Anyone know of a way to get Asterisk to use BEZEQ's BPhone?
I do not have an android or iOS device.
I need to keep a real BEZEQ landline, but having it on my asterisk
system would be really helpful.
I know about FXO cards, but the days of $10 ones are long gone. The only
ones I could find
On Mon, Sep 3, 2012 at 7:13 AM, Geoffrey S. Mendelson
geoffreymendel...@gmail.com wrote:
Anyone know of a way to get Asterisk to use BEZEQ's BPhone?
I do not have an android or iOS device.
I need to keep a real BEZEQ landline, but having it on my asterisk system
would be really helpful
On Mon, Sep 3, 2012 at 7:47 AM, shimi linux...@shimi.net wrote:
On Mon, Sep 3, 2012 at 7:13 AM, Geoffrey S. Mendelson
geoffreymendel...@gmail.com wrote:
Anyone know of a way to get Asterisk to use BEZEQ's BPhone?
I do not have an android or iOS device.
I need to keep a real BEZEQ
On Jan 11, 2011, at 3:49 PM, Erez D wrote:
Did you get any answers ?
Sorry, I never persued it. I'm hoping someone who actually speaks
Hebrew will. :-)
Geoff
--
Geoffrey S. Mendelson, N3OWJ/4X1GM
Those who cannot remember the past are condemned to misquote it.
On Sun, Oct 3, 2010 at 7:35 PM, geoffrey mendelson
geoffreymendel...@gmail.com wrote:
On Oct 3, 2010, at 7:10 PM, Amichai Rotman wrote:
The service is called Bphone. It seems to be a VOB line that you can
access by installing an app called Phone Dialer on your cellular phone. I
couldn't
: Sunday, October 03, 2010 19:45
To: Baruch Shpirer
Cc: 'Amichai Rotman'; 'linux-il Linux'
Subject: Re: asterisk and bezeq
On Oct 3, 2010, at 7:28 PM, Baruch Shpirer wrote:
They have a voip service for home (076) in which they give you an
MP202 adapter with 2 FXS ports And also a hardcoded
not know if it will work with Asterisk, but I'll be glad to hear if
it does.
Thanks. This was an add in Yediot in late August, which showed all ot the
options you could get from BEZEQ and it showed some sort of linkage between
your phone and a laptop. I guess they were talking about Wifi
it isnt more expensive then bezeq
Ill get it checked with asterisk soon enough
Baruch
_
From: linux-il-boun...@cs.huji.ac.il [mailto:linux-il-boun...@cs.huji.ac.il]
On Behalf Of Amichai Rotman
Sent: Sunday, October 03, 2010 19:11
To: geoffrey mendelson
Cc: linux-il Linux
Subject: Re
On Oct 3, 2010, at 7:10 PM, Amichai Rotman wrote:
The service is called Bphone. It seems to be a VOB line that you can
access by installing an app called Phone Dialer on your cellular
phone. I couldn't find what's the software for the PC...
You could call 199 and ask for an English
. That is unfortunately nothing that I want. :-( I already have
a similar adapter with a US line. It works perfectly fine without port
5060 redirect to it. I also do have an asterisk system, but it uses a
different SIP port. IMHO having port 5060 open to asterisk is a way of
finding the security holes
.
Is this true? I'm not talking about any random Israeli DID I get from
a third party, I'm talking about my BEZEQ number, which I still want
attached to my landline.
Has anyone done this?
Have you done it with ASTERISK?
Thanks in advance for any info.
Geoff.
--
Geoffrey S. Mendelson, N3OWJ
Bezeq started to offer SIP trunks (calling it ipri, at least in the PRI
equivalent).
I can only guess that they offer also similar but to FXO.
I do not know if it will work with Asterisk, but I'll be glad to hear if it
does.
Ido
On Sat, Oct 2, 2010 at 20:54, geoffrey mendelson
geoffreymendel
On Oct 2, 2010, at 9:21 PM, ik wrote:
Bezeq started to offer SIP trunks (calling it ipri, at least in the
PRI equivalent).
I can only guess that they offer also similar but to FXO.
I do not know if it will work with Asterisk, but I'll be glad to
hear if it does.
Thanks. This was an add
Hi all
I'm using Ekiga as a client in my ubuntu.
The problem is i can't find a way to dial an extension when the call
answered by an automate on the other side (like click 9 etc...)?
Any other client that does it?
Tnx!
David
--
בברכה,
דוד רונקין
___
2010/9/13 David Ronkin dron...@gmail.com
Hi all
I'm using Ekiga as a client in my ubuntu.
The problem is i can't find a way to dial an extension when the call
answered by an automate on the other side (like click 9 etc...)?
Any other client that does it?
You should try (if the client
it?
Is this a question about Asterisk? If so: you probably need an IVR,
like the 'demo' context in the example extensions.conf .
--
Tzafrir Cohen | tzaf...@jabber.org | VIM is
http://tzafrir.org.il || a Mutt's
tzaf...@cohens.org.il || best
tzaf...@debian.org
On Sep 13, 2010, at 12:54 PM, David Ronkin wrote:
Hi all
I'm using Ekiga as a client in my ubuntu.
The problem is i can't find a way to dial an extension when the call
answered by an automate on the other side (like click 9 etc...)?
Any other client that does it?
Try Zoiper. The dialpad
On Mon, Sep 13, 2010 at 01:12:02PM +0200, geoffrey mendelson wrote:
On Sep 13, 2010, at 12:54 PM, David Ronkin wrote:
Hi all
I'm using Ekiga as a client in my ubuntu.
The problem is i can't find a way to dial an extension when the call
answered by an automate on the other side (like
i use ekiga 3.2.6 and i don't have such a tab.
what version should i download?
thanks
2010/9/13 Tzafrir Cohen tzaf...@cohens.org.il
On Mon, Sep 13, 2010 at 01:12:02PM +0200, geoffrey mendelson wrote:
On Sep 13, 2010, at 12:54 PM, David Ronkin wrote:
Hi all
I'm using Ekiga as a
On Mon, Sep 13, 2010 at 01:27:56PM +0200, David Ronkin wrote:
i use ekiga 3.2.6 and i don't have such a tab.
In the main window?
what version should i download?
Works fine here (3.2.7). IIRC it has always been in that tab.
--
Tzafrir Cohen | tzaf...@jabber.org | VIM is
On Mon, Sep 13, 2010 at 12:05:07PM +, Tzafrir Cohen wrote:
On Mon, Sep 13, 2010 at 01:27:56PM +0200, David Ronkin wrote:
i use ekiga 3.2.6 and i don't have such a tab.
In the main window?
what version should i download?
Works fine here (3.2.7). IIRC it has always been in that tab.
On Sat, Jan 30, 2010 at 21:04, geoffrey mendelson
geoffreymendel...@gmail.com wrote:
I've been playing more with asterisk and fax and wanted to post this to
document what I found and see if anyone else had a better answer.
I'm running Asterisk 1.6.2 under UBUNTU 9.10 using the UBUNTU
On Sat, Jan 30, 2010 at 09:04:52PM +0200, geoffrey mendelson wrote:
I've been playing more with asterisk and fax and wanted to post this to
document what I found and see if anyone else had a better answer.
I'm running Asterisk 1.6.2 under UBUNTU 9.10 using the UBUNTU packages. I
found
On Sun, Jan 17, 2010 at 07:01:57AM +0200, geoffrey mendelson wrote:
On Jan 17, 2010, at 2:29 AM, Tzafrir Cohen wrote:
On Sat, Jan 16, 2010 at 08:05:51PM +0200, ik wrote:
You can use libhdate with some AGI and set variable to calculate
shabat per
place and day.
This is relatively quite
/documentation/module-documentation/day-night-mode-control
Generally FreePBX is not a good source for dialplan snippets: its
dialplan is very complex and in many cases way overly generic to be
useful / comprehensable.
However a search for
asterisk day night mode
gives a number of useful hits
On Sun, Jan 17, 2010 at 12:44, Tzafrir Cohen tzaf...@cohens.org.il wrote:
On Sun, Jan 17, 2010 at 07:01:57AM +0200, geoffrey mendelson wrote:
On Jan 17, 2010, at 2:29 AM, Tzafrir Cohen wrote:
On Sat, Jan 16, 2010 at 08:05:51PM +0200, ik wrote:
You can use libhdate with some AGI and set
.
You can't automate Berkly insert/update without any AGI/AMI running at least
once, unless you know something I do not know.
Sure you can:
asterisk -rx 'database put flags shabat 1'
asterisk -rx 'database del flags shabat'
Run from cron at a time of your choosing.
Likewise for global
($[${DB_EXISTS(flags/shabat)}]:context-for-shabat)
But that's have to be updated every week, or every X amount of time.
You can't automate Berkly insert/update without any AGI/AMI running at
least
once, unless you know something I do not know.
Sure you can:
asterisk -rx 'database put flags
On Jan 17, 2010, at 3:32 PM, ik wrote:
But for that you need to know when is the shabat enter a specific
location, so you need extra program for it (even if it's pure bash),
to calculate the exact time it started. I think that the berkley
should have the exact time and date for each
On 17/01/2010, at 15:32, ik wrote:
But for that you need to know when is the shabat enter a specific location,
so you need extra program for it (even if it's pure bash), to calculate the
exact time it started. I think that the berkley should have the exact time
and date for each week for
On Jan 17, 2010, at 3:58 PM, sammy ominsky wrote:
On 17/01/2010, at 15:32, ik wrote:
But for that you need to know when is the shabat enter a specific
location,
so you need extra program for it (even if it's pure bash), to
calculate the
exact time it started. I think that the berkley
(for Asterisk =
1.6.0) you can use the function DEVICE_STATE from within the dialplan.
--
Tzafrir Cohen | tzaf...@jabber.org | VIM is
http://tzafrir.org.il || a Mutt's
tzaf...@cohens.org.il || best
ICQ# 16849754 || friend
On Jan 17, 2010, at 5:57 PM, Tzafrir Cohen wrote:
You can use the simple fact that the phone is not registered as your
switch. See regexten in sip.conf . Or alternatively (for Asterisk =
1.6.0) you can use the function DEVICE_STATE from within the dialplan.
Thanks, but that would force
On Sun, Jan 17, 2010 at 6:40 PM, geoffrey mendelson
geoffreymendel...@gmail.com wrote:
On Jan 17, 2010, at 5:57 PM, Tzafrir Cohen wrote:
You can use the simple fact that the phone is not registered as your
switch. See regexten in sip.conf . Or alternatively (for Asterisk =
1.6.0) you can
2010/1/18 geoffrey mendelson geoffreymendel...@gmail.com:
On Jan 17, 2010, at 3:58 PM, sammy ominsky wrote:
On 17/01/2010, at 15:32, ik wrote:
But for that you need to know when is the shabat enter a specific
location,
so you need extra program for it (even if it's pure bash), to calculate
Anyone know if there is a shabbat mode, or persistent variable?
I know that I can make a night mode by time, but since shabbat time
changes every week, I'm just looking for something where I can dial an
extension number which would turn on shabbat mode and another
extension to turn it off
You can use libhdate with some AGI and set variable to calculate shabat per
place and day.
Ido
http://ik.homelinux.org/
On Sat, Jan 16, 2010 at 19:34, geoffrey mendelson
geoffreymendel...@gmail.com wrote:
Anyone know if there is a shabbat mode, or persistent variable?
I know that I can
On Sat, Jan 16, 2010 at 08:05:51PM +0200, ik wrote:
You can use libhdate with some AGI and set variable to calculate shabat per
place and day.
This is relatively quite expensive. Maybe a qeekly cron to update a
relevant GotoIfTime line in the dialplan?
--
Tzafrir Cohen |
On Jan 17, 2010, at 2:29 AM, Tzafrir Cohen wrote:
On Sat, Jan 16, 2010 at 08:05:51PM +0200, ik wrote:
You can use libhdate with some AGI and set variable to calculate
shabat per
place and day.
This is relatively quite expensive. Maybe a qeekly cron to update a
relevant GotoIfTime line in
On Sun, Jan 17, 2010 at 7:01 AM, geoffrey mendelson
geoffreymendel...@gmail.com wrote:
Actually I'm quite happy with picking up the phone and entering an
extension number that puts the system into shabbat mode, and doing the
opposite motzi shabbat. What I am looking for is a way of setting
I have a few asterisk/callweaver questions. Callweaver docs seem to be
very rudimentary, so I've looked at asterisk docs, but still can't
find the answers I need.
Here are my questions, any help would be appreciated. I've tried to do
a web search and looked at the Oreily book with no luck
On Wed, Jan 06, 2010 at 01:44:36PM +0200, geoffrey mendelson wrote:
I have a few asterisk/callweaver questions. Callweaver docs seem to be
very rudimentary, so I've looked at asterisk docs, but still can't find
the answers I need.
Here are my questions, any help would be appreciated. I've
On Jan 6, 2010, at 2:09 PM, Tzafrir Cohen wrote:
An invalid extension still goes somewhere in the dialplan. Use that
fallback. In some cases this is as simple as providing a catch-all
extension - _X. and such. Patterns have lower matching priority than
actual extensions.
Ok, thansks, I'll
Hi,
For on-going needs and projects, we are looking for a person with
following knowledge and experience:
- Experience in development of PHP UI / MySQL Db following Project
specs documents.
- Ability to explore and resolve problems independently.
- Preferably experience with Asteriks management
Hi,
There's an outfit looking for a full time asterisk person
--
Thanks,
Uri
Si fractum non sit, noli id reficere.
___
Linux-il mailing list
Linux-il@cs.huji.ac.il
http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
Uri Bruck wrote:
Hi,
There's an outfit looking for a full time asterisk person
Contact email - da...@a.co.il
I have no further details - please don't reply to me
--
Thanks,
Uri
Si fractum non sit, noli id reficere.
___
Linux-il mailing
1. Free calls to U.S. Toll Free numbers (1-800, 888, 877, 866). Use the
following SIP registration:
sip/164164$outn...@sip.tollfreegateway.com (You can create an Asterisk Trunk
for all 1800/888/877/866 using this registartion string.
2. Free 5 minute calls to any US number and most Int'l
Hi all
We use IAX2 protocol in our Asterisk 1.4 PhpAgi application.
I wonder if anyone knows why my Outgoing CallerId dosn't work when i
do: Set(CALLERID(name)=...)
but the call goes with hidden number)?
Is there any value i should put in the iax.conf to enable this?
Thanks in advance.
David
On Thu, Jul 16, 2009 at 12:56:36PM +0300, David Ronkin wrote:
Hi all
We use IAX2 protocol in our Asterisk 1.4 PhpAgi application.
I wonder if anyone knows why my Outgoing CallerId dosn't work when i
do: Set(CALLERID(name)=...)
Are you trying to set the name? number? both?
Also note
Number
Thanks!
2009/7/16 Tzafrir Cohen tzaf...@cohens.org.il
On Thu, Jul 16, 2009 at 12:56:36PM +0300, David Ronkin wrote:
Hi all
We use IAX2 protocol in our Asterisk 1.4 PhpAgi application.
I wonder if anyone knows why my Outgoing CallerId dosn't work when i
do: Set(CALLERID(name
Un-top-posting,
On Thu, Jul 16, 2009 at 01:47:53PM +0300, David Ronkin wrote:
2009/7/16 Tzafrir Cohen tzaf...@cohens.org.il
On Thu, Jul 16, 2009 at 12:56:36PM +0300, David Ronkin wrote:
Hi all
We use IAX2 protocol in our Asterisk 1.4 PhpAgi application.
I wonder if anyone
16, 2009 at 12:56:36PM +0300, David Ronkin wrote:
Hi all
We use IAX2 protocol in our Asterisk 1.4 PhpAgi application.
I wonder if anyone knows why my Outgoing CallerId dosn't work when i
do: Set(CALLERID(name)=...)
Are you trying to set the name? number? both
settings else where in
Asterisk that forces the situation (like in iax.conf or something).
Cheers,
ido
http://ik.homelinux.org/
2009/7/16 David Ronkin dron...@gmail.com
Yep
I tried both - probably your assumption is correct - the provider blocks it
:(.
Tnx again!
David
2009/7/16 Tzafrir
that you do not have a caller id settings else where in
Asterisk that forces the situation (like in iax.conf or something).
Cheers,
ido
http://ik.homelinux.org/
Also be sure, when sending to the PSTN, that you send the CallerID the way
the provider expects it; Usually it's a fully-qualified
block of the number you are trying to force, then
please make sure that you do not have a caller id settings else where in
Asterisk that forces the situation (like in iax.conf or something).
Cheers,
ido
http://ik.homelinux.org/
Also be sure, when sending to the PSTN, that you send the CallerID
that you do not have a caller id settings else where in Asterisk that
forces the situation (like in iax.conf or something).
Cheers,
ido
http://ik.homelinux.org/
Also be sure, when sending to the PSTN, that you send the CallerID the way the
provider expects it; Usually it's a fully-qualified number
wrote:
Hi,
Im looking for a way to send sms to israeli phones through my asterisk.
Currently i have * installed on a wrt54gl (openwrt).
My FXO is an ht488 which is connected to a bezeq line.
My sip client is my nokia (via wifi).
Any simple ways to do that ?
(Im abroad and using my
On Thu, Jun 18, 2009 at 10:10 PM, Erez D erez0...@gmail.com wrote:
Hi,
Im looking for a way to send sms to israeli phones through my asterisk.
Currently i have * installed on a wrt54gl (openwrt).
My FXO is an ht488 which is connected to a bezeq line.
My sip client is my nokia (via wifi
On Sat, Jun 20, 2009 at 06:59:52PM +0200, Erez D wrote:
3. can i send sms via bezeq line, and does Grandstream HandyTone 488 support
it
This is what app_sms (used mostly indirectly, through smsq) is for.
http://svn.digium.com/svn/asterisk/trunk/doc/sms.txt
--
Tzafrir Cohen | tzaf
1.4 afaik
On 6/18/09, Tzafrir Cohen tzaf...@cohens.org.il wrote:
On Thu, Jun 18, 2009 at 09:10:08PM +0200, Erez D wrote:
Hi,
Im looking for a way to send sms to israeli phones through my asterisk.
Currently i have * installed on a wrt54gl (openwrt).
My FXO is an ht488 which is connected
Hi,
Im looking for a way to send sms to israeli phones through my asterisk.
Currently i have * installed on a wrt54gl (openwrt).
My FXO is an ht488 which is connected to a bezeq line.
My sip client is my nokia (via wifi).
Any simple ways to do that ?
(Im abroad and using my nokia via T9 to ssh
On Thu, Jun 18, 2009 at 09:10:08PM +0200, Erez D wrote:
Hi,
Im looking for a way to send sms to israeli phones through my asterisk.
Currently i have * installed on a wrt54gl (openwrt).
My FXO is an ht488 which is connected to a bezeq line.
My sip client is my nokia (via wifi).
Any simple
Hi,
If somebody is interested, we have few positions openings:
1. Support/development.
- Work is done from home (required self-discipline)
- background and understanding of Linux OS, PHP/MySQL.
- preferably understanding of Asterisk and its functionality.
- suitable for students as well.
- Self
host=dynamic ; This peer register with us
subscribemwi=yes ; Only send notifications if this phone
vmexten=asterisk ; dialplan extension to reach mailbox
allowsubscribe=yes
call-limit=99
disallow=all
allow=alaw ; dtmfmode=inband only
Hi,
Have a look at: http://hyppo.com/asterisk/
The patch on the web page is for asterisk 1.2.13 and 1.2.14, whereas
the currently packaged in debian is asterisk-1.4.21.2.
I tried to apply the patch to version 1.4.21.2 of asterisk, but the
compiled version was not able to send the SMSes
Hi,
Can someone kindly send me sections of working extensions.conf file
that enable SMS receival and sending please?
TIA,
--
Arie
=
To unsubscribe, send mail to [EMAIL PROTECTED] with
the word unsubscribe in the message body,
Hi,
Have a look at: http://hyppo.com/asterisk/
Ido
On Tue, Jul 29, 2008 at 10:42 PM, Arie Skliarouk [EMAIL PROTECTED] wrote:
Hi,
Can someone kindly send me sections of working extensions.conf file
that enable SMS receival and sending please?
TIA,
--
Arie
On Wed, May 28, 2008 at 12:56:21PM +0300, Arie Skliarouk wrote:
Hi,
On Tue, May 27, 2008 at 11:31 PM, Tzafrir Cohen [EMAIL PROTECTED]
wrote:
On Sun, May 25, 2008 at 07:35:47PM +0300, Arie Skliarouk wrote:
Hi,
I am trying to set up CallerID for asterisk on Digium 100P card, but so
Hi,
On Tue, May 27, 2008 at 11:31 PM, Tzafrir Cohen [EMAIL PROTECTED]
wrote:
On Sun, May 25, 2008 at 07:35:47PM +0300, Arie Skliarouk wrote:
Hi,
I am trying to set up CallerID for asterisk on Digium 100P card, but so
far,
every time there is incoming call, there is following warning
On Sun, May 25, 2008 at 07:35:47PM +0300, Arie Skliarouk wrote:
Hi,
I am trying to set up CallerID for asterisk on Digium 100P card, but so far,
every time there is incoming call, there is following warning in
/var/log/asterisk/messages:
[May 25 19:10:59] WARNING[1041] chan_zap.c: CallerID
Arie Skliarouk wrote:
Hi,
I am sorry, my card is Digium TDM400P, not the 100P. Does it make a
difference?
In any case, what settings of cidsignalling and cidstart people use?
Just comment them out. The defaults work just fine in Israel and you get
a numeric callerID.
Bezeq doesn't send a
Hi,
I am trying to set up CallerID for asterisk on Digium 100P card, but so far,
every time there is incoming call, there is following warning in
/var/log/asterisk/messages:
[May 25 19:10:59] WARNING[1041] chan_zap.c: CallerID returned with error on
channel 'Zap/1-1'
What is the proper
Last time that I've tried with a 100P card (3 years ago) it was not possible
to get caller id.
However, any other digium/xorcom card supports it
Ohad
On Mon, May 26, 2008 at 12:35 AM, Arie Skliarouk [EMAIL PROTECTED] wrote:
Hi,
I am trying to set up CallerID for asterisk on Digium 100P card
) it was not
possible to get caller id.
However, any other digium/xorcom card supports it
Ohad
On Mon, May 26, 2008 at 12:35 AM, Arie Skliarouk [EMAIL PROTECTED]
wrote:
Hi,
I am trying to set up CallerID for asterisk on Digium 100P card, but so
far, every time there is incoming call
or 60 ms.
See
http://svn.digium.com/svn/asterisk/branches/1.4/doc/rtp-packetization.txt
Note that there's a limit to what you can get from this: if the packets
are already full, you'll gain nothing and only increase the delay (and
even the jitter). You can not use this on iLBC: I'm not sure
Hi,
I wonder if someone here can help me. My customers and I have been
experiencing extremely poor quality on our VoIP lines to the US
lately. i am trying to track down the problem, and wonder if someone
could do me a favor...
I would like to install asterisk on a colocated linux box
wonder if someone here can help me. My customers and I have been
experiencing extremely poor quality on our VoIP lines to the US lately. i
am trying to track down the problem, and wonder if someone could do me a
favor...
I would like to install asterisk on a colocated linux box here
On 06/05/2008, at 21:49, ik wrote:
I'm not sure what you are asking.
I'm asking if someone has a linux box in a data center that I can
install asterisk on, or that already has asterisk installed, that I
can register to an asterisk server in the US with IAX. I would also
want to connect
customers and I have been
experiencing extremely poor quality on our VoIP lines to the US lately. i
am trying to track down the problem, and wonder if someone could do me a
favor...
I would like to install asterisk on a colocated linux box here in Israel
to
test an IAX trunk
lines to the US
lately. i
am trying to track down the problem, and wonder if someone could do
me a
favor...
I would like to install asterisk on a colocated linux box here in
Israel to
test an IAX trunk to the US. My company was supposed to colocate a
server
here in Israel
Tzafrir Cohen wrote:
Hmmm... what version of asterisk is it?
This reminds me of a bug that was fixed very early in 1.4 and also in
later 1.2 .
In any case, this description suggests that we should not rely too much
on whatever is written in the config files and check what is actually
A while back I asked about asterisk dialing out on a Zap group, and
adding a random extension to the top of the group. I then said that I
did a reboot and the problem was resolved.
Well, it was only partially resolved.
If asterisk is run immediately after a reboot, everything is ok.
If I
On Feb 3, 2008 6:19 PM, Shachar Shemesh [EMAIL PROTECTED] wrote:
A while back I asked about asterisk dialing out on a Zap group, and
adding a random extension to the top of the group. I then said that I
did a reboot and the problem was resolved.
Well, it was only partially resolved
On Sun, Feb 03, 2008 at 06:19:25PM +0200, Shachar Shemesh wrote:
A while back I asked about asterisk dialing out on a Zap group, and
adding a random extension to the top of the group. I then said that I
did a reboot and the problem was resolved.
Well, it was only partially resolved
Shachar Shemesh wrote:
Tzafrir Cohen wrote:
Please provide a trace (with verbose level = 3) of this, so we can see
what actually happens.
I'm not next to the system, so I cannot generate a log, but a link
pointing to how I set the verbosity level would be appreciated (or is it
just asterisk
Gilad Ben-Yossef wrote:
You don't need to be next to the system. Log in via SSH or similar and
run:
# asterisk -rv
This will connect you to the asterisk console remotely AND increase
the debug level.
Will it also pick up the physical extension and dial 9?
Then again
Shachar Shemesh wrote:
Gilad Ben-Yossef wrote:
You don't need to be next to the system. Log in via SSH or similar and
run:
# asterisk -rv
This will connect you to the asterisk console remotely AND increase
the debug level.
Will it also pick up the physical extension
Gilad Ben-Yossef wrote:
Then again, it seems like a lightening has striked my firewall (or
some such problem), because my office VPN seems totally down. At
least the PBX still answers the phone, so that was not hurt.
Write a small AGI script to reboot the firewall if offered the right
Hi, some nitpicking:
On Wed, Jan 30, 2008 at 12:00:27PM +0200, Gilad Ben-Yossef wrote:
Shachar Shemesh wrote:
Gilad Ben-Yossef wrote:
You don't need to be next to the system. Log in via SSH or similar and
run:
# asterisk -rv
This will connect you to the asterisk
Tzafrir Cohen wrote:
Please provide a trace (with verbose level = 3) of this, so we can see
what actually happens.
I'm not next to the system, so I cannot generate a log, but a link
pointing to how I set the verbosity level would be appreciated (or is it
just asterisk -vvv?)
group=3
Hi all,
A while back I asked three Asterisk questions. Two of those were
successfully answered by the list members, but one remains:
I have four internal extensions connected to a TDM400 card using four
FXS modules (channels 1-4). I also have two Bezeq lines connected to a
second TDM400
Damn :)
Well dunno, asterisk version btw... ?
On Monday 28 January 2008 18:36:21 Shachar Shemesh wrote:
Noam Rathaus wrote:
Hi,
I once had a similar issue, the TDM card was badly shipped, the modules
weren't the right one I thought they were.. i.e. FXS instead of FXO or
the other way
Noam Rathaus wrote:
Hi,
I once had a similar issue, the TDM card was badly shipped, the modules
weren't the right one I thought they were.. i.e. FXS instead of FXO or the
other way around, resulting in a card having FXS with another 3 FXO (or the
other way) which caused Asterisk to confuse
Hi,
I once had a similar issue, the TDM card was badly shipped, the modules
weren't the right one I thought they were.. i.e. FXS instead of FXO or the
other way around, resulting in a card having FXS with another 3 FXO (or the
other way) which caused Asterisk to confuse, and nothing to work
On Mon, Jan 28, 2008 at 05:19:38PM +0200, Shachar Shemesh wrote:
Hi all,
A while back I asked three Asterisk questions. Two of those were
successfully answered by the list members, but one remains:
I have four internal extensions connected to a TDM400 card using four
FXS modules
On 12/26/07, Shachar Shemesh [EMAIL PROTECTED] wrote:
1. Asterisk does not receive the caller ID from Bezeq.
I hope I did not miss an answer like this one in the long thread - if I did
- sorry for repeating.
Did you make sure that asterisk answers your call on the FXO only after the
second
On Wed, Dec 26, 2007 at 10:30:04PM +0200, Shachar Shemesh wrote:
Both /etc/zaptel.conf and /etc/asterisk/zapata.conf attached.
The provider is Bezeq.
Tzafrir Cohen wrote:
busydetect=yes
Yes. I uncommented busydetect, busycount and busypattern (500,500), and
the problem seems
Both /etc/zaptel.conf and /etc/asterisk/zapata.conf attached.
The provider is Bezeq.
Tzafrir Cohen wrote:
busydetect=yes
Yes. I uncommented busydetect, busycount and busypattern (500,500), and
the problem seems better now (though not perfect). I'll have to tweak
those around a bit, I
to resort to busydetect.
Might be a good idea to add that to the FAQs about installing Asterisk
in Israel. I never saw it mentioned.
You have a typo there:
$ grep asrecieved /usr/lib/asterisk/modules/chan_zap.so
$ strings /usr/lib/asterisk/modules/chan_zap.so | grep asre
asreceived
On Thu, Dec 27, 2007 at 05:43:33AM -0500, Tzafrir Cohen wrote:
$ grep asrecieved /usr/lib/asterisk/modules/chan_zap.so
$ strings /usr/lib/asterisk/modules/chan_zap.so | grep asre
asreceived
It's I before E
Except After C
http://alt-usage-english.org/excerpts/fxibefor.html
HTH,
Muli
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