No . I don't receive any error message after converting from *.wav to *.gsm
but the new announcements cannot be heared (when trying for playback).
On Tue, Sep 8, 2009 at 6:37 AM, Roel Sarmiento
technomage.scratchbu...@gmail.com wrote:
Do you have an error message?
On Tue, Sep 8, 2009 at
check the file formats first if .wav is listed there and if it is, then
check the translation if its activated.
On Tue, Sep 8, 2009 at 2:05 PM, hadi motamedi motamed...@gmail.com wrote:
No . I don't receive any error message after converting from *.wav to
*.gsm but the new announcements
Thank you . Please be informed that the *.wav files cannot be played on my
Asterisk so I had to convert to *.gsm file format .I tried to convert to
*.gsm by making use of sox but the new announcement cannot be heard .
On Tue, Sep 8, 2009 at 7:22 AM, Roel Sarmiento
On Mon, Sep 07, 2009 at 01:47:57PM -0400, Steve Totaro wrote:
On Mon, Sep 7, 2009 at 10:09 AM, Tzafrir Cohen
tzafrir.co...@xorcom.comwrote:
On Mon, Sep 07, 2009 at 07:44:07AM -0400, Steve Totaro wrote:
On Mon, Sep 7, 2009 at 5:58 AM, Tzafrir Cohen tzafrir.co...@xorcom.com
wrote:
/etc/dahdi/system.conf file is auto generated do we need to change in this
file as we do for zaptel ?
Any working examples
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Does asterisk support doing shared call appearances with polycom 550 and 650
phones? I know that a line can be changed to type shared on the polycom
but is it possible to put a call on hold with one phone and resume the held
call with another? Does this work by default or is there some special
I'm not quite sure but i think if you converted the file ex: file.wav using
sox it should produce something like file.ulaw, file.alaw, file.gsm. Check
if its there, then check the translation if you have the codec activated, it
worked for me before.
On Tue, Sep 8, 2009 at 2:40 PM, hadi motamedi
just a hint. you might have # assigned the moh in feature.conf and #3
to starting the recording. check your feature.conf and makesure that #
isn't assigned to anything.
erik de wild
Tripple-o
Your Asterisk migration partner
the Netherlands
Verstuurd vanaf mijn iPhone
Op 7 sep 2009 om 20:40
Hellos,
I need to send personal directory from asterisk to the ersonal directory of
the linksys spa 942. Is this possible?
--
Best Regards,
James Mutuku Ndeti
Agile Systems Limited
+254722490994
www.agile.co.ke
mutuku.wordpress.com
Has your organization implemented a customer relationship
Dear All
I sent you a message regarding my problem with Asterisk Call Parking feature
and you told me that needs to check the polycom sip.cfg file . But my
Asterisk doesn't have sip.cfg file . Can you please let me know how can I
overcome ?
___
--
Hello,
I am using Asterisk 1.4.22 in a debian 4.0 with a kernel version 2.6.18-6-amd64
(SMP). The processor type is Intel(R) Xeon(R) CPU E5420 @ 2.50GHz.
Sometimes, I get a strange behavior from asterisk: The CLI commands does not
work and Asterisk cannot receive calls. The output of every CLI
Erik,
I have placed everything in features.conf in comment ( ; ). Still when I
run show features, I get this :
clarkconnect*CLI show features
Builtin Feature Default Current
--- --- ---
Pickup*8 *8
Blind Transfer
Thank you . Please find below my original and converted sound files
attributes on my Asterisk :
#file FR1.wav
FR1.wav: RIFF (little-endian) data, WAVE audio, ITU G.711 A-law, mono
8000 Hz
#file FR1.gsm
FR1.gsm: data
Can you please let me know what is the problem as the sox does
is there an error on the asterisk cli when you're playing the sound file?
On Tue, Sep 8, 2009 at 4:19 PM, hadi motamedi motamed...@gmail.com wrote:
Thank you . Please find below my original and converted sound files
attributes on my Asterisk :
#file FR1.wav
FR1.wav: RIFF
you should check dialstatus and gotoif. if you use both in the proper
way ( see the wiki) then you have the dialplan behaviour you are
looking for.
erik de wild
Tripple-o
Your Asterisk migration partner
the Netherlands
Verstuurd vanaf mijn iPhone
Op 7 sep 2009 om 21:26 heeft Miguel
using mixmonitor might not be such a good idea. afaik the mixing of
the recordings of the two channels starts after ending the call
causing a high cpu load. if you have recordings going on all the time
moving the 2 files that has to be mixed to a dedicated mixing server
might be a good
8 sep 2009 kl. 10.17 skrev jonas kellens:
Erik,
I have placed everything in features.conf in comment ( ; ). Still
when I run show features, I get this :
clarkconnect*CLI show features
Builtin Feature Default Current
--- --- ---
Pickup
Hi,
we're using a GSM-Gateway on asterisk to forward incoming calls to the
cellphones, but, of course, the cellphones always display the callerid from
the gateway. Does anyone know a symbian app that could (on an incoming call)
connect via grps/3G to a database behind the asterisk and fetch the
When I enable the automon-feature (*1) the callee can start recording
the conversation. No problem there.
But I can't get my user-defined features to work.
I have setup the following test-feature in features.conf :
[applicationmap]
testfeat = *3,self/callee,Playback,tt-weasels
I have the
Please find below the error message that I am receiving on my Asterisk :
-- Executing [s-noans...@macro-dialuser:4] Playback(Zap/95-1, FR1)
in new stack
[Sep 7 11:11:34] WARNING[7624]: format_wav.c:140 check_header: Not a wav
file 6
[Sep 7 11:11:34] WARNING[7624]: file.c:316 fn_wrapper:
Just a hint based on experience. Run top from de linux prompt to
check if any proces causes an enormous cpu load. I once ran into the
same behaviour because some asterisk related php script looped and
took almost all the cpu power available.
erik de wild
Tripple-o
Your Asterisk migration
On 8 Sep 2009, at 10:22, Jay R. Worthington wrote:
we're using a GSM-Gateway on asterisk to forward incoming calls to
the cellphones, but, of course, the cellphones always display the
callerid from the gateway. Does anyone know a symbian app that could
(on an incoming call) connect via
Again, how many calls were you able record using RAMdisk? Anywhere 300?
As I stated before, this is going to be dependent on how you're
manipulating the calls and the gear you're running on. The nice thing
about your 'just broadcast the entire LAN to the recording solution'
is that the
Hi all,
I'm fighting with a really strange problem that is really busting me.
I have an asterisk 1.4.22 ( from a trixbox 2.6.2 ) and mISDN 1.1.7
3 extension on hardphone and 3 extension in softphone ( zoiper )
What happens is that sometimes the people on the other side of communication
hear my
- Ian Wang iyu.w...@gmail.com wrote:
confBridge in Asterisk 1.6.2.0-rc1 doesn't stable.
It causes segment fault very often and results in asterisk crash.
It would be extremely useful if you could file an issue on
https://issues.asterisk.org/
with details about how you are using it
Asterisk sometimes goes to sleep. (And never wakes-up).
Restart it and all will be fine again.
We have a watchdog which sends SIP OPTIONS packet to Asterisk and if it does
not respond – restarts it.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
On Tuesday 08 September 2009 00:14:53 Asterisk User wrote:
Thanks Tilghman for your quick reply.
I know that we should set variables through IAXVAR on source server to
access them on Destination server.
I just wanted to know the reverse case, where IAX channel variables set on
destination
On Tue, Sep 8, 2009 at 5:08 AM, Erik de Wildi...@meetmecall.nl wrote:
using mixmonitor might not be such a good idea. afaik the mixing of
the recordings of the two channels starts after ending the call
causing a high cpu load.
Incorrect.
The 'mix' in Mixmonitor() is that two legs of a call are
On Tue, 8 Sep 2009, hadi motamedi wrote:
I sent you a message regarding my problem with Asterisk Call Parking feature
and you told me that needs to check the polycom sip.cfg file . But my
Asterisk doesn't have sip.cfg file . Can you please let me know how can I
overcome ?
sip.cfg is not an
I just have a T1 TE121 Card, if you want I can send you my file.
What kind of card is the TE420P, I think the card is for 4 T1/E1 card, Am I
right?
De: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] En nombre de ABBAS SHAKEEL
Enviado el: Lunes, 07 de
8 sep 2009 kl. 15.40 skrev Benny Amorsen:
I see a lot of these on an otherwise idle Asterisk 1.6.0.15:
Extension Changed 773[Hints] new state Ringing for Notify User
792-00041327d17e-1. Then a little while later it changes to InUse or
Idle, completely randomly. It happens for many different
Hello,
I have 2 sites. One(Site 1) has an asterisk PBx and the Other(site 2) has 2
remote soft phones. The latency btw both sites is btw 500ms-700ms. I know
this is a shot in the dark...but are there ways of improving the voice
quality for the remote calls(btw site 1 and site 2), Other than
I see a lot of these on an otherwise idle Asterisk 1.6.0.15:
Extension Changed 773[Hints] new state Ringing for Notify User
792-00041327d17e-1. Then a little while later it changes to InUse or
Idle, completely randomly. It happens for many different combinations of
phones and watchers.
There are
I imagine this setup will need those two communicating entities to
be part
of the pabx. But support extension 100 of PABX A (legacy) calls 101
on the
same platform. I want asterisk connected to PABX A via E1/T1 to
know about
that call and start recording (tap) without bridging or
Hello Every one!
I am little bit new to asterisk. I am doing research on different telecom
options as well.
I have question for you professionals
In order to get E1 line working with Asterisk. What E1 line parameters need
to be specified in Asterisk(configuration files).
They vary from country
Hi
Is anybody using these ?
Gaetan
Begin forwarded message:
From: Gaëtan Minet gminet...@mcit.be
Date: Sat 22 Aug 2009 16:29:42 GMT+02:00
To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com
Subject: [asterisk-users] Patton smartnode 463x (BRI) 25ms
Hi everyone,
I have a problem at my Trixbox that is version Asterisk 1.2.26.1 svn
rev 79171 and 2.6.9-34.0.2.ELsmp kernel version. Two Digium 4fxs+4fxo
card has been installed and everything was working before made yum
update and at this server. (Centos 4.0). After update I faced with
zaptel not
Thanks Juan!
Yeah you are exactly right. Please send me your file. thanks
On Tue, Sep 8, 2009 at 7:40 PM, Juan Cardoza jcard...@tpmex.com wrote:
I just have a T1 TE121 Card, if you want I can send you my file.
What kind of card is the TE420P, I think the card is for 4 T1/E1 card, Am I
Every time I upgrade, I run into more issues than I previously had, so I
tend to stay where I am unless I absolutely have to upgrade. I ran 1.0.3
for 2+ years with no issues. I upgraded to whatever the latest 1.2 was at
the time and it crashed three times within a week. 1.2.32 and .34 seem to
On Mon, Sep 7, 2009 at 7:50 PM, Greg Woods g...@gregandeva.net wrote:
On Mon, 2009-09-07 at 07:20 -0600, Greg Woods wrote:
incoming calls
through the FXO line are dropped as soon as there is a button press.
The error logged is:
[Aug 23 18:15:39] WARNING[6532] chan_dahdi.c: Cannot
On Tue, Sep 08, 2009 at 07:11:26PM +0500, silent sayz wrote:
Hello Every one!
I am little bit new to asterisk. I am doing research on different telecom
options as well.
I have question for you professionals
In order to get E1 line working with Asterisk. What E1 line parameters need
to
Carlos Chavez escribió:
I just upgraded from 1.6.0.14 to 1.6.1.6 and now my realtime static
configuration for extensions.conf will not load.
Just curious, is there any specific reason for you to upgrade from the
latest 1.6.0.14 to 1.6.1?
Cheers,
--
Ing. Miguel Molina
Grupo de
A deadlock? In 1.2? Really? :)
Peder wrote:
I am running 1.2.34 (also tried on 1.2.32) and whenever I launch an AGI, I
get the avoided deadlock message below.
*CLI == Spawn extension (CONTEXT3, 6080, 8) exited non-zero on
'SIP/3211-1-081c40a8'
-- Executing
On Tue, 2009-09-08 at 13:03 +1200, Matt Riddell wrote:
On 8/09/09 5:35 AM, Carlos Chavez wrote:
Today is a strange day. My asterisk server is suddenly saying that all
extensions are on hold. All my hints are like this:
-= Registered Asterisk Dial Plan Hints =-
I am running 1.2.34 (also tried on 1.2.32) and whenever I launch an AGI, I
get the avoided deadlock message below.
*CLI == Spawn extension (CONTEXT3, 6080, 8) exited non-zero on
'SIP/3211-1-081c40a8'
-- Executing NoOp(SIP/3211-1-081c40a8, ) in new stack
-- Executing
Hi,
I'm using asterisk 1.4.22-4 in Trixbox with snom 360 phones. When
calls come in on our POTS lines, the caller id shows up like
555-555-1...@192.168.1.10 where 555-555-1234 is the correct phone
number and 192.168.1.10 is my pbx server IP. This format does not work
for redialing on
I just upgraded from 1.6.0.14 to 1.6.1.6 and now my realtime static
configuration for extensions.conf will not load. All other realtime
configs work (SIP, IAX2, Voicemail). I cannot find any reference or
documentation about the structure of the realtime static database for
1.6.1.x but I
Peder wrote:
I am running 1.2.34 (also tried on 1.2.32) and whenever I launch an
AGI, I get the avoided deadlock message below.
On Tue, 8 Sep 2009, Alex Balashov wrote:
A deadlock? In 1.2? Really? :)
Well, that was helpful.
As a fellow 1.2 Luddite, I have boxes running xxx
Thanks Tzafrir Cohen!
May be i get this wrong
http://www.voip-info.org/wiki/view/Asterisk+PRI#CountryVariations
Any body help me what i must know about the E1 cable before asking a company
to give me an E1 connection for Asterisk Digiums Card.
Can i get complete list ???
Like Some one Advice
We have some installations with smartnode 4554, (same tail echo
cancellation) without problems so far.
Jorge Mendoza
Gaëtan Minet wrote:
Hi
Is anybody using these ?
Gaetan
Begin forwarded message:
*From: *Gaëtan Minet gminet...@mcit.be mailto:gminet...@mcit.be
*Date: *Sat 22 Aug 2009
Un-top-posting...
Peder wrote:
I am running 1.2.34 (also tried on 1.2.32) and whenever I launch an
AGI, I get the avoided deadlock message below.
Sep 8 11:48:43 WARNING[564]: channel.c:780 channel_find_locked:
Avoided initial deadlock for '0x818edf8', 9 retries!
On Tue, 8 Sep 2009,
On Tue, 2009-09-08 at 11:54 -0500, Miguel Molina wrote:
Carlos Chavez escribió:
I just upgraded from 1.6.0.14 to 1.6.1.6 and now my realtime static
configuration for extensions.conf will not load.
Just curious, is there any specific reason for you to upgrade from the
latest 1.6.0.14
*I am getting below CLI in my asterisk :*
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing AGI(SIP/cc101-b7910cc0, agi://127.0.0.1:4577/call_log)
in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial(SIP/cc101-b7910cc0,
Carlos Chavez cur...@telecomabmex.com writes:
On Tue, 2009-09-08 at 11:54 -0500, Miguel Molina wrote:
Carlos Chavez escribió:
I just upgraded from 1.6.0.14 to 1.6.1.6 and now my realtime static
configuration for extensions.conf will not load.
Just curious, is there any specific reason
I have 2 sips configured :
1) register =sama:xx...@209.51.191.xxx:5060
2) register =sama:xx...@209.51.192.xxx:5060
Both are active.
5060 port will be same or different ?
On Wed, Sep 9, 2009 at 12:29 AM, David @ULC ucoms2...@gmail.com wrote:
*I am getting below CLI in my asterisk :*
Hi,
Asterisk database is made of familykey records such as:
fam key1 val1
fam key2 val2
...
fam key100 val100
I'm looking for the smartest way to iterate among different keys associated
to a given family.
One way to do this is to parse database show fam response.
Is there something
- Barry Miller asterisk-us...@notanet.net wrote:
On Fri, Sep 04, 2009 at 04:10:43PM -0500, Doug Bailey wrote:
- Barry Miller asterisk-us...@notanet.net wrote:
Hi,
Using 1.4.26.1 DAHDI 2.2.0.2, FSK VMWI devices off a TDM840
work
fine.
With
That's the general idea. The application is designed to send a TIFF over an
established connection. You can detect that it is a fax or just assume so.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joan Antoni
Terre
Hi everybody,
I've installed Free Fax For Asterisk in my Asterisk box but I don't
understand how it works as when using SendFax application from dialplan, I
can't find how to introduce destination fax number.
How this application works? Do I have to call destination fax using Dial
application,
thanks Danny,
just another stupid question, as far as I know, when a call is answered
after Dial application, it doesn't execute other dialplan priorities until
it's hung up, which execute h priority, so how can I make it execute a
SendFAX, or whatever else, when it's answered?
thanks again
Please chime in if you've ever wished for digium to make a 4-port daughter
board with a combination of 2FXO AND 2FXS ports on the same card.
When using the 800 series cards, one must either choose 4-port permutations
of FXS/FXO, OR one must give up 2 valuable ports.
In other words, when you add
You can use the M parameter to run a macro after the channel picks up
or the g parameter to jump to a given context. there is also a
parameter to run an AGI script. Check the dial() cmd on the wiki for
further details.
Erik de Wild
Tripple-o
Your Asterisk migration partner
the Netherlands
Un-top-posting...
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of abdelkader
I am using Asterisk 1.4.22 in a debian 4.0 with a kernel version
2.6.18-6-amd64 (SMP). The processor type is Intel(R) Xeon(R) CPU E5420 @
2.50GHz.
I have a cron job that restarts Asterisk every night.
This is supposed to be an old Asterisk best practice for 1.2.* but I think it
does not harm.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Definitely, 10 votes from me.
For the home user, 2xFXO + 6FXS, in a single slot small profile box is
ideal, but only able to offer 2xFXO + 4xFXS at the moment.
SIP phones don't exactly have the appropriate WIFE factor. A standard off
the shelve, no frills phone does the job.
Alec Davis
Thank you for your message . But I tried to find it on my server , as the
followings :
#find / -name sip.cfg -print
But it didn't return any result . Can you please let me know where can I
find it ?
On Tue, Sep 8, 2009 at 2:58 PM, Steve Edwards asterisk@sedwards.comwrote:
On Tue, 8 Sep
On Wed, 9 Sep 2009, hadi motamedi wrote:
Thank you for your message . But I tried to find it on my server , as the
followings :
#find / -name sip.cfg -print
But it didn't return any result . Can you please let me know where can I
find it ?
You probably have not setup central provisioning
Hello,
How can I reset CDR time , let's say after 30 seconds of answer signal,
reset CDR to 0 , any idea ??
Thanks.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15 Phoenix, Arizona
At 07:31 PM 9/8/2009, you wrote:
For the home user, 2xFXO + 6FXS, in a single slot small profile box is
ideal, but only able to offer 2xFXO + 4xFXS at the moment.
Wow, I can't imagine ever using an analog phone on Asterisk. SIP
phones are just so much better!
Ira
On Tue, 8 Sep 2009, Ira wrote:
At 07:31 PM 9/8/2009, you wrote:
For the home user, 2xFXO + 6FXS, in a single slot small profile box is
ideal, but only able to offer 2xFXO + 4xFXS at the moment.
Wow, I can't imagine ever using an analog phone on Asterisk. SIP
phones are just so much better!
On 9/09/09 4:34 PM, B.Masoud @ SH wrote:
Hello,
How can I reset CDR time , let’s say after 30 seconds of answer signal,
reset CDR to 0 , any idea ??
:) Use the ResetCDR application?
--
Cheers,
Matt Riddell
Director
___
A little more help is appreciated, I know about ResetCDR() , but I want some
code that resets the call data after 30 seconds!
And where to put the code exactly.
Thanks.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
On 9/09/09 5:14 PM, B.Masoud @ SH wrote:
A little more help is appreciated, I know about ResetCDR() , but I want some
code that resets the call data after 30 seconds!
And where to put the code exactly.
What a strange request. Why exactly are you wanting to do this?
If you're wanting all your
On Monday 07 September 2009 16:27:30 Carlos Chavez wrote:
It seems that older Aastra phones (9112i, 9133i, 480i, 480i CT)
have a problem with the new SIP implementation in Asterisk 1.6.X that makes
them unable to dial. They can receive calls but when you attempt to dial
the phone
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