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so it seems Asterisk Versions does not support video I guess
On Mon, Sep 1, 2014 at 9:30 AM, Khalid Touati khalidtou...@gmail.com
wrote:
Any article that goes through this (seems to be tedious) task to add video
support and patents?
On Mon, Sep 1, 2014 at 8:14 AM, Joshua Colp jc
if
community versions offer video calls at all.
Thank you!
--
Khalid Touati
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New to Asterisk? Join us for a live introductory webinar every Thurs
Any article that goes through this (seems to be tedious) task to add video
support and patents?
On Mon, Sep 1, 2014 at 8:14 AM, Joshua Colp jc...@digium.com wrote:
Khalid Touati wrote:
Hi Guys,
Kia ora,
Do you know of any asterisk community version that does video codec
trans-coding
!
--
Khalid Touati
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:45 AM, khalid touati khalidtou...@gmail.comwrote:
Hi Guys,
I wanted to try video on my asterisk 1.8, I was wondering if you guys
know a good softphone able to make video calls and compatible with PC (win
7) and Android? if yes do you guys know what codec is being used for video,
if any Codec
', on SIP/USPBX2-07d5
-- SIP/8425-07d4 Playing 'pbx-transfer.gsm' (language 'en')
and it gets disconnected. Anyone has a clue?
Thank you!
--
Khalid Touati
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That's fine guys I figured it out:
under features.conf:
[featuremap]
;blindxfer = #1; Blind transfer (default is #) -- Make
sure to set the T and/or t option in the Dial() or Queue() app call!
blindxfer = *
I changed it to * and got rid of the pb
--
Thank you Jeremy! I just posted the solution :)
On Wed, Jun 27, 2012 at 4:17 PM, Jeremy Kister asterisk...@jeremykister.com
wrote:
On 6/27/2012 3:44 PM, khalid touati wrote:
#, this happened:
-- Started music on hold, class 'default', on SIP/USPBX2-07d5
-- SIP/8425-07d4 Playing
Hi All,
I have a simple urgent question that I couldn't find the answer yet, can we
customize the voicemail attachment format *per user* in asterisk *1.2 *(like
all receive wav attch but one or two users receive attch in gsm format)? if
yes can you show me how please?
--
Khalid Touati
options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Khalid Touati
Network Administrator at Endosoft, LLC
CCNA
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New to Asterisk? Join
Good to hear you got it working with Digium's help. One thing: if
bri_presistentlayer means that the drivers will force the D-channel to
always be up then do not be surprised if BT disables the ISDN port. They
don't like it if a customer forces them to power the D-channel all the
time at their
list
To UNSUBSCRIBE or update options visit:
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Khalid Touati
Network Administrator at Endosoft, LLC
CCNA
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!?
no master found
I didn't change my config in my previous post, anyone familiar with this
type of errors?
On Wed, May 9, 2012 at 3:09 PM, khalid touati khalidtou...@gmail.comwrote:
Yeah they have a wonderful policy that says ISDN team are not
contactable :( thanks a lot!!
On Wed, May 9
Thank you Patrick for the detailed info, it does make perfect sense to me,
I never expected that Digium cards have such an problem!
On Thu, May 10, 2012 at 4:13 PM, Patrick Lists
asterisk-l...@puzzled.xs4all.nl wrote:
On 10-05-12 21:10, khalid touati wrote:
Hi All,
I did downgrade
Thank you Kevin! thanks Patrickhope a new release will come out soon!
On Thu, May 10, 2012 at 7:37 PM, Patrick Lists
asterisk-l...@puzzled.xs4all.nl wrote:
On 10-05-12 23:47, Kevin P. Fleming wrote:
On 05/10/2012 03:20 PM, khalid touati wrote:
Thank you Patrick for the detailed info
Patrick,
I got confused though is this true:
Any Asterisk soft+digium hdw = it doesn't work
Any Asterisk soft+sangoma hdw = it works
Patched asterisk soft+digium hdw = it will work (per Kevin)
On May 10, 2012 9:06 PM, khalid touati khalidtou...@gmail.com wrote:
Thank you Kevin! thanks Patrick
,
Judging from that bug report I *think*:
On 11-05-12 03:39, khalid touati wrote:
Patrick,
I got confused though is this true:
Any Asterisk soft+digium hdw = it doesn't work
There seem to be combinations that do work. It is my understanding from
that bugreport that an older libpri works
Should I understand that no Asterisk user has issues with ISDN system
access configuration from UK? or maybe no one is using Asterisk In UK :) ?
On Tue, May 8, 2012 at 12:46 PM, khalid touati khalidtou...@gmail.comwrote:
Hi All,
I am posting this thread with the hope that someone in UK
asterisk-l...@puzzled.xs4all.nl wrote:
On 09-05-12 18:46, khalid touati wrote:
Should I understand that no Asterisk user has issues with ISDN system
access configuration from UK? or maybe no one is using Asterisk In UK
:) ?
I have no idea. But other than the error you have given very
, khalid touati wrote:
Thank you for your answer, I think I posted dhadi version and so but
let me add more details and recap them below:
We are using asterisk 1.8.12.0 with dahdi 2.6.0, on CentOS 6.2. it's a
digium card 1HA8-0400BLF
output of dahdi_hardware: pci::04:08.0
Yeah they have a wonderful policy that says ISDN team are not contactable
:( thanks a lot!!
On Wed, May 9, 2012 at 3:06 PM, Patrick Lists
asterisk-l...@puzzled.xs4all.nl wrote:
On 09-05-12 20:57, khalid touati wrote:
Yeah sorry for that, I realized something is missing after I sent
://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users
--
Khalid Touati
Network Administrator at Endosoft, LLC
CCNA
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(R) error in state 7(Multi-frame established)
We are using asterisk 1.8.12.0 with dahdi 2.6.0, on CentOS 6.2. I can post
configuration and/or further information if needed.
--
Khalid Touati
Network Administrator
CCNA
Hi All,
I am having an issue with layer 2 in BRi connection configured using Misdn:
after the BRI line working fine, a technician from our phone company came
in to add another number, after testing with his ISDN phone and BRI line is
working, from our asterisk server it is not :(.
When I check
Hi All,
I am having an issue with layer 2 in BRi connection configured using Misdn:
after the BRI line working fine, a technician from our phone company came
in to add another number, after testing with his ISDN phone and BRI line is
working, from our asterisk server it is not :(.
When I check
20 Dec 2011, khalid touati wrote:
Thank you Raj,
so with VOIP license calls can go beyond our pbx to PSTN (india),
right, if so this what i needed to know to call Indian cellphone
from US (or other countries)
If your objective is to originate calls in the US (using whatever
technology
Hi All,
Because I am pretty sure we have people in this DL from India, I was hoping
to get the 100% accurate information, is it legal to make calls from any
coutry to Indian mobile phones through an Asterisk server based in India?
--
Khalid Touati
Network Administrator
or update options visit:
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Khalid Touati
Network Administrator at Endosoft, LLC
CCNA
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New
in this
case, is queues.conf will still be useful in the case figure?
--
Khalid Touati
Network Administrator at Endosoft, LLC
CCNA
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call1.tmp /var/spool/asterisk/outgoing
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *khalid touati
*Sent:* Friday, August 12, 2011 9:56 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re
Ok, for the variables, I can retrieve some of them like the caller number
and so on (I would assume that all the variables that last for duration of
call are there), but I still think that I sould not use the h extension to
continue after ReceiveFAX use, it's like not a lot of people use FFA,
@ Bryant: thanks so much for the interesting figure of use.
Why do so may people think their problems are unique. Many people use FFA
and spandsp. They all come across this. The issue is widely known, well
understood, and not at all strange once you think about it.
Steve
@ Steve: don't
Hi I wanted to help out with dial plan, but it's not obvious what you want
to achieve, also I do recommend to read the chapter that talks about
contexts and dialplan from future of asterisk book. but if you're in rush
just try to make clear how you want your system to behave and i'll be glad
to
Hi all,
I am running to the following problem, when using the below dialplan to
receive fax, everything works perfect till this line
exten = receive,n,ReceiveFAX(${FAXFILE}):
and then the following line cannot be executed, it's like asterisk can't go
back to dialplan and continue, the good news is
fax through email, and let me tell you
(first time using Free Fax from Asterisk) ReceiveFAX catch well faxes, just
a couple tries but got them all, let's see with more faxes what will happen.
On Sun, Jun 19, 2011 at 12:24 PM, khalid touati khalidtou...@gmail.comwrote:
Hi all,
I am running
Yeah I am using a TDM410P, thanks for the answer.
On Fri, Jun 3, 2011 at 4:30 AM, A J Stiles asterisk_l...@earthshod.co.ukwrote:
On Thursday 02 Jun 2011, khalid touati wrote:
Hi Guys,
Actually My question is as in the subject, may I use a regular phone line
to receive faxes with FFA (Fax
Hi Guys,
Actually My question is as in the subject, may I use a regular phone line to
receive faxes with FFA (Fax For Asterisk), I am using asterisk 1.6.2.8.
--
Khalid
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Also you guys may need to use:
sip.conf
[general]
allowguest=no
*alwaysauthreject = yes*
On Thu, Jun 2, 2011 at 1:01 PM, Al lists asteris...@gmail.com wrote:
I'll check this option and see if it helps next time,
just to clarify, there were no actual calls in place, just DOS register
attack.
Hi Guys,
I have a the following issue when I ma trying to place a call through my
voip provider, I am using an asterisk 1.2.21.1, do you have an idea what
could fix this issue (as you can see when the other party answered, the call
get dropped because of probably sip incompatibility)
Nov 12
Ok, it seems like I don't have g729 codec intsalled, can I install this
codec within asterisk 1.2 or just 1.4 and 1.6 are supported?
On Fri, Nov 12, 2010 at 2:56 PM, khalid touati khalidtou...@gmail.comwrote:
Hi Guys,
I have a the following issue when I ma trying to place a call through my
Hi Cassius,
it may be slightly offsubject but i did connect two 1.6 asterisk boxes, and
the only issue i had is these two statements missing:
calltokenoptional=209.16.236.73/255.255.255.0
requirecalltoken=no
hope it helps!
On Tue, Oct 19, 2010 at 12:54 PM, Paul Belanger
'33'*
what i want to do now is to make the user aware that he typed the wrong
number, so i looked for an option in MeetMe that announces the conf #, but
unfortunately it's not there, do you know any way to reach my goal?
thanks for any help!
**
On Tue, Sep 21, 2010 at 8:47 AM, khalid touati
actually same thing happened to us a year ago (under asterisk 1.2) we solved
the same day discovered by putting both:
allowguest=no
alwaysauthreject = yes
On Sun, Oct 3, 2010 at 7:17 PM, Barry Miller asterisk-us...@notanet.netwrote:
On Sun, Oct 03, 2010 at 02:19:35PM -0600, Greg Saunders
Hi List,
I did follow the procedure to install Free Fax for Asterisk successfully
till i came accross this isssue: i can't load the fax module:
pbx3*CLI module load res_fax_digium.so
Unable to load module res_fax_digium.so
Command 'module load res_fax_digium.so' failed.
[Sep 30 10:50:12]
give me
a link to know about that, cause i've never done it !
Thank you guys!
On Thu, Sep 30, 2010 at 11:23 AM, Kevin P. Fleming kpflem...@digium.comwrote:
On 09/30/2010 09:51 AM, khalid touati wrote:
Hi List,
I did follow the procedure to install Free Fax for Asterisk successfully
till i
.
David:
i see what you mean, you're right it's there,thank you for your help!
On Thu, Sep 30, 2010 at 12:09 PM, David Backeberg dbackeb...@gmail.comwrote:
On Thu, Sep 30, 2010 at 11:46 AM, khalid touati khalidtou...@gmail.com
wrote:
thanks for replies,
I am using Asterisk 1.6.2.11
Hi All,
I am using an Asterisk 1.6.2.6, and when I use this part of the dialplan:
exten = 8355,1,Dial(SIP/${EXTEN}IAX2/${EXTEN},18,tTWwr)
exten = 8355,n,Dial(IAX2/8366,48,tTWwr)
(i made that simple to exhibit issue)
I got just 1 ring in 8366 extension before it hangup, what i noticed is the
and Paul!
On Fri, Sep 24, 2010 at 6:54 PM, Steve Edwards asterisk@sedwards.comwrote:
On Thu, Sep 23, 2010 at 10:06 AM, khalid touati khalidtou...@gmail.com
wrote:
do you guys know how i can turn debug on or just know why it's not
getting enabled?
On Fri, 24 Sep 2010, Paul Belanger wrote
Hi Guys,
i could turn debug on in a asterisk 1.6 box (by enabling debug in
logger.conf and core set debug to 0), but my issue is i cannot enable
debugging in a 1.2 box by doing the same 2 steps, also this is a production
server so i can't restart with debug enabled, do you guys know how i can
...@lists.digium.com] On Behalf Of khalid
touati
Sent: 20 September 2010 14:06
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Not able to join conference
it's going to put you in conf no 500 without prompting you to enter a
conference number I guess
, Andrew Thomas a...@datavox.co.uk wrote:
What happens if you put in a 'room' number?
Eg: exten = 8080,3,MeetMe(500|MDci)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of khalid
touati
Sent: 17 September
Hi All,
We are running to a weird problem, we're using asterisk 1.2 as a production
server (I'm wiling to move very soon to more recent version) and our problem
is when somebody try to join a conference he's told that he's the only one
in the conference but in fact there is some 3 or 5 or whatever
On Fri, Sep 17, 2010 at 9:24 AM, khalid touati khalidtou...@gmail.com
wrote:
in the dialplan, that would be a big help if you guys can help diagnose
the
issue.
A debug log of the actually problem will be more helpful.
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan
thank you guys for your responses!
sorry, actually i was not accurate in asking this question, my search is
restricted to soft-phone to use within Black Berry and integrate with
Asterisk.
It seems like the cheapest solution available now that you can integrate
with asterisk and install in Black
Hi All,
i have a question, is there any soft-phone available for Black Berry use,
I've been told there is a firefly one, but when i looked, i found nothing,
is any body has an update on this please?
--
_
-- Bandwidth and
so nobody seems to like dealing with fax!!
2010/7/12 khalid touati khalidtou...@gmail.com
Hi Guys,
i am using the latest version of asterisk 1.4 (1.4.33.1), dahdi (2.3.0.1)
and FFA (Applications: 1.4_1.2.0, Digium FAX Driver: 1.4_1.2.0). the issue
i'm having is that i'm able to receive faxes
'Fax for Asterisk' is a commercial application sold by Digium. This is not
their official support channel. Since you paid for the product, why not
contact them directly about your problem?
i did get this version for free after buying a (actually several) digium
telephony card, but i realized that
Glad you found the issue, sorry for not being able to help.
2010/7/9 Paul Belanger paul.belan...@polybeacon.com
On Fri, Jul 9, 2010 at 7:38 AM, Adil Zaaraoui adilzeaara...@yahoo.fr
wrote:
ok it works i had a problem with a syntax:
i had to wrire:
exten
2010/7/8 Kyle Kienapfel doctor.w...@gmail.com
On Wed, Jul 7, 2010 at 10:34 AM, Adil Zaaraoui adilzeaara...@yahoo.fr
wrote:
Dear list.
Is it possible to use both IAX2 and SIP protocole during a dial?
Illustration:
I have peer A communicate with my Asterisk using IAX2 protocole.
I
Hi Guys,
for people who may have the same issue:
i was just not using STRFTIME the right way, after consulting docs, i'm
using it like this:
exten =
,n,Set(FAXFILENOEXT=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},America/New_York,%F_%T)})
instead of this:
exten =
thanks i'll keep that in mind.
2010/6/2 Roderick A. Anderson raand...@cyber-office.net
khalid touati wrote:
Hi Guys,
for people who may have the same issue:
i was just not using STRFTIME the right way, after consulting docs, i'm
using it like this:
exten =
,n,Set(FAXFILENOEXT
thank you Barry, you're right, it is also working.
well, happy that i have a bunch of choices that work (after wrong output).
thanks for all!
2010/6/2 Barry Miller asterisk-us...@notanet.net
On Wed, Jun 02, 2010 at 10:26:12AM -0400, khalid touati wrote:
Hi Guys,
for people who may have
Hi Guys,
I'm having a non-obvious issue, i am using Fax for asterisk to receive
faxes, so when i test using a website that send faxes it's working great:
the fax is received and the fax2mail app is called and i get it in my email
box. but when i try using a regular fax machine everything in logs
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *khalid touati
*Sent:* Tuesday, May 04, 2010 11:35 AM
*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 *khalid touati
*Sent:* Wednesday, May 05, 2010 8:36 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Code
Hi Guys,
so when i dial from an asterisk 1.2 to asterisk 1.4 i get the following
warning:
WARNING[640]: file.c:738 ast_readaudio_callback: Failed to write frame
is anyone familiar with?
2010/4/29 khalid touati khalidtou...@gmail.com
Hi Guys,
Danny: as i said from pbx1 (1.4) to pbx2 (1.2) it's
Hi Guys,
i spent some time to figure this out (since i love how dialplan is written)
but i decided to ask for your help guys.
i have two asterisk servers one is 1.2 the other is 1.4, from 1.4 (pbx1) to
1.2 (pbx2) i can leave a voice mail without any pb, but from pbx2 to pbx1 it
just hang up.
in
In PBX1, where are you actually dialing the phone? The first line of the
macro just says “goto dialstatus” with no Dial statement.
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *khalid touati
*Sent:* Thursday, April 29
/4/19 Steve Edwards asterisk@sedwards.com
Un-top-posting...
2010/4/14 khalid touati khalidtou...@gmail.com
i've connecting two pbx server successfully for several times using
the following config:
register = USPBX:myp...@122.11.176.35uspbx%3amyp...@122.11.176.35
is
set to trTWw that would work regardless of the other side but didn't try
though. have a headeache-free experience with asterisk the future of
telephony :)!
2010/4/14 khalid touati khalidtou...@gmail.com
Hi Guys,
i've connecting two pbx server successfully for several times using the
following
Hi Guys,
i've connecting two pbx server successfully for several times using the
following config:
register = USPBX:myp...@122.11.176.35 uspbx%3amyp...@122.11.176.35
[PBX1]
type=friend
host=122.11.176.35
trunk=yes
sercret=mypass
context=external
deny=0.0.0.0/0.0.0.0
Hi Guys,
i have a weird thing here: when using time variables (%F %T) in a shell
script, out of dial plan (particularly system() app); it displays the right
time (same as output of date), but when same variables are used in system()
application it displays a wrong time/date (ahead of 6 hours). I
Hi Guys,
i wanted to share this with u and ask for little help at the same time:
i used iptables to secure my server, so i wnet ahead and blocked avery thing
except a couple of domain protocols and UDP ports of SIP, IAX2 and that
range 15000 to 2, tested it and OK. when in production, the
...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *khalid touati
*Sent:* Tuesday, April 13, 2010 1:08 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Time variables in system application
Hi Guys,
i have a weird thing here
DNS!! i believe it has to do with call setup and rtp protocol cause all
devices shows as sip peers at the call time, but not 100% sure. any iptables
plz :) !
2010/4/13 Gordon Henderson
gordon+aster...@drogon.netgordon%2baster...@drogon.net
On Tue, 13 Apr 2010, khalid touati wrote:
Hi Guys
i believe not only today :D, but thank u anyway for the spirit of helping
people!!
2010/4/13 Danny Nicholas da...@debsinc.com
Just what I thought - guess that's the X'th time I wuz wrong today.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
any clue Guys???!!!
2010/4/5 khalid touati khalidtou...@gmail.com
Hi Juan,
my system is an asterisk 1.2 on gentoo, it is configured to receive faxes
through rxfax and then to use fax2email to convert the tiff to pdf and send
it to front desk:
exten =
3772,1,Set(FAXFILE=/var/spool/asterisk
Hi Guys,
i have a small issue but bothering me, after restarting asterisk (version
1.4 running on centos) i have the following message that comes repeatedly
when i am connected to the CLI:
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
--
...@lists.digium.com] *On Behalf Of *khalid touati
*Sent:* Monday, April 05, 2010 10:29 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Continuous bothering message -- Remote UNIX
connection disconnected
Hi Guys,
i have a small issue but bothering
can anyone help me out in this, a big number of my faxes are lost everyday!
i would really appreciate any help on how i can tweak asterisk (rxfax) to
receive all faxes!
2010/4/2 khalid touati khalidtou...@gmail.com
i went ahead and i used this line:exten = 3772,n,rxfax(${FAXFILE}|debug
is to navigate away from the home screen of FreePBX. Verbose should stay at
3 if you want to see what is happening on your asterisk server.
Zeeshan A Zakaria
--
Sent from my Android phone with K-9 Mail.
On 2010-04-05 11:09 AM, khalid touati khalidtou...@gmail.com wrote:
Thank you Danny, i did
!
Regards,
Juan
khalid touati wrote:
can anyone help me out in this, a big number of my faxes are lost everyday!
i would really appreciate any help on how i can tweak asterisk (rxfax) to
receive all faxes!
2010/4/2 khalid touati khalidtou...@gmail.com
i went ahead and i used this line:exten
that was
it two instances.
Thanks to all of you guys, that was a ggod help!
2010/4/5 Watkins, Bradley bradley.watk...@compuware.com
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *khalid touati
*Sent
Hi Guys,
do any body know how to receive debug info on RxFAX application? i am
experiencing a lot of fax failures and can't guess the reason behind.
Thank you very much for any help!
--
Abdullah
--
_
-- Bandwidth and Colocation
, khalid touati wrote:
Hi Guys,
do any body know how to receive debug info on RxFAX application? i am
experiencing a lot of fax failures and can't guess the reason behind.
Thank you very much for any help!
They have a hard-wired log file. Make sure Asterisk can write
/4/2 khalid touati khalidtou...@gmail.com
thank you guys for responses,
Danny-, am i going to receive debug info in the CLI or in a default file
(/var/log /*)? is FAXOPT supported whitin asterisk 1.2 (sorry I forgot to
mention that i am using 1.2)?
2010/4/2 Tzafrir Cohen tzafrir.co
/27 khalid touati khalidtou...@gmail.com
Thank you very mutch Philip, i'll use these commands and get back with the
output.
2010/3/26 Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de
Hi!
it should be some commands that can give me a better idea about the
codecs, if anyone know
:) all users are having the same issue, even those connected to this server
from abroad!
2010/3/29 Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de
i have the same model polycom phone configured with another server
(asterisk 1.4), and guess what no noise at all. any guess!
Thank you very mutch Philip, i'll use these commands and get back with the
output.
2010/3/26 Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de
Hi!
it should be some commands that can give me a better idea about the
codecs, if anyone know them, please help!
Use sip show
Hi Philip,
So i looked at the codecs in the device (polycom) it says only G.711 and
ulaw can be used, i made an internal call using two phones that are
configured just with sip (so IAX not involved) but the static noise is
there, i typed show sip peer username and this is the only thing i got:
Hi Guys,
i have recently connected my (working) asterisk 1.2 server, with two 1.4
asterisk servers (one using SIP the other using IAX), since then (i believe)
people starts complaining about a high background noise when using the
handset on Polycom phones (but when using the speaker it's fine, and
OK Guys i got fixed the phones i was using were registered in both servers
which is not good, once i removed them it started working!
2010/3/4 khalid touati khalidtou...@gmail.com
Hi Guys,
i am using the following config in pbx1:
register = pbx1:endop...@172.16.200.175 pbx1%3aendop
-- Called pbx2/8021
[Mar 4 16:49:13] WARNING[3392]: chan_sip.c:12679 handle_response_invite:
Received response: Forbidden from 'Khalid Touati
sip:8...@172.16.200.176 sip%3a8...@172.16.200.176;tag=as1dcf5ff2'
-- SIP/pbx2-09cf4468 is circuit-busy
== Everyone is busy/congested at this time (1
file which is normally [default]
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *khalid touati
*Sent:* Friday, January 29, 2010 11:54 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re
outbound context.
[iax-inbound]
Include = outbound-conext
[outbound-context]
Exten = _1NXXNXX,1,Dial(DAHDI\g1\${EXTEN})
Something like that.
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *khalid
And calls from Here to There are Dial(IAX2/SRV2-SRV1/X) where is
in destination “from-iax” context
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *khalid touati
*Sent:* Tuesday, January 26, 2010 10:11 AM
Hi Guys,
i am using two PBX's i can call from pbx1 a cellphone tied to pbx2 that way:
1) use a phone in PBX1
2) call extension in PBX2
3) extensions in PBX2 ring to a cellphone (as this specific ext is tied to a
cellphone)
my questions now is : am i gonna be able to dial from an IPphone
Hi All,
i want to make an extension from pbx1 able to tlak to another extension from
pbx2 or use pbx2's trunk to dial outside calls.
so i edited in both servers accordinally the iax.conf:
register = pbx1:p...@172.16.200.175 pbx1%3ap...@172.16.200.175
[pbx2]
type=friend
host=dynamic
trunk=yes
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