I have a Problem with incoming ISDN calls in Austria, I use zaptel and asterisk
bristuffed from Debian/Etch.
- If someone outside is dialing the phonenumber and the extension on an ISDN
phone, asterisk catches the call and puts into in the s extension before the
extension was submittet.
- The
I am using asterisk 1.4.18. I cant at this stage upgrade to any latest
version. Linear strategy for queues is not in asterisk 1.4.18. I have to
use ringall instead.
Is it possible Disabling call-waiting for my agents only?? While other
sip users have call waiting functionality.
regards
Sarò in vacanza fino a martedì 19 agosto con scarsa possibilità di accedere a
e-mail e telefono. Per richieste urgenti, vi prego di contattare Wildix srl al
numero di telefono 0461 74 30 891 o all'indirizzo e-mail [EMAIL PROTECTED],
altrimenti vi risponderò al mio rientro.
Dimitri Osler
I
A week ago, I tried give realtime priority to asterisk proces using -p
switch,
asterisk was running inside astcanary,
but yestarday asterisk probably starts eating all cpu and lock any
access to computer, only ping was possible,
so, anybody have experience, that ascanary process does really work
--- On Tue, 8/5/08, Atis Lezdins [EMAIL PROTECTED] wrote:
Have you tried powering it on, while holding reset button?
Yes, several times. I contacted Grandstream's helpdesk and they told me to keep
the reset button pressed while I plug the power off and back on again. I even
tried keeping it
hi all,
i just finished developing some incoming call features in a macro. that
macro gets executed everytime an incoming call is received and a new mysql
connection is made using the MYSQL cmd in dialplan. i want to use a single
mysql connection for every incoming call.
my idea of doing it is
Good question, I'll check.
Regards,
Steve
2008/8/6 Tom Lynn [EMAIL PROTECTED]:
Steve, what kind of Avaya system is this? They make several.
On Tue, Aug 5, 2008 at 11:36 AM, Steve Davies [EMAIL PROTECTED] wrote:
Hi,
Sorry this is so long, but I am reasonably desparate.
I am having real
HI
This is my setup of the features.conf but it had not any reaction after I
pushed the *2 while calling was acting ! Could you tell me the reason ? Or
give my the method of the setting.
Thanks!
LARRY
[general]
parkext = 700
parkpos =
Hi,
My apologies for the OT. My googling came up empty and hopefully there
are some members in the community that could give me a hint how to solve
this issue:
Cisco 7961 with SIP firmware 8.3.3. Needed to downgrade it to 8.0.4SRS2.
The downgrade process started off good. The 7961 got it's IP
Hi All,
Would just like to know if anyone has encountered this:
i a user is currently registered using SPA 941, i then tried deleting
the user in the realtime db. then i tried to make a call from the SPA i
can still make calls even though user has been deleted.
i tried the same thing this
I have a server with Asterisk 1.4.21.1 and some prompts recorded in GSM
format. I have these same prompts in another server with Asterisk 1.4.18, on
this server the prompts sound pretty nice, but on the first one they sound
pretty choppy. Was there any changes on the transcoding code between this
I am told it is an IP Office 400 series.
I have not been on site physically which does not help.
Regards,
Steve
2008/8/6 Tom Lynn [EMAIL PROTECTED]:
Steve, what kind of Avaya system is this? They make several.
On Tue, Aug 5, 2008 at 11:36 AM, Steve Davies [EMAIL PROTECTED] wrote:
Hi,
Hello,
I would like to configure hylafax(4.4.4) + iaxmodem(1.1.1). I use Asterisk
and I work on Redhat.
I installed the two hylafax and iaxmodem.
My configuration of iaxmodem is: (in the file /etc/iaxmodem/ttyIAX0)
device /dev/ttyIAX0
owner uucp:uucp
mode 660
port 4570 #each line should have
I would make absolutely sure you've got your linux distro's version of
libgsm installed. I can't really speak to the difference between those
two versions of Asterisk without looking at a change-log, but I highly
doubt a serious modification to the gsm code took place between sub-
versions.
On Wed, Aug 6, 2008 at 4:05 PM, [EMAIL PROTECTED] wrote:
Hello,
I would like to configure hylafax(4.4.4) + iaxmodem(1.1.1). I use Asterisk
and I work on Redhat.
I installed the two hylafax and iaxmodem.
My configuration of iaxmodem is: (in the file /etc/iaxmodem/ttyIAX0)
device
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Patrick
Sent: Wednesday, August 06, 2008 7:34 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] OT: Cisco 7961 SIP downgrade from 8.3.3 -
8.0.4SRS2 failing
Hi,
My apologies for the OT. My
I'm using OpenSUSE 10.3, the funny thing is: if the softphone is using GSM
the sounds is perfect, if I use Alaw as the softphone CODEC the sounds is
pretty bad. The softphone is in the same LAN as the Asterisk server, so I
don't think it's a bandwidth issue.
Best Regards,
On Wed, Aug 6, 2008 at
I am a **BIG, BIG** fan of OpenSUSE.
:)
Use yast under 'Software Management' and do a search for 'gsm'.
Make sure gsmlib and gsmlib-devel are *both* installed. Then scroll
down and make sure that libgsm and libgsm-devel are *both* installed.
After that, you'll have to recompile Asterisk.
On Tue, Aug 05, 2008 at 02:13:15PM -0500, Tilghman Lesher wrote:
above the original post is very confusing. Please stop doing this.
The format of this post is in reverse, to demonstrate why posting a reply
option is only in trunk. So no, it would not help him out.
Yes, it works the same
Guilherme Loch Waltrick Góes wrote:
I have a server with Asterisk 1.4.21.1 http://1.4.21.1 and some
prompts recorded in GSM format. I have these same prompts in another
server with Asterisk 1.4.18, on this server the prompts sound pretty
nice, but on the first one they sound pretty choppy.
On Wednesday 06 August 2008 08:13:09 Darren Sessions wrote:
I would make absolutely sure you've got your linux distro's version of
libgsm installed. I can't really speak to the difference between those
two versions of Asterisk without looking at a change-log, but I highly
doubt a serious
My iax.conf looks like this:
[iaxmodem]
type=friend
host=127.0.0.1
secret=x
context=fax-out
permit=127.0.0.1
disallow=all
allow=ulaw
after editing inittab I reload it by running: /sbin/init q
I also reboot the system with shutdown -r now and I had the following
message:
init: Id mo
Hi,
is there meanwhile the possibility for some actions besides dialling in *?
Namely, I would like that if a remote IAX or SIP user logs in AND there
are new messages, they automatically get a call and be connected to the
voicemail. The only method I know by now is make a context in the
I have used virtually all versions of Asterisk 1.0+ (literally, either
in production or testing) with OpenSUSE 10+ and 11 on AMD and Intel
and haven't had any issues with gcc optimizations with regards to
audio sounding choppy. This scenario for me has always been the gsm
libs.
Hi,
Am Mittwoch, den 06.08.2008, 17:24 +0200 schrieb Nadjia Boumédiène:
My iax.conf looks like this:
[iaxmodem]
type=friend
host=127.0.0.1
secret=x
context=fax-out
permit=127.0.0.1
disallow=all
allow=ulaw
after editing inittab I reload it by running: /sbin/init q
I also
Sarò in vacanza fino a martedì 19 agosto con scarsa possibilità di accedere a
e-mail e telefono. Per richieste urgenti, vi prego di contattare Wildix srl al
numero di telefono 0461 74 30 891 o all'indirizzo e-mail [EMAIL PROTECTED],
altrimenti vi risponderò al mio rientro.
Dimitri Osler
I
Hi Matt,
Thank you for your suggestion. Comment inline.
Matt Gibson wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Patrick
Sent: Wednesday, August 06, 2008 7:34 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] OT: Cisco 7961
I suppose, too. So see below. I also verified that the dial command is
using Ww (which I had to fudge), but still, no monitoring.
Anything else I can check?
pig*CLI feature show
Builtin Feature Default Current
--- --- ---
Pickup *8 *8
Blind Transfer # #
Attended Transfer
One
2008/8/5 Steve Davies [EMAIL PROTECTED]:
Hi,
Sorry this is so long, but I am reasonably desparate.
I am having real fun with hooking an Avaya system to Asterisk using
ISDN30. I have the ISDN signalling all sorted one way, and can pass
calls from the real world (ie. the telco and asterisk)
Hello All,
I'am doing a video call between two Video Phones, and i see
that Asterisk is stripping the fmtp parameters for the h263 video line in
SDP.
For example a line similar to the below is stripped,
a=fmtp:xx CIF=4;QCIF=2;F=1;K=1
Asterisk is configured NOT to
Sarò in vacanza fino a martedì 19 agosto con scarsa possibilità di accedere a
e-mail e telefono. Per richieste urgenti, vi prego di contattare Wildix srl al
numero di telefono 0461 74 30 891 o all'indirizzo e-mail [EMAIL PROTECTED],
altrimenti vi risponderò al mio rientro.
Dimitri Osler
I
On Wed, Aug 06, 2008 at 10:15:00AM -0500, Tilghman Lesher wrote:
On Wednesday 06 August 2008 08:13:09 Darren Sessions wrote:
I would make absolutely sure you've got your linux distro's version of
libgsm installed. I can't really speak to the difference between those
two versions of Asterisk
hi,
wanted to ask if anybody has experienced setting up two asterisk 1.2 boxes
connected via iax trunk. have u guys ever stress tested the trunks i.e how
many concurrent calls can a trunk handle and whether codec has any effect on
it.
___
-- Bandwidth
Hi all,
Our users are complaining about beeps that happen in the middle of some
calls. They are similar to the sound heard you are in a call and press
any button in your phone. Please find bellow some examples of these
beeps(the recordings are in Portuguese, but the beeps are easy to
identify):
maybe you are using the L option in Dial app to limit the conversation time.
Check those channel variables (just a wild guess)
*LIMIT_PLAYAUDIO_CALLER
**LIMIT_PLAYAUDIO_CALLEE
**LIMIT_TIMEOUT_FILE
**LIMIT_CONNECT_FILE
**LIMIT_WARNING_FILE
*
Felippe Silvestre wrote:
Hi all,
Our users are
Rosli Sukri wrote:
hi,
wanted to ask if anybody has experienced setting up two asterisk 1.2
boxes connected via iax trunk. have u guys ever stress tested the
trunks i.e how many concurrent calls can a trunk handle and whether
codec has any effect on it.
Hi All,
Would just like to know if anyone has encountered this:
i a user is currently registered using SPA 941, i then tried deleting
the user in the realtime db. then i tried to make a call from the SPA i
can still make calls even though user has been deleted.
i tried the same thing this time
Hi all,
my goal is to connect my trixbox server (CentOS 5.2 - kernel
2.6.18-53.1.4.el5 - * 1.4.20-1) with a legacy Philips PBX with 4 bri
links provided from Digium B410P.
For this reason I set all the 4 ports of Digium's card in NT mode
(Philips can not do this). Then i opportunely
edited
On Wednesday 06 August 2008 15:07:03 Nhadie wrote:
Would just like to know if anyone has encountered this:
You sent the exact same email this morning at 7:47 a.m. If nobody has
responded, it's because nobody has ever seen that before. Duplicating
the message tells everybody on the list that
Guys I have been reading for days on how to get this to work with asterisk
and for some reason every time I call the call goes to intercom. I know I
must be doing something wrong with the way I am adding the steps to my call;
I am not familiar with variables and flags.
Here is my
I am sorry, this is the actual extensions.conf:
exten=s,1,SIPAddHeader(Alert-Info: http://127.0.0.1\;info=Family)
exten=s,2,GotoIf($[${SIP_HEADER(Call-Info)}=answer-after=0]?2:3)
exten=s,3,SIPAddHeader(Call-Info: answer-after=0)
exten=s,4,Dial(${ARG2},20)
exten=s,5,Goto(s-${DIALSTATUS},1)
Felippe Silvestre wrote:
Hi all,
Our users are complaining about beeps that happen in the middle of some
calls. They are similar to the sound heard you are in a call and press
any button in your phone. Please find bellow some examples of these
beeps(the recordings are in Portuguese, but
I've also seen systems where the IRQ between the card and another heavily
loaded device (disk controller) are shared causing clicks, beeps, and pops to
be present in the audio stream.
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
- Alan Lord [EMAIL PROTECTED] wrote:
I have two Asterisk 1.4 boxes connected via IAX over a VPN tunnel on a
10Mbit link. We never did any stress testing as it's a temporary
arrangement, but we've never had any call quality issues or run up
against concurrent call limitations. I'm mostly routing internal
extensions over the
Hi friends,
Where can I get some information to understand messages like the following
ones?
*NOTICE[6455] chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary
D-channel of span 1*
*NOTICE[6455] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel
of span 1*
* ERROR[6455] chan_zap.c:
We've searched but thus far have not successfully found a solution for this…
We're looking for a way to set a variable using get digits for a DISA
application. Sometimes we're away from the office and get a voicemail that
I need to respond to quickly and would prefer for the caller to be
i apologize, coz i had some experiences that my mail did not go thru.
again i apologize.
Tilghman Lesher wrote:
On Wednesday 06 August 2008 15:07:03 Nhadie wrote:
Would just like to know if anyone has encountered this:
You sent the exact same email this morning at 7:47 a.m. If nobody has
Randy:
Kudos to you for running the outstanding VOIP user's conference. I
have an idea to toss into this public forum. I'm hoping that you and
others will consider it give some feedback. The idea would be to
begin each show with comments corrections from the previous week's
show. Sometimes
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