On 23/10/2019 12.33, Joshua C. Colp wrote:
>
>> Does Asterisk use the readline library? Does it use /etc/inputrc ?
>> Can the behavior described above be configured ?
>
> It uses the editline library and all usage of it is within asterisk.c[1].
> There is no configuration ability as far as I know,
On Tue, Oct 22, 2019, at 11:44 PM, Fourhundred Thecat wrote:
> Hello,
>
> I have Asterisk 16.2 on Debian.
>
> In the Asterisk CLI, I would like to change 2 things:
>
> 1) change the keybindings for commandline editing
> (what in bash is called "readline" editing of the command line)
>
> The C
Hello,
I have Asterisk 16.2 on Debian.
In the Asterisk CLI, I would like to change 2 things:
1) change the keybindings for commandline editing
(what in bash is called "readline" editing of the command line)
The CLI is missing some very useful keybindings, and even worse, has
misconfigured oth
On Wednesday 15 August 2018 at 23:10:21, Dovid Bender wrote:
> Hi,
>
> I am trying to install wanpipe
Tell us how you are trying to install things.
> with dahdi on a CentOS7 box and I am running in to a few issues. My setup.
>
> CentOS 7
> asterisk-15.5.0
> libpri-1.6.0
> dahdi linux and dahdi
Hi,
I am trying to install wanpipe with dahdi on a CentOS7 box and I am running
in to a few issues. My setup.
CentOS 7
asterisk-15.5.0
libpri-1.6.0
dahdi linux and dahdi tools - 2.11.1
There are two issues.
1) For some reason dahdi_tools isnt being built.
2) When I try to load chan_dahdi and I
On Thu, Apr 6, 2017, at 12:17 PM, Richard Kenner wrote:
> > I would say this is a bug in func_speex and not in codec_siren14. This
> > is because the datalen is zero.
>
> Ah! So, like?
>
> *** func_speex.c.orig 2017-02-13 15:00:19.0 -0500
> --- func_speex.c2017-04-06 11:16:03.
> I would say this is a bug in func_speex and not in codec_siren14. This
> is because the datalen is zero.
Ah! So, like?
*** func_speex.c.orig 2017-02-13 15:00:19.0 -0500
--- func_speex.c2017-04-06 11:16:03.0 -0400
***
*** 185,189
}
!
On Thu, Apr 6, 2017, at 10:57 AM, Richard Kenner wrote:
> I'm seeing Asterisk crashes with the following frame at func_speex.c:188:
>
> (gdb) p *frame
> $6 = {frametype = AST_FRAME_VOICE, subclass = {integer = 0,
> format = 0xe2f9e20, frame_ending = 0}, datalen = 0, samples = 640,
> malloc
I'm seeing Asterisk crashes with the following frame at func_speex.c:188:
(gdb) p *frame
$6 = {frametype = AST_FRAME_VOICE, subclass = {integer = 0,
format = 0xe2f9e20, frame_ending = 0}, datalen = 0, samples = 640,
mallocd = 1, mallocd_hdr_len = 232, offset = 64,
src = 0x2ac07413e7f8 "
Thanks Annus, Amit.
Yes, Amit, the plus sign in front is necessary. I was able to get this to
work by changing the codecs that the SIP trunk will use. I had to set up
wireshark on my Asterisk instance, gather that it didn't work for a 488 SIP
"No acceptable here" message, which led me to discoveri
Hi
Your extensions.conf should have +17775551212 extension and not 17775551212
Add + sign before your number. This should solve your issue.
[from-external]
exten => +17775551212,1,Log(WARNING, TWILIO)
same => n,Hangup()
*Thanks & Regards,*
Amit Patkar
--
_
Thanks for your quick responses, Annus.
As you can see, in my original post, I forward to context "from-external".
I forward to "from-twilio-remove-plus" only to check if
http://orourketech.com/elastix-plus-sign-caller-id-messing-things/ will
solve my problem. It did not. At no point do I point it
Maybe is because now it's a different context:
from-twilio-remove-plus
before
from-internal
is right?
regards
El 02/12/2015 a las 10:22, Sonny Rajagopalan escribió:
Yes, I have tried that too (i.e, exten => +17775551212,1,Log(WARNING,
TWILIO)). It does not work and NO error message in CLI.
Yes, I have tried that too (i.e, exten => +17775551212,1,Log(WARNING,
TWILIO)). It does not work and NO error message in CLI.
I have also tried
http://orourketech.com/elastix-plus-sign-caller-id-messing-things/ since I
first emailed this group, but that does not seem to work either.
Here is my lo
Hello,
try to change:
exten => 17775551212,1,Log(WARNING, TWILIO)
same => n,Hangup()
with:
exten => +17775551212,1,Log(WARNING, TWILIO)
same => n,Hangup()
Regards
--
_
-- Bandwidth and Colocation Provided by http://www.api
Hello,
I am running Asterisk 13.6.0 in an AWS instance, and I set it up with
Twilio SIP trunk using pjsip_wizard.conf (nice feature!). I see that the
calls actually "reach" the PBX, but for some reason, they are not caught by
any of my extensions context.
Here's what I observe when I test this fr
n-Commercial Discussion
mailto:asterisk-users@lists.digium.com>>
Date: Tuesday, October 20, 2015 at 2:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
mailto:asterisk-users@lists.digium.com>>
Subject: [asterisk-users] Issues with
I am trying to get my Linksys/Cis
I am trying to get my Linksys/Cisco SPA3102 to connect to asterisk 13 PJSIP. It
is registered just fine but when I dial one of my known extensions on the
server. As far as I can tell it should be able to translate as also pasted
below.
Can anyone help me?
res_pjsip_sdp_rtp.c:324 set_caps:
Nick Awesome wrote:
May someone help with the sourcecode, trying find where can I
manually send response on Received INFO request in PJSIP
ASTERISK-24986 issues opened already more the 2 month and calls from
customers still drops. very annoying :( maybe some one could help me
figure out where Re
May someone help with the sourcecode, trying find where can I manually send
response on Received INFO request in PJSIP
ASTERISK-24986 issues opened already more the 2 month and calls from customers
still drops. very annoying :( maybe some one could help me figure out where
Received INFO request
Nick Awesome wrote:
Hi guys, have really annoying problem with trunks when I calling over voip
provider..
after awhile provider sends INFO packages but for some reason Asterisk doesn’t
answer on it.
after 8 packagers provider just drops the call, here is the package
<--- Received SIP request
Hi guys, have really annoying problem with trunks when I calling over voip
provider..
after awhile provider sends INFO packages but for some reason Asterisk doesn’t
answer on it.
after 8 packagers provider just drops the call, here is the package
<--- Received SIP request (555 bytes) from UDP:
2011/12/10 giovanni.v
> Il 10/12/2011 8.03, Olivier ha scritto:
>
> Yes, assuming no hardware/configuration problems this shouldn't happen
>>> > on PTP.
>>>
>>
> Can you explain why taking down layer 1 on idling spans happens on
>> PtmP and can't happen on PTP ?
>>
>
> Where possible worse tha
Il 10/12/2011 8.03, Olivier ha scritto:
Yes, assuming no hardware/configuration problems this shouldn't happen
> on PTP.
Can you explain why taking down layer 1 on idling spans happens on
PtmP and can't happen on PTP ?
Where possible worse than my English: "ne devrait pas arriver",
"généra
2011/12/9, giovanni.v :
> Il 09/12/2011 17.11, Olivier ha scritto:
2. On a more general plan, is taking down layer 1 on idling spans
>> something PBXs are negociating with each other (the public switch
>> trying to take the layer one down, listening to an acknowledge
>> from
Il 09/12/2011 17.11, Olivier ha scritto:
2. On a more general plan, is taking down layer 1 on idling spans
>> something PBXs are negociating with each other (the public switch
>> trying to take the layer one down, listening to an acknowledge
>> from the private PBX) or is it more brutal than t
On Fri, Dec 09, 2011 at 05:11:47PM +0100, Olivier wrote:
> 2011/12/8, Shaun Ruffell :
> > On Thu, Dec 08, 2011 at 06:41:46PM +0100, Olivier wrote:
> >> 2. On a more general plan, is taking down layer 1 on idling spans
> >> something PBXs are negociating with each other (the public switch
> >> tryi
2011/12/8, Shaun Ruffell :
> On Thu, Dec 08, 2011 at 06:41:46PM +0100, Olivier wrote:
>> 2011/12/8, Shaun Ruffell :
>> > On Thu, Dec 08, 2011 at 09:26:14AM +0100, Olivier wrote:
>> >
>> >> 2. Is it normal to see this IRQ changing from time to time ?
>> >
>> > Normally, after things have stabilized,
On Thu, Dec 08, 2011 at 06:41:46PM +0100, Olivier wrote:
> 2011/12/8, Shaun Ruffell :
> > On Thu, Dec 08, 2011 at 09:26:14AM +0100, Olivier wrote:
> >
> >> 2. Is it normal to see this IRQ changing from time to time ?
> >
> > Normally, after things have stabilized, it should remain
> > constant on a
2011/12/8, Shaun Ruffell :
> On Thu, Dec 08, 2011 at 09:26:14AM +0100, Olivier wrote:
>> Hi,
>>
>> On a (bogus) system equiped with a Digium HA8 and 8 BRI ports, I'm reading
>> this:
>> # asterisk -rx "dahdi show status"
>> Description Alarms IRQbpviol CRC4 Fra Codi Options LBO
>> HA8-
On Thu, Dec 08, 2011 at 09:26:14AM +0100, Olivier wrote:
> Hi,
>
> On a (bogus) system equiped with a Digium HA8 and 8 BRI ports, I'm reading
> this:
> # asterisk -rx "dahdi show status"
> Description Alarms IRQbpviol CRC4 Fra Codi Options LBO
> HA8- RED 1090 0
Hi,
On a (bogus) system equiped with a Digium HA8 and 8 BRI ports, I'm reading this:
# asterisk -rx "dahdi show status"
Description Alarms IRQbpviol CRC4
Fra Codi Options LBO
HA8- RED 1090 0
CCS AMI YEL 0 db
Tzafrir,
I am in front of the server.
Le dimanche 30 octobre 2011 à 22:13 +0100, Eric van der Vlist a écrit :
> Tzafrir,
>
> Le dimanche 30 octobre 2011 à 10:30 +0200, Tzafrir Cohen a écrit :
> > The problem is elsewhere. What happens if
> > you manually run:
> >
> > /usr/share/dahdi/xpp_fxl
Tzafrir,
Le dimanche 30 octobre 2011 à 10:30 +0200, Tzafrir Cohen a écrit :
> On Sat, Oct 29, 2011 at 08:14:55PM +0200, Eric van der Vlist wrote:
> > Hi,
> >
> > Xorcom astribanks get initialized straight on when using Ubuntu 11.10
> > packages but I am having a hard time to get the same result r
On Sat, Oct 29, 2011 at 08:14:55PM +0200, Eric van der Vlist wrote:
> Hi,
>
> Xorcom astribanks get initialized straight on when using Ubuntu 11.10
> packages but I am having a hard time to get the same result running in a
> qemu/libvirt image.
qemu? qemu+kqemu (the kernel module)? kvm? I would e
On Sat, Oct 29, 2011 at 3:14 PM, Eric van der Vlist wrote:
> Hi,
>
> Xorcom astribanks get initialized straight on when using Ubuntu 11.10
> packages but I am having a hard time to get the same result running in a
> qemu/libvirt image.
>
> The first difficulty is that astribanks devices get differ
Hi,
Xorcom astribanks get initialized straight on when using Ubuntu 11.10
packages but I am having a hard time to get the same result running in a
qemu/libvirt image.
The first difficulty is that astribanks devices get different usb device
ids during their initialisation process, requiring hot pl
I have used those packages:
[Apr 7 01:09:51] WARNING[27966]: loader.c:434 load_dynamic_module:
Error loading module 'app_voicemail_imapstorage.so':
/usr/lib/asterisk/modules/app_voicemail_imapstorage.so: undefined
symbol: copy
[Apr 7 01:09:51] WARNING[27966]: loader.c:777 l
On 06/06/2011 08:07 PM, Andrew Joakimsen wrote:
Anyone have an update as to when Digium will ship a working package?
According to https://issues.asterisk.org/view.php?id=18748 new packages
should already have been pushed. If not perhaps you could join #asterisk
or #asterisk-dev on irc.freenod
Anyone have an update as to when Digium will ship a working package?
-- Forwarded message --
From: Andrew Joakimsen
Date: Wed, Mar 23, 2011 at 23:53
Subject: Issues with Digum Repos / AsteriskNOW & Bad Packages
To: Asterisk Users Mailing List - Non-Commercial Discussion
I wish
On Thu, Mar 24, 2011 at 15:34, Kevin P. Fleming wrote:
> On 03/23/2011 10:53 PM, Andrew Joakimsen wrote:
>>
>> I wish to use AsteriskNOW (the Digium repository + CentOS) with imap
>> voicemail storage and Asterisk 1.4.
>>
>> After having installed AsteriskNOW with Asterisk 1.4 and Asterisk GUI
>>
On Thu, Mar 24, 2011 at 15:34, Kevin P. Fleming wrote:
> On 03/23/2011 10:53 PM, Andrew Joakimsen wrote:
>>
>> I wish to use AsteriskNOW (the Digium repository + CentOS) with imap
>> voicemail storage and Asterisk 1.4.
>>
>> After having installed AsteriskNOW with Asterisk 1.4 and Asterisk GUI
>>
On 03/23/2011 10:53 PM, Andrew Joakimsen wrote:
I wish to use AsteriskNOW (the Digium repository + CentOS) with imap
voicemail storage and Asterisk 1.4.
After having installed AsteriskNOW with Asterisk 1.4 and Asterisk GUI
I run the yum package manager and replace voicemail with imap
voicemail a
011 11:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Issues with Digum Repos / AsteriskNOW & Bad Packages
I wish to use AsteriskNOW (the Digium repository + CentOS) with imap
voicemail storage and Asterisk 1.4.
After having installed AsteriskNOW wi
I wish to use AsteriskNOW (the Digium repository + CentOS) with imap
voicemail storage and Asterisk 1.4.
After having installed AsteriskNOW with Asterisk 1.4 and Asterisk GUI
I run the yum package manager and replace voicemail with imap
voicemail and attempt to start Asterisk, however the voicemai
On 10-12-02 12:22 PM, Bryant Zimmerman wrote:
> Karsten
> I do not see it in the changlog for the 1.8.1 rc version.
> How would I get the SVN version to test?
>
$ svn co http://svn.asterisk.org/svn/asterisk/branches/1.8
--
Paul Belanger
Digium, Inc. | Software Developer
twitter: pabelanger | IRC:
>> On Wed, Dec 1, 2010 at 9:44 AM, Bryant Zimmerman
>> wrote:
>> >> I am having issues with Blind Transfer on asterisk 1.8
>>
>> >What specific version: 1.8.0, 1.8.1-rc1, SVN branch? What OS?
>>
>> Verison 1.8.0, Suse 11.1
Try the latest SVN branch for 1.8 and see if that resolves your issue:
$
ubject: Re: [asterisk-users] Issues with 1.8 and BlindTransfer
Hi,
Am Donnerstag, den 02.12.2010, 11:02 -0500 schrieb Bryant Zimmerman:
> Replys from Bryant
>
> On Wed, Dec 1, 2010 at 9:44 AM, Bryant Zimmerman
> wrote:
> >> I am having issues with Blind Transfer on asterisk
Hi,
Am Donnerstag, den 02.12.2010, 11:02 -0500 schrieb Bryant Zimmerman:
> Replys from Bryant
>
> On Wed, Dec 1, 2010 at 9:44 AM, Bryant Zimmerman
> wrote:
> >> I am having issues with Blind Transfer on asterisk 1.8
>
> >What specific version: 1.8.0, 1.8.1-rc1, SVN branch? What OS?
>
> Verison
Replys from Bryant
On Wed, Dec 1, 2010 at 9:44 AM, Bryant Zimmerman
wrote:
>> I am having issues with Blind Transfer on asterisk 1.8
>What specific version: 1.8.0, 1.8.1-rc1, SVN branch? What OS?
Verison 1.8.0, Suse 11.1
>> If I call from one Grandstream phone to another and us the transf
On Wed, Dec 1, 2010 at 9:44 AM, Bryant Zimmerman wrote:
> I am having issues with Blind Transfer on asterisk 1.8
What specific version: 1.8.0, 1.8.1-rc1, SVN branch? What OS?
> If I call from one Grandstream phone to another and us the transfer key
> to do a blind transfer everything works
I am having issues with Blind Transfer on asterisk 1.8
If I call from one Grandstream phone to another and us the transfer key
to do a blind transfer everything works fine.
When calling in on a sip trunk and then trying to use the transfer key
to transfer from Grandstream phone to G
--
> *De: *"Mark Murawski"
> *Para: *asterisk-us...@lists.digium.com
> *Enviadas: *Terça-feira, 16 de Novembro de 2010 15:15:16
> *Assunto: *Re: [asterisk-users] Issues with Local Channel
>
> Local channels behave like an endpoint. So instead of a sip phone
>
sagem original -
De: "Mark Murawski"
Para: asterisk-users@lists.digium.com
Enviadas: Terça-feira, 16 de Novembro de 2010 15:15:16
Assunto: Re: [asterisk-users] Issues with Local Channel
Local channels behave like an endpoint. So instead of a sip phone
picking up the call, asteris
Local channels behave like an endpoint. So instead of a sip phone
picking up the call, asterisk is picking up the call.
Instead of someone speaking into a sip phone, asterisk can play tracks,
or record digits, etc.
You need to make sure that the call does not end before you're done with
your
Hello,
I don't really understand how channel Local works. I need that asterisk
initiate a call and get some data (DTMF).
So to do that I've created this dialplan :
; extensions.conf - the Asterisk dial plan
;
[general]
static=yes
writeprotect=no
clearglobalvars=no
[dtmf]
exten =
The issue we are having is that in-call RFC2833 DTMF digits are being
dropped with Broadvox and Level 3. This is happening with Grandstream GXP
and Snom phones. We did some testing with the vendors and here is one of
the responses we got back. Is there any way to force asterisk to modify the
DT
> Maybe your engine is "tone deaf". Try showing the ${SPEECH_SCORE(0)} when
> you get the foobared result.
I repeated the experiment, this time noting the score, which I output.
This time, the result was always "2" and the score was pretty
high: 711, 743, 752.
--
___
1:36 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Issues with Vestec ASR
> Make sure that you only have the "one" grammar active when doing your
test.
> You want the voice engine to basically only have 11 possibilities to chew
on
> (0-9 plus "oh").
> Make sure that you only have the "one" grammar active when doing your test.
> You want the voice engine to basically only have 11 possibilities to chew on
> (0-9 plus "oh").
I always only load one grammar. In the test I did below, there were
exactly TWO possibilities:
> I'm having a lot of pro
lists.digium.com] On Behalf Of Richard Kenner
Sent: Monday, June 07, 2010 10:45 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Issues with Vestec ASR
I'm having a lot of problem with it recognizing "oh" for zero.
I've tried both "o" and "oh"
I'm having a lot of problem with it recognizing "oh" for zero.
I've tried both "o" and "oh". In one case, I just tried:
$digit = o { out = "0"; } | fundamental {out = "2"; };
So I gave it a choice that was VERY far away from what I said.
But when I said "o o o o o", more than 75% of th
Hello list,
I would like to seek your expert opinion on a setup I am trying as part of my
research. I have not been able to successfully make a call so far.
In my setup, I use two laptops that are interconnected by means of a
stand-alone IS1581 switch. Thus there is no LAN involved.
I have assi
Hi all,
We've been having a very frustrating time with our Asterisk install (well,
okay, actually Switchvox). We have an open ticket with Digium/Switchvox but I
was wondering if anyone here might have some helpful tips.
We're basically getting glitching on the line, and in the error logs it's
- Original Message -
From: "Michiel van Baak" <[EMAIL PROTECTED]>
To:
Sent: Friday, November 02, 2007 11:47 AM
Subject: Re: [asterisk-users] issues with downloads.digium.com
> On 09:38, Fri 02 Nov 07, Tony Mountifield wrote:
>> Does anyone from Digium wa
Am Freitag, den 02.11.2007, 12:12 +0200 schrieb Atis Lezdins:
> On 11/2/07, Michiel van Baak <[EMAIL PROTECTED]> wrote:
> > On 09:38, Fri 02 Nov 07, Tony Mountifield wrote:
> > > Does anyone from Digium want to comment on why this Eloqua stuff has been
> > > used, instead of just allowing Apache to
On Fri, Nov 02, 2007 at 12:12:20PM +0200, Atis Lezdins wrote:
> I wonder - why they can't get all the info from logs. They can even
> put .htaccess to route all downloads trough PHP that will log whatever
> else it needs..
Any solution of that kind would be better than the current arrangement.
>
On 11/2/07, Michiel van Baak <[EMAIL PROTECTED]> wrote:
> On 09:38, Fri 02 Nov 07, Tony Mountifield wrote:
> > Does anyone from Digium want to comment on why this Eloqua stuff has been
> > used, instead of just allowing Apache to serve the directory tree directly?
> > And whether this decision migh
On 09:38, Fri 02 Nov 07, Tony Mountifield wrote:
> Does anyone from Digium want to comment on why this Eloqua stuff has been
> used, instead of just allowing Apache to serve the directory tree directly?
> And whether this decision might be reconsidered?
>
I think it's some sort of tool to track d
In article <[EMAIL PROTECTED]>,
Tony Mountifield <[EMAIL PROTECTED]> wrote:
> In article <[EMAIL PROTECTED]>,
> Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
> > As you can see, that script doesn't really redirect. It does not point
> > me to the new file name. If I use a web browser, I still get the
>
Carlos Chavez wrote:
> On Wed, 2007-10-31 at 15:26 +0200, Tzafrir Cohen wrote:
>> On a slightly different matter:
>> http://asterisk.org/downloads still points to zaptel 1.4.5.1 and libpri
>> 1.4.1 .
>>
>
> Yes, I noticed that too and was wondering if it is just because they
> have not upda
On Wed, 2007-10-31 at 15:26 +0200, Tzafrir Cohen wrote:
> On a slightly different matter:
> http://asterisk.org/downloads still points to zaptel 1.4.5.1 and libpri
> 1.4.1 .
>
Yes, I noticed that too and was wondering if it is just because they
have not updated the site or if there is a
On a slightly different matter:
http://asterisk.org/downloads still points to zaptel 1.4.5.1 and libpri
1.4.1 .
--
Tzafrir Cohen
icq#16849755 jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
http://www.xorcom.com iax:[EMAIL PR
On Mon, Oct 29, 2007 at 09:02:14AM -0400, Dave Fullerton wrote:
> Tzafrir Cohen wrote:
> > Hi
> >
> > Sorry to use this public place, but IRC and emails to [EMAIL PROTECTED]
> > have not helped in the past.
> >
> > I have several issues with using the files server downloads.digium.com,
> > which
Dave Fullerton wrote:
> Tzafrir Cohen wrote:
>> Let's look at http://downloads.digium.com/pub/telephony/
>>
>> I get a list of items. I have to guess which of them is a file and which
>> is a directory. There is no proper date of change.
> Not sure I completely understand what you mean by "I hav
Tzafrir Cohen wrote:
> Furthermore, I cannot follow links directly. Links are redirections.
>
> For instance, the link marked with "aadk" points to:
>
>
> http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.digium.com/pub/telephony/aadk
>
> $ HEAD
> 'http://www.digium.com/elqNow/
In article <[EMAIL PROTECTED]>,
Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
>
> Sorry to use this public place, but IRC and emails to [EMAIL PROTECTED]
> have not helped in the past.
>
> I have several issues with using the files server downloads.digium.com,
> which has replaced the simple ftp/http
Tzafrir Cohen wrote:
> Hi
>
> Sorry to use this public place, but IRC and emails to [EMAIL PROTECTED]
> have not helped in the past.
>
> I have several issues with using the files server downloads.digium.com,
> which has replaced the simple ftp/http file server ftp.digium.com.
>
> In downloads.d
Hi
Sorry to use this public place, but IRC and emails to [EMAIL PROTECTED]
have not helped in the past.
I have several issues with using the files server downloads.digium.com,
which has replaced the simple ftp/http file server ftp.digium.com.
In downloads.d.c the directory listing is served thro
[EMAIL PROTECTED] wrote:
> ZAP/G1 will dial starting from channel 1. ZAP/R1 will dial starting
> from the last channel of the group.
>
Actually, ZAP/g1 mean start with first channel and work up. ZAP/G1 mean
start with last channel and work down. 'r' and 'R' operate in similar
directions, bu
Hi Tzafrir!
Thank you for your answer, and my apologies for my delayed
response. I regret to say that the patch test's results were not succesful.
I shall describe the whole procedure in detail for you to establish whether
I did something wrong.
The fact is that I am as much a Linux beginner a
ZAP/G1 will dial starting from channel 1. ZAP/R1 will dial starting
from the last channel of the group.
Pablo, please tell us what version of Linux and which distribution are
you using. Maybe for the time being try the stock asterisk of your
distro or the one they provide in the buildservice?
On
Pablo - You said you have 1/2 E1 - which half???
That might be your problem. Unless 1/2 E1 means something else...
Asterisk normally dials out on the low end unless you specify
G instead of g ??? or something like that.
Brett
___
--Bandwidth and Colo
I run this command
[EMAIL PROTECTED] ~]# cat /proc/zaptel/1
Span 1: WCT1/0 "Wildcard TE12xP Card 0"
IRQ misses: 40
1 WCT1/0/1
2 WCT1/0/2
3 WCT1/0/3
4 WCT1/0/4
5 WCT1/0/5
6 WCT1/0/6
7 WCT1/0/7
8 WCT1/0/
On Thu, Oct 18, 2007 at 10:53:15AM -0500, Pablo Almido wrote:
> I have unload and load the module, it is output
>
>
> ippbx*CLI> module unload chan_zap.so
> == Unregistered application 'ZapSendKeypadFacility'
>
> ippbx*CLI> module load chan_zap.so
> == Registered application 'ZapSendKeypadFa
I have unload and load the module, it is output
ippbx*CLI> module unload chan_zap.so
== Unregistered application 'ZapSendKeypadFacility'
ippbx*CLI> module load chan_zap.so
== Registered application 'ZapSendKeypadFacility'
== Parsing '/etc/asterisk/zapata.conf': Found
[Oct 18 10:46:38] WARN
Yes, the module is load
# asterisk -r
ippbx*CLI> module show like chan_zap.so
Module Description
Use Count
chan_zap.soZapata Telephony
0
1 modules loaded
ippbx*CLI>
ippbx*CLI>
2007/10/18, Brian West <[EMAIL PROTECTED]>:
> Make sure chan_zap.so is
Why would a config error stop the module from loading? That seems
like a suboptimal behavior.
/b
On Oct 18, 2007, at 9:50 AM, Jared Smith wrote:
That would seem to indicate that the chan_zap.so module isn't being
loaded. What happens if you type "module unload chan_zap.so" and then
"module
On Thu, Oct 18, 2007 at 08:35:01AM +0100, Alan Lord wrote:
> Tzafrir Cohen wrote:
> ... is required to properly start zaptel. It will also run
> ztcfg. Otherwise
> > users run into issues where misconfigured zaptel.conf fails loading of a
> > module. That is a buggy behaviour.
> >
> > If your car
On Thu, 2007-10-18 at 09:34 -0500, Pablo Almido wrote:
> [Oct 18 09:13:50] WARNING[2377]: channel.c:3232 ast_request: No
> channel type registered for 'Zap'
> [Oct 18 09:13:50] WARNING[2377]: app_dial.c:1106 dial_exec_full:
> Unable to create channel of type 'Zap' (cause 66 - Channel not
> implemen
Make sure chan_zap.so is loaded.
/b
On Oct 18, 2007, at 9:34 AM, Pablo Almido wrote:
> Hi List,
>
> I am from Peru, I have installed an asterisk server in my company with
> digium card E1 TE120P, I am having issues when i make calls, here the
> error from my server
>
>
> [Oct 18 09:13:50] WARNIN
Hi List,
I am from Peru, I have installed an asterisk server in my company with
digium card E1 TE120P, I am having issues when i make calls, here the
error from my server
[Oct 18 09:13:50] WARNING[2377]: channel.c:3232 ast_request: No
channel type registered for 'Zap'
[Oct 18 09:13:50] WARNING[2
Tzafrir Cohen wrote:
... is required to properly start zaptel. It will also run
ztcfg. Otherwise
> users run into issues where misconfigured zaptel.conf fails loading of a
> module. That is a buggy behaviour.
>
> If your card is an analog one, take a look at http://bugs.digium.com/7613
> and tell
On Wed, Oct 17, 2007 at 06:37:21PM -0300, Aldo D. Sudak wrote:
> Greetings to all list members!
>
> My name is Aldo Sudak and I am an Asterisk newbie. I am writing now because I
> have not
> been able to find any mention to the issues described below, neither in this
> list nor in the wiki.
>
Greetings to all list members!
My name is Aldo Sudak and I am an Asterisk newbie. I am writing now because I
have not
been able to find any mention to the issues described below, neither in this
list nor in the wiki.
I am performing preliminary tests with Asterisk 1.4.11 and Zaptel 1.4.5.1 on
I am currently struggling to implement this feature in realtime asterisk. I
have configured realtime asterisk and it works great, however I can not get
regserver to update via the rtsavesysname in sip.conf. Heres my configs..
sip.conf
[EMAIL PROTECTED] asterisk]# more sip.conf
[general]
displays
At the SIP menu,
RTP Packet Size: 0.030
Erick Perez wrote:
where to change packet size?
On 3/9/07, Luki <[EMAIL PROTECTED]> wrote:
> Any gurus out there with experience with the SPA 2102 against
asterisk 1.2.14?
They work fine with Asterisk; most likely it's your wireless link
that's the
Erick Perez wrote:
where to change packet size?
Admin Login -> Advanced
Voice->SIP Tab
RTP Packet Size: .02
On 3/9/07, Luki <[EMAIL PROTECTED]> wrote:
> Any gurus out there with experience with the SPA 2102 against
asterisk 1.2.14?
They work fine with Asterisk; most likely it's your w
Via the web GUI of the phone. I don't remember exactly which screen it
is on. It defaults to .30, you need to change it to .20 to fix audio
problems with that device.
Erick Perez wrote:
where to change packet size?
On 3/9/07, Luki <[EMAIL PROTECTED]> wrote:
> Any gurus out there with exper
where to change packet size?
On 3/9/07, Luki <[EMAIL PROTECTED]> wrote:
> Any gurus out there with experience with the SPA 2102 against asterisk 1.2.14?
They work fine with Asterisk; most likely it's your wireless link
that's the cause of your problem. The jitter buffer will only affect
receiv
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