned by the package and removal of
the package removes most of what was installed...
Thanks for the reply.
I must be missing something, since all
Tim Panton ha scritto:
[ ... snip .. ]
I'm interested to use it as IAX2 API within my UI, so something
like:
- open IAX2 channel
- call 123456
- answer a call
- close IAX2 channel
It is definitely capable of that with an added class or 2.
Could you point me in the
On Thu, Jan 15, 2009 at 02:13:58AM +0100, Olivier wrote:
hello,
Here (http://updates.xorcom.com/astribank/bristuff/1.4/INSTALL.html) you can
read :
cd qozap
modprobe zaptel
insmod qozap.o (for kernel 2.4)
insmod qozap.ko (for kernel 2.6)
ztcfg
I should also point out that those are
Hi all,
here you can find the demo site: http://www.yosd.at/corraleta/
I have also opend a forum for further discussion of the corraleta sdk...
http://www.yosd.at/index.php?option=com_joomlaboardItemid=39func=showcatcatid=7
regards,
Wolfgang
Wolfgang Pichler schrieb:
Hi all,
thanks Tim and
On 15 Jan 2009, at 07:30, Wolfgang Pichler wrote:
Hi all,
thanks Tim and Mexuar for releasing this here...
I have already taken the source - and compiled a little java applet
which is self signed to test the whole thing.
That was quick :-)
I will put it on my site (and allow users to
Hi,
there is no gsm codec - thats correct - i must have seen something
else... (is there a gsm - or other - codec implementation available for
free use ?)
I will test it further - and if it fits my needs - then i will put some
work into it...
I will put it on sourceforge if you want - but i
I gues understood his email wrong. Seemed to be that he wante to make 2
calls via the web and bridge them.
- Original Message -
From: C. Savinovich c.savinov...@itntelecom.com
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Thursday,
On Thu, Jan 15, 2009 at 4:09 AM, Rilawich Ango maillist...@gmail.com wrote:
Hi,
I wonder how I can relate the CDR records for the case of call
transfer. I can't find their relationship in CDR. Any can advice?
ango
You may want to read this thread.
On Thu, Jan 15, 2009 at 12:18 AM, David @ULC ucoms2...@gmail.com wrote:
I am getting this Error on my Asterisk.
How to solve it ?
ERROR[2654]: chan_sip.c:11355 handle_request: Missing Cseq. Dropping this
SIP message, it's incomplete.
If the error message being reported by Asterisk is correct
Hi All,
I'd appreciate some help on how to implement call stealing. That is,
where you dial a code to redirect any call on the system to your
handset.
I'm getting rid of my BRI service and I'm trying to replace the
functionality of my existing ISDN2e PBX (Cybergear Gold) with VOIP and
Asterisk.
On 15 Jan 2009, at 10:06, Wolfgang Pichler wrote:
Hi,
there is no gsm codec - thats correct - i must have seen something
else... (is there a gsm - or other - codec implementation available
for
free use ?)
I think there is an LGPL gsm implementation in java.
I will test it further -
Johansson Olle E schrieb:
14 jan 2009 kl. 18.57 skrev Philipp Kempgen:
Klaus Darilion schrieb:
Philipp Kempgen schrieb:
Klaus Darilion schrieb:
Is it somehow possible to evaluate the SIP response code inside the
dialplan?
No.
Part of the reasoning is that Asterisk is meant to be a
That is very nice, but where are the HANGUPCAUSE values documented?
Thanks.
on Thursday 01/15/2009 Johansson Olle E(o...@edvina.net) wrote
14 jan 2009 kl. 14.02 skrev Klaus Darilion:
Hi!
Is it somehow possible to evaluate the SIP response code inside the
dialplan?
I
and if you use the trasnfer app whit the features chann?
David
2009/1/15 Geoff Lane ge...@gjctech.co.uk
Hi All,
I'd appreciate some help on how to implement call stealing. That is,
where you dial a code to redirect any call on the system to your
handset.
I'm getting rid of my BRI service
15 jan 2009 kl. 12.42 skrev Klaus Darilion:
Johansson Olle E schrieb:
14 jan 2009 kl. 18.57 skrev Philipp Kempgen:
Klaus Darilion schrieb:
Philipp Kempgen schrieb:
Klaus Darilion schrieb:
Is it somehow possible to evaluate the SIP response code inside
the
dialplan?
No.
Part of the
15 jan 2009 kl. 13.02 skrev John covici:
That is very nice, but where are the HANGUPCAUSE values documented?
That's the issue...
include/asterisk/causes.h is a good reference for now.
/O
Thanks.
on Thursday 01/15/2009 Johansson Olle E(o...@edvina.net) wrote
14 jan 2009 kl. 14.02 skrev
So i made a backup long time ago of the g729 license file for one of my
servers, problem is I dont remember which one. Anybody know how I can
identify which server this license file belongs to?
___
-- Bandwidth and Colocation Provided by
Dmitry Andrianov wrote:
Did I miss something? Is Asterisk capable of handling 16KHz audio already?
Can it mix 16KHz streams in the meetme rooms? Can it downsample them to 8kHz
for Zap channels?
Asterisk 1.6 can handle 16KHz streams and resample between 8KHz and
16KHz. The current
hi i am reading about new codecs and new stuff to be added to asterisk. (and
i say thanks to all the guys who are working to add all the new features).
will be R2 added to the main core of asterisk like ISDN?
Thanks
David
--
(\__/)
(='.'=)This is Bunny. Copy and paste bunny into your
Hi,
Here's part of the log that I see.
In this case I'm testing on a box that unfortunately doesn't have a
PRI connection.
I've so far tested with just voice calls so far, but as you can see,
FaxGateway can't even dial out to the SIP trunk properly.
Here's also what the dialplan looks like:
Jon Weisman wrote:
So i made a backup long time ago of the g729 license file for one of my
servers, problem is I dont remember which one. Anybody know how I can
identify which server this license file belongs to?
Use the 'asthostid' tool to get the Host-ID for the candidate servers,
and
Hi to all
i'm using PlayDTMF with AJAM, after the authentication, i make a
request like this:
host:8088/asterisk/mxml?action=PlayDTMFChannel=SIP/200-sdadsadkioahDigit=1
the result is:
ajax-response
response type='object' id='unknown'generic response='Success'
message='DTMF successfully queued'
Why not use call-conferencing? If you transferred your call into a
conference room, you could join the conference from any extension on your *.
When the caller hangs up, just end the conference.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Ayman, after you BUY the license/firmware, etc, to cisco, I use 7911G with
Astterisk, my xml conf file is in the wiki
: )
2009/1/13 Steve Edwards asterisk@sedwards.com
On Tue, 13 Jan 2009, Ayman Boules (Live.COM) wrote:
It will be great if someone can help me upgrade a Cisco 7971G-GE
When I use below line sin extension.conf file
[from-ipkall]
exten = 901835,1,NoOp(from-ipkall)
exten = 901835,2,NoOp(${CALLERIDNAME}/${CALLERIDNUM})
exten = 901835,3,Dial(Local/200 at internal)
I get below CLI :
*Quote:*
login as: root
r...@192.168.0.2's password:
Last login: Wed Jan 14
Is in the process of being merged.
http://bugs.digium.com/view.php?id=12509
http://reviewboard.digium.com/r/40/
http://www.libopenr2.org/
Moisés Silva
On Thu, Jan 15, 2009 at 9:44 AM, David fire ddf...@gmail.com wrote:
hi i am reading about new codecs and new stuff to be added to asterisk.
awesome thanks!
- Original Message -
From: Kevin P. Fleming kpflem...@digium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, January 15, 2009 9:48 AM
Subject: Re: [asterisk-users] G729 host id
Jon Weisman wrote:
So i
thanks for the answer.
any idea in wich version it will be merged?
thanks
2009/1/15 Moises Silva moises.si...@gmail.com
Is in the process of being merged.
http://bugs.digium.com/view.php?id=12509
http://reviewboard.digium.com/r/40/
http://www.libopenr2.org/
Moisés Silva
On Thu, Jan 15,
Hi,
Our potentiel next phone provider ask me a question i can't answer for sure,
maybe someone here knows ?
He says that is equipement only support VN4 protocol or more, or ETSI,
however i can't find matching terms in the digium documentation or
the chan_dahdi/dahdi/system.conf files...
Any
On Thursday, January 15, 2009, David fire wrote:
and if you use the trasnfer app whit the features chann?
Thanks for the suggestion. I'll see if I can find it in the docs.
--
Geoff
___
-- Bandwidth and Colocation Provided by
On Thursday, January 15, 2009, Danny Nicholas wrote:
Why not use call-conferencing? If you transferred your call into a
conference room, you could join the conference from any extension on
your *. When the caller hangs up, just end the conference.
Thanks for the reply.
AIUI, you need to set
Geoff Lane wrote:
On Thursday, January 15, 2009, Danny Nicholas wrote:
Why not use call-conferencing? If you transferred your call into a
conference room, you could join the conference from any extension on
your *. When the caller hangs up, just end the conference.
Thanks for the
Hello
Does anyone have any experience with configuring BT (British Telecom)
ISDN2e lines to work with Patton SmartNodes?
I have a Patton SmartNode 4638, which is now connected to 3 x ISDN2e
lines - and in turn connected to our internal LAN. I'm having huge
issues configuring the SmartNode to
On Thursday, January 15, 2009, Drew Gibson wrote:
Would SLA (Shared Line Appearance) work for this?
Put call on hold, press button beside flashing light on second handset?
Thanks for the reply.
I don't think it would work with my hardware. I've got two Nortel 355
analog handsets, one plugged
Quoth Geoff Lane ge...@gjctech.co.uk...
AIUI, you need to set up the conference before leaving the extension
on which you took the call.
Yes you do. You'd need to explicitly send the call to a conference,
listen and remember the conference number.
FWIW, Call Stealing is a feature I miss from
Here's a working scenario from my asterisk -
I have a static conference 6350 set up with no password. When a call comes
in, I transfer it to 6350. I can then access this call from any extension
by dialing 6350.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Hi all,
Anyone has this card installed and configured without echos between SIP
and VPB channels ?
I have 2 Openswitch cards and i always have echo problems in Analog
Lines.
If i operated SIP through SIP i have no echos, but if i try to operate
SIP through VPB there is alot of
My provider migrated from an old EOL softswitch to Trixbox.
I have a number (8159093011) on a different server on a different network. It
appears as though the incoming calls are trying to authenticate against that
number, which isn't present on the box. Could someone help me decode this
On Thursday, January 15, 2009, Jeff LaCoursiere wrote:
Cordless phones?
Sorry, couldn't resist :)
I've got some but the range isn't good enough to cover my entire
house. Besides which it's bad enough playing find the phone when a
cordless handset gets eaten by the settee or wanders off to the
Hi Olle,
Johansson Olle E schrieb:
14 jan 2009 kl. 18.57 skrev Philipp Kempgen:
Klaus Darilion schrieb:
Philipp Kempgen schrieb:
Klaus Darilion schrieb:
Is it somehow possible to evaluate the SIP response code inside the
dialplan?
No.
Part of the reasoning is that Asterisk is meant to be
On Thu, 15 Jan 2009, Geoff Lane wrote:
On Thursday, January 15, 2009, Drew Gibson wrote:
[snip]
However, SLA is functionally almost the same as call parking. In that
system, I transfer the call to extension 700 and the parking system
tells me the number (usually 701) I need to dial to
On Thu, 15 Jan 2009, Geoff Lane wrote:
On Thursday, January 15, 2009, Jeff LaCoursiere wrote:
Cordless phones?
Sorry, couldn't resist :)
I've got some but the range isn't good enough to cover my entire
house. Besides which it's bad enough playing find the phone when a
cordless handset
I get this warning in the Asterisk CLI once in a while, and it usually
corresponds with a phone not ringing when it should.
Warning in CLI: Inringing for peer [PEER] 0
What does it mean and what is the likely cause of this?
___
--
Stefan Schmidt s...@sil.at writes:
maybe a better solution is to set the callerid to anonymous or something
else and use the cdr userfield to set the callerid. so you still have
the information and the client doesnt see the callerid in any way.
Adaptive CDR (and custom CDR, if you prefer
On Thursday, January 15, 2009, Jeff LaCoursiere wrote:
I'm a bit confused as to how your old system exactly worked. When
you initially answer the phone (on presumably the wrong
extension), what did you do with that handset before getting up and
going to the right extension to steal it?
I'm confused as to why you think leaving a phone off the hook is better than
parking the call and hanging up the phone. The phone that's off the hook can't
receive any more calls after you've 'pulled' the one it was on the line with,
assuming you don't walk back to that phone and subsequently
What about Chanspy()?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Geoff Lane
Sent: Thursday, January 15, 2009 2:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
I'm trying to capture faxes on 1.6.1-beta4. AFAICT, app_fax is working
OK. I'm then using fax2mail to send the fax. That wasn't working, so i
posted for help using the System() cmd, since fax2mail did work from the
command line. But now I realize it's fax2mail and mime-construct itself.
I set
Have you tried your system stuff under su - asterisk? Once it works that
way, the system() command will work.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy
Sent: Thursday, January 15, 2009 2:45 PM
On Thursday, January 15, 2009, David Gibbons wrote:
I'm confused as to why you think leaving a phone off the hook is
better than parking the call and hanging up the phone.
Simply that you don't have to remember to park the call. With call
parking, if you forget to park the call before moving
Have you tried your system stuff under su - asterisk? Once it works that
way, the system() command will work.
asterisk is running as root, I run the command at the terminal as root.
I am guessing he doesn't even have an asterisk user.
___
--
On Thursday, January 15, 2009, Danny Nicholas wrote:
What about Chanspy()?
Thanks for the reply, but I suspect it won't do what I want.
AIUI, ChanSpy() doesn't transfer the call - it just lets another
extension listen in (and join in the conversation in whisper mode). So
(AFAICT) the call will
Joseph L. Casale wrote:
Have you tried your system stuff under su - asterisk? Once it works that
way, the system() command will work.
asterisk is running as root, I run the command at the terminal as root.
I am guessing he doesn't even have an asterisk user.
Well I do have an asterisk
If you are running the script within Asterisk as root, then it's a path
environment issue. My guess(and I run into this with cron jobs all the
time) is that the path is different from the command line than the
environment that the script runs under.
There are times where the fix is to use the
hey it is preatty easy
now i understand the problem
is simple
hangup in new location
dial steal code for asterisk is just an extension and it should start an
AGI
the system search for the call in the same group
bridge the channel to the current channel asterisk 1.6
or
the system
Look int the ChannelRedirect command.
Geoff Lane wrote:
Hi All,
I'd appreciate some help on how to implement call stealing. That is,
where you dial a code to redirect any call on the system to your
handset.
I'm getting rid of my BRI service and I'm trying to replace the
functionality of
That's Digium's folks decision. It was said they wanted it for 1.6.3,
but, that's not for sure, as I said, they will decide.
On Thu, Jan 15, 2009 at 11:54 AM, David fire ddf...@gmail.com wrote:
thanks for the answer.
any idea in wich version it will be merged?
thanks
2009/1/15 Moises Silva
this link:
http://www.telos-systems.com/techtalk/digiphones/digiphones_4.htm
States the following:
Generic PBXs will not do for our broadcast application – they just
don’t have the features necessary. For example, while lines may
certainly be shared to multiple phones, there is no way to switch
Can anyone tell me how I can completely move an established call off of one
Asterisk server to another?
In our case we have a server with our IVR. Depending upon digits entered,
the call can be transferred to any of our other servers depending where the
extension or queue reside.
We would like
Are you planning on connecting your two Asterisk servers with SIP or IAX?
Check out this tutorial if using SIP:
http://hostseries.com/connecting-to-asterisk-servers-via-sip/
You should be able to adapt it to your needs. Good luck!
Paul wrote:
Can anyone tell me how I can completely move an
On Thursday 15 January 2009 13:02:32 Geoff Lane wrote:
On Thursday, January 15, 2009, Jeff LaCoursiere wrote:
Cordless phones?
Sorry, couldn't resist :)
I've got some but the range isn't good enough to cover my entire
house. Besides which it's bad enough playing find the phone when a
On Thu, 15 Jan 2009, Tilghman Lesher wrote:
On Thursday 15 January 2009 13:02:32 Geoff Lane wrote:
On Thursday, January 15, 2009, Jeff LaCoursiere wrote:
Cordless phones?
Sorry, couldn't resist :)
I've got some but the range isn't good enough to cover my entire
house. Besides which it's
a strange problem of multiple sip registrations and peer selection in
sip.conf is calling for your suggestions!!
let's examine this scenario:
some numbers and passwords hidden with HHHs to protect the guilty :)
I have 3 distinct sip subscriptions with cordiaip.net provider in US. For
each of
Lyle Giese wrote:
If you are running the script within Asterisk as root, then it's a path
environment issue. My guess(and I run into this with cron jobs all the
time) is that the path is different from the command line than the
environment that the script runs under.
There are times
On Thursday 15 January 2009 17:36:31 Jeff LaCoursiere wrote:
On Thu, 15 Jan 2009, Tilghman Lesher wrote:
On Thursday 15 January 2009 13:02:32 Geoff Lane wrote:
On Thursday, January 15, 2009, Jeff LaCoursiere wrote:
Cordless phones?
Sorry, couldn't resist :)
I've got some but the
If you want to email me your fixed script I'll put it up on the web site...
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy
Sent: January 15, 2009 7:08 PM
To: Asterisk Users List
Subject: Re:
Hello everyone!
I just installed the latest asterisk from svn. Now I'm retrying my luck with
gtalk and jingle. I have moved so my basic setup has changed a bit... I'm not
sure if it helps or hurts.
I tried this:
call myself:
channel originate gtalk/gtalk_account/julienco...@googlemail.com
Hello!
Sorry for not being able to phrase the problem in one line. My phone
situation is this:
The calls go over analog line (or NGN/vip) I don't really get to see it. I
have got a router with a lot of jacks. One or two of them are for ISDN phones
or other ISDN capable devices. Can I use
I don't understand why so many government sites fail to provide some
sort of feed to their daily bulletins. What I am venting about in
specific are the Canadian CRTC and FCC sites, every day I have to go
to the website and when I reach the content, usually it isn't even
HTML but a Word or PDF
Has anyone been able to configure portech's mv-378 gateway with asterisk?
I did the configuration as per the manual but it does not work.
My server sees the portech gateway, but when the gateway is trying to
register to my server it fails. It says peer is not suppose to register.
The gateway
What about this?
http://www.thefederalregister.com/rss/department/FEDERAL_COMMUNICATIONS_COMMISSION/
Shidan wrote:
I don't understand why so many government sites fail to provide some
sort of feed to their daily bulletins. What I am venting about in
specific are the Canadian CRTC and FCC
I was Using freepbx-2.1.1, Asterisk 1.2.29 and Asterisk Addons 1.2.9
i tried to upgrade to Asterisk 1.4.22.1 and Addons 1.4.7
all of the IAX trunks got not working at all.
I tried to downgrade by make clean; make; make install in Atserisk 1.2.29
directory.but make gives errors in the end.
Ah, But Asterisk if not your Generic PBX!
You could do a few things.
For each show, (I take it that this is talk radio) You can set up a queue()
for each air studio. Callers would then be greeted with a custom greeting
that would be unique for each air studio.
How you interface with your
thanks moises
and Digium's folks put it asap please not until 1.6.3
thanks
2009/1/15 Moises Silva moises.si...@gmail.com
That's Digium's folks decision. It was said they wanted it for 1.6.3,
but, that's not for sure, as I said, they will decide.
On Thu, Jan 15, 2009 at 11:54 AM, David fire
Thanks all. I think click to call can fulfill my purpose.
On Thu, Jan 15, 2009 at 6:10 PM, Dovid Bender asteriskus...@dovid.net wrote:
I gues understood his email wrong. Seemed to be that he wante to make 2
calls via the web and bridge them.
- Original Message -
From: C. Savinovich
OCG Technical Support wrote:
If you want to email me your fixed script I'll put it up on the web site...
Well I'd be pleased to have any script of mine put up on any web site,
but the only thing I did was to hard wire my location of mime-construct:
MimeC=/usr/local/bin/mime-construct
and
On Thu, Jan 15, 2009 at 06:11:59PM +0100, Benoit wrote:
Hi,
Our potentiel next phone provider ask me a question i can't answer for sure,
maybe someone here knows ?
He says that is equipement only support VN4 protocol or more, or ETSI,
however i can't find matching terms in the digium
I saw that already. It's not a listing of the FCCs headlines. It's
just a very lame, unusable, unordered list of a few snippets from
random FCC meetings. Check the data in my feed and compare it that and
I think the answer to what about this becomes obvious.
Cheers,
Shidan
On Fri, Jan 16, 2009
Le 16.01.2009 04:11, Benoit a écrit :
Hi,
Our potentiel next phone provider ask me a question i can't answer for sure,
maybe someone here knows ?
He says that is equipement only support VN4 protocol or more, or ETSI,
however i can't find matching terms in the digium documentation or
the
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