Hallo Group!
My Name is Guenther Sohler and I registred to this group, because
I think asterisk could be interesting for me.
I have got a small server at home running linux.
It does NAT and a Firewall. There is an intranet with my home PC
and a hardware SIP phone.
This SIP phone registers at
hi,
and first off all ... welcome!
now it would be handy if you provide us with the name of your phone for
ex 'a linksys spa942' or somthing
kr,
Jan de Coster
Guenther Sohler wrote:
Hallo Group!
My Name is Guenther Sohler and I registred to this group, because
I think asterisk could be
My Phone identifies as
USer-Agent: ALL7950 02.09.23
I suppose its AllNet 7950
Hope this helps :)
Original-Nachricht
Datum: Thu, 20 Sep 2007 08:36:59 +0200
Von: Jan De Coster [EMAIL PROTECTED]
An: Asterisk Users Mailing List - Non-Commercial Discussion
Hi,
My Asterisk server process irregularly segfaults, ie.
it usually works fine (is stable) when there's low
traffic but repeatedly crashes during morning hours
when there are more calls.
I gdb'ed the core dump files and found that the
culprit may be format_mp3. So I disabled MOH today and
will
I have an asterisk 1.4, that was working properly,
but from last week, without any changing in the config of asterisk, all of
calls,fall in loop detected error.
there is two strange actions:
1-the first call after restarting the asterisk, is done successfully .
2-no packet , was sent to the
On Thursday 20 September 2007 11:34:44 Vieri wrote:
My Asterisk server process irregularly segfaults, ie.
it usually works fine (is stable) when there's low
traffic but repeatedly crashes during morning hours
when there are more calls.
I gdb'ed the core dump files and found that the
culprit
--- Atis Lezdins [EMAIL PROTECTED] wrote:
We also experienced this problem on 1.2, but i'm not
sure that this is
registered in bug database. You should check
bugs.digium.com and if it's
still valid for 1.4, you should post your backtraces
there.
Actually, I'm using 1.2.21.1 so since 1.2
Am Donnerstag, den 20.09.2007, 08:30 +0200 schrieb Guenther Sohler:
Hallo Group!
My Name is Guenther Sohler and I registred to this group, because
I think asterisk could be interesting for me.
Hi Guenther, this place probably is the right one. Welcome!
I have got a small server at home
We're having a horrid problem with our asterisk setup.
Sometimes calls just go dead - we can't hear what the other end is
saying. (I think they can't hear us either). The call doesn't hang up
until one of the callers gets bored.
Internaly we use Thomson ST2030 SIP phones.
Externaly we have 3
Hi all,
I have an interesting problem with Asterisk 1.4.11 - 3 SIP phones:
[phone1]
allow=g722
allow=alaw
[phone2]
allow=alaw
allow=g722
[phone3]
allow=alaw
Now, when I try to call:
1. phone1 calling phone2, I got through, using G.722 codec
2. phone2 calling phone1, I get through,
Router modell number ? On a linksys or netgear on incoming calls the wrong
phones start ringing (unless the router is sip aware)
- Original Message -
From: Jerry Jones [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent:
Hi John!
I have the same problem, the system contains 1 port Billion ISDN BRI
card, and 1 sip trunk. This is a trixbox with Asterisk
1.2.22-BRIstuffed-0.3.0-PRE-1y-i
The ISDN call is forwarded to a ring-group. The 6 sip phones are
welltech lp399 series.
If incoming the call get wrong, we can
On Thu, Sep 20, 2007 at 01:24:52PM +0200, Péter Tóth wrote:
Hi John!
I have the same problem, the system contains 1 port Billion ISDN BRI
card, and 1 sip trunk. This is a trixbox with Asterisk
1.2.22-BRIstuffed-0.3.0-PRE-1y-i
The ISDN call is forwarded to a ring-group. The 6 sip phones
On Thu, Sep 20, 2007 at 12:22:43AM +0200, Tzafrir Cohen wrote:
Why is it looking for files that obviously
don't exist?
That spec uses quite a few discourged methods for rpm packages. There
are a number of well-maintained RPM packages of Asterisk. Use one of
them or modify one of them.
On 9/19/07, Christoph Adomeit [EMAIL PROTECTED] wrote:
Especially I do not see how I could add a wait to the dialplan
as somebody suggested because there seems no dialplan invoked
when I send sms.
Can you not invoke a shell script and put the sleep in there?
Hi all,
I have since days now a strange problem with two Thomson ST2030 phones (FMW
3.56) on Asterisk 1.4.11. They are both in a queue (only one phone per queue
to get the MoH played ...)
No i see often times in the CDR these:
33012
2007-09-20 14:16:52+02 s
On Thu, 2007-09-20 at 12:49 +0200, John Hughes wrote:
We're having a horrid problem with our asterisk setup.
Sometimes calls just go dead - we can't hear what the other end is
saying. (I think they can't hear us either). The call doesn't hang up
until one of the callers gets bored.
Tzafrir Cohen wrote:
On Thu, Sep 20, 2007 at 01:24:52PM +0200, Péter Tóth wrote:
Just as you have rtp debug, you have bri debug .
bri debug span 1
and hope for a friendly ISDN guru on the list...
Does nothing for me - my isdn is connected via chan-capi.
On Thu, Sep 20, 2007 at 02:25:19PM +0200, Marcus Franke wrote:
Same situation for Ubuntu using the debian package format, but
I have not found a repository so far and Ubuntu delivers just
the old 1.2 release. :)
Ubuntu packages of Asterisk are slightly modified Debian ones.
As for the Debian
Am Mittwoch, den 19.09.2007, 15:25 +0200 schrieb Christoph Adomeit:
Hi there,
I experience the same problem here with asterisk 1.2.24 on
an E1 Line, only 2 of 3 sms are sent, this happens always and
is reproducable.
Did someone find out more about the problem ?
Especially I do not
Tzafrir Cohen wrote:
On Thu, Sep 20, 2007 at 01:24:52PM +0200, Péter Tóth wrote:
Hi John!
I have the same problem, the system contains 1 port Billion ISDN BRI
card, and 1 sip trunk. This is a trixbox with Asterisk
1.2.22-BRIstuffed-0.3.0-PRE-1y-i
The ISDN call is forwarded to a
Patrick wrote:
On Thu, 2007-09-20 at 12:49 +0200, John Hughes wrote:
We're having a horrid problem with our asterisk setup.
Sometimes calls just go dead - we can't hear what the other end is
saying. (I think they can't hear us either). The call doesn't hang up
until one of the callers
On 9/19/07, Alex Epshteyn [EMAIL PROTECTED] wrote:
Also, Asterisk restart results in all the watchers being lost. Is there a
way to force the phone to subscribe to notifications after restart (short
of
rebooting it) and is it phone specific?
Usually resubscribe-interval for extensions is
On 9/20/07, Marcus Franke [EMAIL PROTECTED] wrote:
Do you have any examples for these spec files?
I found a repository for installing Asterisk on Centos, but it
took a while before I discovered it. Ok, just checked the link
its for RHEL, but as Centos is just recompiled this won't matter.
On Thu, Sep 20, 2007 at 02:59:49PM +0200, John Hughes wrote:
Yeah, I was being lazy, just using the asterisk from the Debian
repositories.
Bristuff includes its own ancient version of chan_capi. Debian removes
it.
--
Tzafrir Cohen
icq#16849755
On Thu, 20 Sep 2007 02:15:11 am Scott Moseman wrote:
I'm getting frustrated simply trying to get this g729 working.
For what it is worth, I had a similar issue to you, and managed to get g729
working by installing the binary files from http://asterisk.hosting.lv
I'm trying a simple Echo test and it's failing for g729...
exten = 1267,1,Answer()
exten = 1267,2,Echo()
Test #1 (failure)
gateway33 codecs g729a, g729b
[gateway33]
type=friend
host=gateway33
context=default-inbound
disallow=all
allow=g729
gateway33 INVITE = g729b
Asterisk 200 OK = no media
On Thu, 2007-09-20 at 09:01 -0400, James FitzGibbon wrote:
www.atrpms.net has pretty solid RPMs, and you can grab the SRPMS in
order to get the spec file (either install the SRPM or use rpm2cpio to
convert the package and extract the specfile manually).
I've looked at a lot of Asterisk spec
Thanks for the reply. I actually found the problem by inserting print
statements in the code and checking which operations took the most time. I
found that one line was taking a long time to run. The line was the following
RTPManager.addtarget(destAddress)
After googling this for a while,
Hello, I've been looking at some SIP packet dumps captured with tcpdump on the
PBX itself, and analyzed with Wireshark 0.99.4. I'm noticing something
strange, at least to me. All of the SIP packets going out from our Asterisk
PBX to either of our 2 VoIP providers are consistently 50% out of
Did you set externip= ?
- Original Message -
From: Christian [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, September 12, 2007 3:23 PM
Subject: [asterisk-users] Problems with Asterisk behind a firewall
Hi
Or, in full:
[Sep 20 17:11:26] WARNING[18373]: app_queue.c:2705 try_calling: The
device state of this queue member, SIP/612, is still 'Not in Use' when
it probably should not be! Please check UPGRADE.txt for correct
configuration settings.
So, what do I check in UPGRADE.txt?
This is with
John Hughes wrote:
Or, in full:
[Sep 20 17:11:26] WARNING[18373]: app_queue.c:2705 try_calling: The
device state of this queue member, SIP/612, is still 'Not in Use' when
it probably should not be! Please check UPGRADE.txt for correct
configuration settings.
So, what do I check in
Dear All:
Just as the name suggests, and evolving from regular Click-to-Call,
Click-to-Call WITH VIDEO provides web sites with the ability to engage
their visitors with a live video agent (plus the phone call). All with just
a click of a button placed on the customer's web site. Please
Here is what I use.
sub devstate2str($)
{
#func name stolen directly from asterisk
#takes int devstate and returns string val
my $ids = shift;
my $devstatestring = {};
$devstatestring-{0} = Unknown; #0 AST_DEVICE_UNKNOWN | Valid,
but unknown state
Oh one other note, when asking questions such as this, it is really wise
to include which version # you are using.
Philipp Kempgen wrote:
Hi,
Is there a list of all the extension states as sent by the
manager interface? (I know I could look them up in the source
but that involves some
John Hughes wrote:
Or, in full:
[Sep 20 17:11:26] WARNING[18373]: app_queue.c:2705 try_calling: The
device state of this queue member, SIP/612, is still 'Not in Use' when
it probably should not be! Please check UPGRADE.txt for correct
configuration settings.
So, what do I check in
On 9/20/07, Luke Groeneveld [EMAIL PROTECTED] wrote:
I'm getting frustrated simply trying to get this g729 working.
For what it is worth, I had a similar issue to you, and managed to get
g729 working by installing the binary files from http://asterisk.hosting.lv
Thanks for the suggestion.
Tzafrir Cohen a écrit :
On Wed, Sep 19, 2007 at 12:25:35PM +0200, Benoît Mérouze wrote:
Is there any reason this can be fixed in the asterisk-perl-0.10
(not yet included in Trixbox)? Or is this more an issue from
Asterisk (since Asterisk 1.2.19 or 1.2.20)?
Why not give it a shot?
Install
I'm curious if anyone has implemented the following:
Need to setup an on-call queue, that activates after 5PM and de-activates at
8AM, also that activates/deactivates on demand(I'm thinking a feature code
here). The agents need to log in via cell phones, and when calls come in
from outside to
Anthony Francis wrote:
Here is what I use.
sub devstate2str($)
{
#func name stolen directly from asterisk
#takes int devstate and returns string val
my $ids = shift;
my $devstatestring = {};
$devstatestring-{0} = Unknown; #0 AST_DEVICE_UNKNOWN | Valid,
but
Anthony Francis wrote:
Oh one other note, when asking questions such as this, it is really wise
to include which version # you are using.
Right. Sorry.
1.4.11 (for the archives)
Regards,
Philipp Kempgen
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use
I am trying to connect two machines to each other with an T1 crossover
cable. The first machine has two TE120P cards - one connecting to the telco
on an ISDN PRI. The second to a crossover T1 cable to a second machine which
has one TE120P card.
Telco -cA- Machine1 -cB- Machine2
Machine1: Two
Philipp Kempgen wrote:
Anthony Francis wrote:
Here is what I use.
sub devstate2str($)
{
#func name stolen directly from asterisk
#takes int devstate and returns string val
my $ids = shift;
my $devstatestring = {};
$devstatestring-{0} = Unknown; #0
Philipp Kempgen wrote:
Anthony Francis wrote:
Oh one other note, when asking questions such as this, it is really wise
to include which version # you are using.
Right. Sorry.
1.4.11 (for the archives)
Regards,
Philipp Kempgen
That is what I thought, makes what I said
The device is a femtocell device; I would bet that they keep it in a format
that works with their existing equipment, rather than use SIP.
The device is also licensed for a specific frequency that is owned by the
carrier, you wouldn't be able to use this device for any other purpose without
Can we please block these clowns? It appears they are incapable of learning.
C. Savinovich wrote:
Dear All:
Just as the name suggests, and evolving from regular Click-to-Call,
Click-to-Call WITH VIDEO provides web sites with the ability to engage
their visitors with a live video agent
voip crazy wrote:
Hello all,
I am getting the following error in /var/log/syslog. I have got 2 B410P
cards in this box.
Sep 19 17:13:31 localhost kernel: hfcmulti_rx: fifo(0) reading 128 bytes
(z1=0153, z2=00d3) TRANS
Sep 19 17:13:31 localhost kernel: hfcmulti_tx: fifo(0) has 382 bytes
Does anyone know of an IAX softphone in Java, whether applet or
application? Even the most minimum featureset, just voice and dialing,
or even embedded in some other app/let. Preferably GPL. Thanks.
--
(C) Matthew Rubenstein
___
Sign up now
Alex Balashov wrote:
On Wed, 19 Sep 2007, Anthony Francis wrote:
IMHO asterisk is a softswitch, it may not be a very high capacity one
(right now) but it can be and if you don't mind splitting your physical
trunk calls over multiple machines it works very well as a call routing
engine,
On Thu, 20 Sep 2007, Matthew Fredrickson wrote:
Actually, I have been working on an SS7 stack for asterisk called
libss7. SS7 support is already in trunk, and should be in the next
stable release of Asterisk. Right now it only does ISUP/MTP3/MTP2
Look forward to it! That will certainly
He changed the title of his response. Your post remains intact on the
list in its original form. In the interest of letting others decide, I
think it was a spammy post as well. Others have decided... Hence the
title of the list Asterisk Users Mailing List - Non-Commercial
Discussion. They decided
On Thursday 20 September 2007, C. Savinovich wrote:
We posted in this forum because it is a contribution to the asterisk
community, and because it is free for a month, and maybe even longer
if the community so demands it. If you agree or disagree with it
fine, but let others decide. They
On Thu, 2007-09-20 at 13:20 -0400, Brian Alexander wrote:
Machine 1 experiences almost the same behavior on its span. The only
differance I am noticing is that instead of the S-frame error I get
the following notice:
chan_zap.c:8457 pri_dchannel: PRI got event: HDLC Bad FCS (8) on
I use click2call. http://www.geocities.com/babarnazmi/index2.htm
It is an activex control though.
--
--
Steven
http://www.glimasoutheast.org
Matthew Rubenstein [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
Does anyone know of an IAX softphone in Java, whether applet or
On 9/20/07, Eric Chamberlain [EMAIL PROTECTED] wrote:
The device is a femtocell device; I would bet that they keep it in a format
that works with their existing equipment, rather than use SIP.
The device is also licensed for a specific frequency that is owned by the
carrier, you wouldn't be
Matthew Rubenstein wrote:
Does anyone know of an IAX softphone in Java, whether applet or
application? Even the most minimum featureset, just voice and dialing,
or even embedded in some other app/let. Preferably GPL. Thanks.
Mexuar's Coraletta is nice, but isn't GPL.
I was interested in it - commercial or otherwise.but only because I
used to work for the competition.
Commercial or otherwise it looks like a very cool technology and
something I'd be interested in - but only as a one of purchase price
rather than an ASP.
Regards,
Dean Collins
Cognation
Matthew Rubenstein a écrit :
Does anyone know of an IAX softphone in Java, whether applet or
application? Even the most minimum featureset, just voice and dialing,
or even embedded in some other app/let. Preferably GPL. Thanks.
Did you try JIAXClient ?
http://www.hem.za.org/jiaxclient/
Steven, how reliable is that freeware?
I tried it when it first came out but I couldn't get it to work. It
didn't matter at the time as I was working for Mexuar at the time but
now I don't have their service anymore I'd like to use it/something like
it for my other consultancy services.
C F wrote:
AFAIK, the calls are free when you use it thru that device. Sprint
however charges $15 a month per phone or $30 for family plan. While I
agree that sprint should pay me for this, it's not as bad. T-mobile on
the other hand, does the same thing with wifi enabled phones, it
doesn't
snip
Please don't change the title of my post. It is
disrespectful. One thing
is to give your opinion about its content, and another to be
self appointed
editor of this forum.
We posted in this forum because it is a contribution to the asterisk
community, and because it is free for a
Hallo Martin and Group!
Thank you very much for your perfect introduction into asterisk.
I managed to
* get asterisk server running
* configuring the internal numbers
* registering to 2 sip gateways
* outbound phoning to sipgate works perfect
* outbound phoning to mujtelefon not yet tested
The
On Thu, 2007-09-20 at 14:23 -0400, Mike Clark wrote:
Matthew Rubenstein wrote:
Does anyone know of an IAX softphone in Java, whether applet or
application? Even the most minimum featureset, just voice and dialing,
or even embedded in some other app/let. Preferably GPL. Thanks.
OK gentlemen, thank you very much.
Best Regards
C. Savinovich
VideoReps.net
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tilghman
Lesher
Sent: Thursday, September 20, 2007 11:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
As far as I know Jiaxclient is dead - the developer hasn't touched it in at
least 18 months.
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL
Hi All,
I'm arriving around noon in Phoenix on Tuesday the 25'Th and wouldn't
mind sharing a cab or car service. I spoke with the hotel and the
'Super Shuttle' service can take 2-3 hours because the resort is the
last stop on the route. A cab or car service will only take 30-40
minutes.
If
Are you confident it's not a defect in Wireshark's RTP analyser?
On Thu, 20 Sep 2007, Jason Martin wrote:
Hello, I've been looking at some SIP packet dumps captured with tcpdump on the
PBX itself, and analyzed with Wireshark 0.99.4. I'm noticing something
strange, at least to me. All of the
Dear Dean:
It is a very cool technology indeed, and please, do not see me as your
competition, but as a friend. I know you have a click-to-call product, and
if there is any way I can be of help with providing the video technology for
you, I will be glad to set it up for you. You are most
Hello all,
We have an Asterisk server that has worked without issue for a while.
Before, only Sipura and Polycom 500 series phones were used.
Recently, we've added a few POE switches and 20 or so Polycom 330's.
The 330's seem to lock up often. One easy way to do this is by hitting
I haven't been involved with Mexuar for about 4 months.
In the middle of moving at the moment but will be in touch in about 2
weeks once things get back to normal.
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).
-Original
Hallo Martin and Group!
Thank you very much for your perfect introduction into asterisk.
I managed to
* get asterisk server running
* configuring the internal numbers
* registering to 2 sip gateways
* outbound phoning to sipgate works perfect
* outbound phoning to mujtelefon not yet tested
The
Dean Collins a écrit :
As far as I know Jiaxclient is dead - the developer hasn't touched it in at
least 18 months.
Correct, but this is free software, anybody with the skills can revive it :)
Regards,
--
Jean-Denis Girard
SysNux Systèmes Linux en Polynésie française
Speaking for the Asterisk community as a whole, we demand that it be
free forever. Please honor your statement below, We posted in this
forum because it is a contribution to the asterisk
community, and because it is free for a month, and maybe even longer
if the
community so demands it.
Please honor your statement below, We posted in this forum because it is
a contribution to the asterisk community, and because it is free for a
month, and maybe even longer
if the community so demands it.
I take your request as an official demand from the community to provide
the service for
I am having a weird issue with setting the recording file for the
Page app. Here is some quick background info
I have a macro that pages all my phones:
[macro-pageall]
; Context for paging all devices.
; This will search the sip table in the realtime database
; for all phones
[EMAIL PROTECTED] wrote:
Hello all,
We have an Asterisk server that has worked without issue for a while.
Before, only Sipura and Polycom 500 series phones were used.
Recently, we've added a few POE switches and 20 or so Polycom 330's.
The 330's seem to lock up often. One easy way
I've got a macro that tries to find the first available SIP trunk to send
outgoing calls on. It tracks the usage of the lines (since each trunk has a
call-limit of 2) by using GROUP(). My problem is that once a call switched
to ANSWER state, ``group show channels`` stops listing it and then my
Hi everyone,
I am running into wall today with simultaneous call limits. I have two
Asterisk machines (fast 3GHz C2D with 2GB of ram). I tried to create a
lot of sip calls from one machine to the other by issuing AMI Originate
commands to one machine. The machine that makes calls plays a
try
Nicholas Blasgen wrote:
I've got a macro that tries to find the first available SIP trunk to send
outgoing calls on. It tracks the usage of the lines (since each trunk has a
call-limit of 2) by using GROUP(). My problem is that once a call switched
to ANSWER state, ``group show
exten = 555,1,Dial(Local/1234567890/n)
note the /n
I'm going to try this in a bit (can't hurt anything, might as well), but I'd
like to understand you're reasoning. You're dialing an extra extension?
I'm also going to be trying this with Asterisk 1.6 TRUNK to see if it's even
a current
Just thinking about it quickly, it's always possible it has nothing to do
with Asterisk. There are many instances where I run into issues with a
poorly configured servers when they have even a little bump in HTTP
traffic. This was years ago though, and it was an issue to do with a web
server and
Hi Jeremy,
A few thoughts that come to mind. We have a queue that is open between
certain hours. I have a few checks in place before a caller enters,
first it checks to see if there it is within the time window, then
checks to see if there are any agents log into queue, if any fail they
get
Interesting. I am using PCLinuxOS(Mandrak) in console mode. Here is my
memory info and you can see that I still have a lot of memory while
asterisk is running
[EMAIL PROTECTED] ~]# cat /proc/meminfo
MemTotal: 2076000 kB
MemFree: 1855636 kB
Buffers: 17224 kB
Cached:
Nicholas Blasgen wrote:
exten = 555,1,Dial(Local/1234567890/n)
note the /n
I'm going to try this in a bit (can't hurt anything, might as well), but I'd
like to understand you're reasoning. You're dialing an extra extension?
I'm also going to be trying this with Asterisk 1.6 TRUNK to
On 9/20/07, Jason Parker [EMAIL PROTECTED] wrote:
C F wrote:
AFAIK, the calls are free when you use it thru that device. Sprint
however charges $15 a month per phone or $30 for family plan. While I
agree that sprint should pay me for this, it's not as bad. T-mobile on
the other hand, does
I am in need of some guidance regarding the following problem:
I need to dial an external number from a list(PSTN)
I need to check if the number is busy, no answer or fail
If any of the above are met then I try another number from a list
If none of the above happen then I first need to
Snip headers
On 9/20/07, Jason Parker [EMAIL PROTECTED] wrote:
C F wrote:
AFAIK, the calls are free when you use it thru that device. Sprint
however charges $15 a month per phone or $30 for family plan.
While I
agree that sprint should pay me for this, it's not as bad.
T-mobile on
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