2 apr 2007 kl. 10.16 skrev Dovid B:
Hi Guys,
I started getting this error only from one of our ITSP's once we
upgraded from 1.2.16 to 1.2.17.
Can anyone shed light ?
--- (12 headers 0 lines) ---
Transmitting (NAT) to 209.212.93.53:5060:
SIP/2.0 603 Declined (no dialog)
Via: SIP/2.0/UDP
2 apr 2007 kl. 10.46 skrev Olivier:
Hi,
What is the best way to implement Automatic Redial on No Answer ?
Looking at http://www.ietf.org/internet-drafts/draft-ietf-sipping-
service-examples-12.txt I can see how Automatic Redial on Busy
could (should) be done.
How would you do it on No
2 apr 2007 kl. 13.50 skrev sravana:
Anybody done LDAP authentication in Asterisk? can you explain how?
Thanks in advance
There's some code available in the issue tracker. Please check in
bugs.digium.com
for res_auth
/Olle
---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training
2 apr 2007 kl. 19.32 skrev Raj Jain:
I found a subtle difference between the two traces you sent (the
call that works and the call that gets dropped). This may or may
not be what's causing the problem.
The call that gets dropped had a retransmission of INVITE from UAC
to UAS (and
On Mon, 2007-04-02 at 22:12 -0400, Matthew Rubenstein wrote:
What it means is that Flash memory cells wear out after a large number
of read/write cycles, but not nearly as large as hard drives:
http://en.wikipedia.org/wiki/Flash_rom#Limitations . So using Flash in
place of RAM, even
On Mon, 2 Apr 2007, Matthew Rubenstein wrote:
But I'm not talking about using the Flash as RAM, just using it for a
low-load persistent store like a HD, where a HD would be overkill in
every way.
I boot my systems off a flash IDE drive. There's a partition with just
enough of a root
2 apr 2007 kl. 21.43 skrev John C. Wolosuk Jr.:
Does any one know if there's an mechanism (internal to asterisk or
otherwise) to replicate dynamic SIP device registrations across a
pool of asterisk servers?
I'm in the process of creating a asterisk cluster using a SIP
hardware load
2 apr 2007 kl. 22.21 skrev James FitzGibbon:
I'm building a dialplan for use with a bunch of GXP2000 desk sets.
During testing, we had some user issues surrounding the lack of an
on-phone dialplan. Users would hit 9 and sit there waiting for a
redial tone, and the GXP would time out,
On Mon, 2 Apr 2007, Peer Oliver Schmidt wrote:
Hello Armin,
[EMAIL PROTECTED]:~# grep capi /var/log/asterisk/messages
[Mar 31 16:17:03] ERROR[3850] chan_capi.c: Unable to listen on contr1
(error=0x100f)
Is this helpful, or do you need more information?
Yes, at this state it
On Sat, March 24, 2007 19:10, Bruce Reeves wrote:
You might get a faster response on freepbx/amp mailing list.
On 3/24/07, Francesco Peeters (Asterisk) [EMAIL PROTECTED] wrote:
SNIP
Just an update:
Still have NOT been approved for either the mailing list *or* the forum!
I am pretty
Je suis absent du 2/04/2007 au 11/04/2007.
Je répondrai à votre message dès mon retour. Pour toute urgence, contacter
Emmanuelle Parache Moga ou Cédric Buzay.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To
Hello,
I've wrote a dialplan script which uses the H extension to do something
similar to what you want. In general it uses the internal ASTDB for this:
- When there is no answer (or busy) the caller hangs up, initiate a new call
with some special code (*41 is used here by the public
I am trying to make a dialplan that when I dial 90 I can go round a
whole set of extensions and leave them a short message, hangup and go
on the next one.
I use the M facility of dial, with something like this
[messages]
exten = 90,n(calcnextchan),Set(DIALCHAN=...)
exten =
Ciao Patricio,
have you compiled Asterisk from sources?
At the moment you can only add chan_celliax support if you compiled from source.
If this is the case, I can give you full instruction.
Giovanni
On 4/3/07, Patricio Valarezo Lozano [EMAIL PROTECTED] wrote:
Hi, I've installed asterisk
Good Morning Armin,
tried the patch, but it did not work. It waits quite a long time
before the chan-capi error message comes up, according to the time
stamp it is about 12 seconds. It is kind of strange, that the whole
startup process for asterisk usually takes only about 4-5 seconds.
Olle,
It depends on how strictly the UA adheres to the offer/answer model. The
issue would be that a RE-INVITE from Asterisk will have the version
number incremented by more than one, which will break the following rule.
Quoting from RFC 3264 Section 8:
When issuing an offer that modifies
Hi all
I have a problem with an tdm400 with 2 modules 1 fxo 1 fxs it just
doesnt load the fxs module i dunno why...
zaptel.conf
loadzone=es
defaultzone=es
# qozap span definitions
# most of the values should be bogus because we are not really zaptel
span=1,1,3,ccs,ami
span=2,0,3,ccs,ami
The call that gets dropped had a retransmission of INVITE from UAC
to UAS (and therefore retransmission of 200 OK from UAS to UAC).
There is nothing wrong with the re-transmission as such, but I
noticed a potential bug in Asterisk in the way it responds to an
INVITE retransmission.
Monday, April 2, 2007, 7:30:57 PM, Giedrius wrote:
Has anybody debian and misdn working fine? Maybe you can advices , what
kernel and misdn versions to use...
I use kernel 2.6.20.1 and misdn 1.1.0 with fritz card, and working
fine. The kernel, asterisk (1.2.15) and misdn also compiled from
You can use the following to display what you receive from user (dtmf):
exten= 1,1,Read(test)
exten= 1,2,NoOp(DTMF Received: $test)
exten= 1,3,Hangup
On 4/3/07, Nitin Gupta [EMAIL PROTECTED] wrote:
I upgraded to 1.4.1 and my DTMF has stopped working, I tried
rfc2833compensate=yes and
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
Have you got zaptel installed on your box? And loading the zaphfc module
during bootup? I discovered the problem you reported when I switched to
mISDN and having installed/loading zaphfc during bootup. Than asterisk
doesn't start and the system
Hi,
I have a couple of questions about Quad-BRI solutions for Asterisk,
and was hoping that I might get some feedback based on other people's
experience.
We currently use the Junghanns card, which is a pure Zaptel solution,
which is fantastic, but they have no hardware EC solution, and their
Hello,
I've seen this already asked and answered but it is still a no go for
me.
I'm trying to do some preprocessing in the middle of a call, before
bridging.
I've seen two choices: M() and G() parameters of the Dial() command.
G() was discarded because I don't know if it is possible to
uxbod == [EMAIL PROTECTED] writes:
uxbod Hi, I have a requirement for sending and receiving faxes and
uxbod was wondering the best way to achieve it with Asterisk as I
uxbod only have one phone line.
uxbod I currently have a TDM11B in my server (1 x FXO, 1 x FXS), so I
uxbod was thinking that
Hi All,
I have a CentOS server that I am trying to configure Asterisk on 1.4 on.
Everything seems to go ok, with regards to compiling Zaptel, Libpri,
Asterisk (will be using kernel 2.6 timer and ztdummy)
Unfortunately I can't insmod / modprobe ztdummy.
[root @xyz src]# modprobe
Chris Blunt wrote:
I have a CentOS server that I am trying to configure Asterisk on 1.4 on.
Everything seems to go ok, with regards to compiling Zaptel, Libpri,
Asterisk (will be using kernel 2.6 timer and ztdummy)
Unfortunately I can't insmod / modprobe ztdummy.
Did you
yum install
Is it exists?
Regards,
Hong
Now that's room service! Choose from over 150,000 hotels
in 45,000 destinations on Yahoo! Travel to find your fit.
http://farechase.yahoo.com/promo-generic-14795097
On Tue, 3 Apr 2007, Peer Oliver Schmidt wrote:
Good Morning Armin,
tried the patch, but it did not work. It waits quite a long time
before the chan-capi error message comes up, according to the time
stamp it is about 12 seconds. It is kind of strange, that the whole
startup process for
Hello Armin,
here are the results, after modprobe kernelcapi showcapimsgs=3
/var/log/kern.log
Apr 3 15:06:14 server42 kernel: [255113.053170] CAPI Subsystem Rev 1.1.2.8
Apr 3 15:06:18 server42 kernel: [255116.814334] fcpci: AVM FRITZ!Card
PCI driver, revision 0.7.2
Apr 3 15:06:18
I'm trying to get the Mexuar development team to write some code to work
with an existing asterisk USER PORTAL that presents a user with
customized image of their Asterisk activities;
* Address book,
* Fop or some other kind of gui activity display
* Voicemail access
*
Je suis absent du 2/04/2007 au 11/04/2007.
Je répondrai à votre message dès mon retour. Pour toute urgence, contacter
Emmanuelle Parache Moga ou Cédric Buzay.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To
Hello,
We have 32 DECT clients connected to a Kirk Wireless 600/v3, the Kirk
server is connected to an Asterisk 1.2.17 with realtime configuration
(MySQL).
Our problem is that our Asterisk Server uses the latest inserted user
to places calls each time a call is made.
Exemple: we have 3 phones
Hi,
I needed my CDR's to be stored using a RADIUS server. I found cdr_radius in the src directory.
Looked in /docs for how to install it and I got it to work. Just want to say thanks to those who
helped write this.
Has anybody else used this, any comments, cause I found nothing using
This looks like the same issue I have with Astra phones, see the thread
Multi-line phones - Asterisk uses wrong callerid.
I do not know of a resolution for this yet.
regards,
Drew
Vincent renaville wrote:
Hello,
We have 32 DECT clients connected to a Kirk Wireless 600/v3, the Kirk
server
Dear
the following is the asterisk's dbase(Mysql5).
if the extension =17171000
asterisk run appdata=22, but I prefer to run
appdata=333.
let me know how I can run the appdata=3
best
Mani
mysql select * from ext;
Hello -
I've read Asterisk should be able to activate a do not disturb feature
to turn off the ringers on extensions. I checked the wiki and can't
find documentation for how to do it.
Here's my attempt, added to extensions.conf:
[dnd-on]
exten = _#78,1,Answer
exten = _#78,n,Wait(1)
exten =
Hello All,
I would like to only use the gsm codec for all the calls, is it possible I want
to use minimum possible bandwidth as we have most of calls over Internet.
Regards,
Sanjay Rajdev
___
--Bandwidth and Colocation provided by Easynews.com --
On Tue, 3 Apr 2007, Peer Oliver Schmidt wrote:
Hello Armin,
here are the results, after modprobe kernelcapi showcapimsgs=3
/var/log/kern.log
...
Apr 3 15:06:22 server42 kernel: [255120.102281] capi20: Rev 1.1.2.7: started
up with major 68 (middleware+capifs)
Apr 3 15:06:35
Brian McEntire wrote:
I've read Asterisk should be able to activate a do not disturb feature
to turn off the ringers on extensions. I checked the wiki and can't
find documentation for how to do it.
Here's my attempt, added to extensions.conf:
[dnd-on]
exten = _#78,1,Answer
exten =
Brian,
DND is not real hard. You basicly want to to note the extension is set to
DND and then when someone calls that extension you check for DND status and
if it is yes then you go on to voicemail instead of dial. It sounds like you
are miss understanding the dialplan and how to use it. In your
Try looking at this link:
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf
+sorting
Bobby
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
On 4/2/07, Klaverstyn, David C [EMAIL PROTECTED] wrote:
I have a brand new TE120P card that I have installed and asterisk is not
starting as I am getting the error: ERROR[5054] chan_zap.c: Unknown
signalling method 'pri_cpe'
Make sure you have libpri installed and that it is the right version
Anyone know of any tools for interpreting master.csv call logs?
(Excel is kind of basic)
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Group
I'm having trouble getting hints to work correctly using
SVN-branch-1.4-r59289
I have hints working on several other systems but I must be missing
something this time around.
VoIPGW*CLI show hints
-= Registered Asterisk Dial Plan Hints =-
[EMAIL
Hello Armin,
thanks a lot for your help.
Can you please do the same with 'showcapimsgs=2'?
It may give more info on the commands itself, maybe some parameters are
wrong here.
Here you go. 17:23:17 is the magic time.
Apr 3 17:23:09 server42 kernel: [263323.308388] fcpci: AVM FRITZ!Card
Bruce Reeves wrote:
exten = *73,1,Answer()
exten = *73,n,Wait(0.5)
exten = *73,n,Set(DB(${CALLERID(number)}/DND)=1)
Would prefer Set(DB(${DND/CALLERID(num)})=1)
exten = *73,n,Playback(do-not-disturb)
exten = *73,n,Playback(enabled)
exten = *73,n,Hangup()
and then
When someone calls
William Moore wrote:
On 4/2/07, Klaverstyn, David C [EMAIL PROTECTED] wrote:
I have a brand new TE120P card that I have installed and asterisk is not
starting as I am getting the error: ERROR[5054] chan_zap.c: Unknown
signalling method 'pri_cpe'
Make sure you have libpri installed and that it
Hello,
I have got two zap channels configured in our asterisk server, one of them
is connected to the PSTN directly and the other one is connected to a gsm
track, only for mobile calls.
Both of them are basic lines.
I just connect an iax softphone (idefisk) to the asterisk PBX. If I make a
I still can't figure out why res_config_mysql module not showing up with many
attempt. Anyone have any idea on this?
checking for mysql_config... /usr/bin/mysql_config
checking for mysql_init in -lmysqlclient... yes
configure: creating ./config.status
config.status: creating
On Tue, 3 Apr 2007, Sanjay Rajdev wrote:
Hello All,
I would like to only use the gsm codec for all the calls, is it possible
Yes, it's possible.
I want to use minimum possible bandwidth as we have most of calls over
Internet.
Good move if you're prepared to sacrifice call quality,
Hi Sanjay,
I'm not sure about that, but I think you can configure it in, for
example, /etc/asterisk/sip.conf.
There is an option that you configure for each channel like:
only=gsm
It instructs the sip protocol, that only gsm codec must be used.
I hope it has helped you.
Regards,
Ronaldo.
On Tue, Apr 03, 2007 at 10:18:56AM +1000, Klaverstyn, David C wrote:
OK,
Found the problem.
It looks like the configuration file is not correct.
I added the following line to /etc/sysconfig/zaptel
MODULES=$MODULES wcte12xp # TE120P - Single Span T1 Card
Actually, make that:
Je suis absent du 2/04/2007 au 11/04/2007.
Je répondrai à votre message dès mon retour. Pour toute urgence, contacter
Emmanuelle Parache Moga ou Cédric Buzay.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To
On 4/3/07, Olle E Johansson [EMAIL PROTECTED] wrote:
I turned on the early dial option on the GXP, which causes each
digit to be sent as it is pressed, and the user response was much
more favourable. Now I come to set up my international dialplans
and I'm running into a problem.
You
KC wrote:
I still can't figure out why res_config_mysql module not showing up with many
attempt. Anyone have any idea on this?
checking for mysql_config... /usr/bin/mysql_config
checking for mysql_init in -lmysqlclient... yes
configure: creating ./config.status
config.status: creating
On Tue, Apr 03, 2007 at 11:15:36AM +0200, Christoph Fürstaller wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
Have you got zaptel installed on your box? And loading the zaphfc module
during bootup? I discovered the problem you reported when I switched to
mISDN and having
On Mon, Apr 02, 2007 at 08:30:57PM +0300, Giedrius Augys wrote:
Hi,
I have FRITZ!Card PCI card. I have installed misdn-1.1.0 on stable debian
3.1 with 2.6.8. kernel. Then I reboot system, and it doesn't boot, it stops
near Apache2 starting I started my system with recovery kernel,
and
On Tue, Apr 03, 2007 at 11:57:57AM +0100, Chris Blunt wrote:
Hi All,
I have a CentOS server that I am trying to configure Asterisk on 1.4 on.
Everything seems to go ok, with regards to compiling Zaptel, Libpri,
Asterisk (will be using kernel 2.6 timer and ztdummy)
it shows empty string
On 4/3/07, Rizwan Hisham [EMAIL PROTECTED] wrote:
You can use the following to display what you receive from user (dtmf):
exten= 1,1,Read(test)
exten= 1,2,NoOp(DTMF Received: $test)
exten= 1,3,Hangup
On 4/3/07, Nitin Gupta [EMAIL PROTECTED] wrote:
I upgraded to 1.4.1
Good day everyone.
I have Cisco 1760 routers that do site to site voip. Each router has 2 fxs
ports that connect to the local pbx and use sip to connect to other routers
over the WAN. I am thinking of putting in an asterisk box at the hub site
for interconnectivity with our global office voip
That would be because $test is not a valid dialplan variable. You would
want ${test}
Nitin Gupta wrote:
it shows empty string
On 4/3/07, *Rizwan Hisham* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
You can use the following to display what you receive from user (dtmf):
On Tuesday 03 April 2007 07:48, Alan Chandler wrote:
I am trying to make a dialplan that when I dial 90 I can go round a
whole set of extensions and leave them a short message, hangup and go
on the next one.
I use the M facility of dial, with something like this
[messages]
exten =
Je suis absent du 2/04/2007 au 11/04/2007.
Je répondrai à votre message dès mon retour. Pour toute urgence, contacter
Emmanuelle Parache Moga ou Cédric Buzay.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To
Ahh... got it now. Thanks for all the replies.
I was thinking that it was a function that was already built in, but
I see by setting a value and then testing it before ringing
extensions, it's easily added to the dialplan.
On 4/3/07, Philipp Kempgen [EMAIL PROTECTED] wrote:
Bruce Reeves
Ok, I'll bite. This is the 4th message like this I've gotten today. I
don't speak French but it looks like an autoresponder. If so, why is it
replying back to the list, why not on every message sent, and why is it
incrementing the issue number?
Or am I missing something?
Jay
[EMAIL
I too was curious about this, so I copied the text into Babel Fish, and this is
the result:
I miss of the 2/04/2007 to the 11/04/2007. I will answer your message as of my
return. For any urgency, to contact Emmanuelle Parache Moga or Cédric Buzay.
If this guy is really going to be out until
Jay Moore wrote:
Ok, I'll bite. This is the 4th message like this I've gotten today. I
don't speak French but it looks like an autoresponder.
I'm away from ... to ...
I'm going to respond to your message when I'm back. In urgent
cases contact ... or ...
If so, why is it
replying back
Unless he's European, if so these messages will stop next Wednesday,
hopefully
Bails
john beaman wrote:
I too was curious about this, so I copied the text into Babel Fish, and this is
the result:
I miss of the 2/04/2007 to the 11/04/2007. I will answer your message as of my
return. For any
Philipp Kempgen wrote:
/kick him ;)
I'm not really sure whether fb is to blame or if mailman
should be in charge of filtering autoresponders. Thus I did
not put fb on my black(mail)list - yet. :)
Regards,
Philipp
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
john beaman wrote:
I too was curious about this, so I copied the text into Babel Fish, and this is
the result:
I miss of the 2/04/2007 to the 11/04/2007. I will answer your message as of my
return. For any urgency, to contact Emmanuelle Parache Moga or Cédric Buzay.
If this guy is really
john beaman wrote:
I too was curious about this, so I copied the text into Babel Fish, and this
is the result:
I miss of the 2/04/2007 to the 11/04/2007. I will answer your message as of
my return. For any urgency, to contact Emmanuelle Parache Moga or Cédric
Buzay.
If this guy is
John, it's the 11th of April not 4th of November.
I think everyone on this list should send [EMAIL PROTECTED] one of their
favorite photos, nothing rude or crass, just a nice thank you for wasting our
bandwidth.
Regards,
Dean
-Original Message-
From: [EMAIL PROTECTED]
November?
It's DD/MM/ in his case, not MM/DD/. Either way, even two days is more
than enough for me.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of john beaman
Sent: Tuesday, April 03, 2007 12:43
To: asterisk-users@lists.digium.com
Subject: Re:
On Tue, 3 Apr 2007, Peer Oliver Schmidt wrote:
Hello Armin,
thanks a lot for your help.
Can you please do the same with 'showcapimsgs=2'?
It may give more info on the commands itself, maybe some parameters are
wrong here.
Here you go. 17:23:17 is the magic time.
This log below
Greetings,
(Apologies if this is an FAQ, but I've Googled for hours and haven't
come up with anything yet.)
I have an Asterisk system deployed at a customer's site. It is connected
to the outside world by a local SIP provider. When someone calls in
through the trunk to leave a voicemail,
Odd question here but if I have asterisk running on PC (and mplayer
installed).
and a video phone calls up the asterisk PC can that video image be
played on mplayer?
If so how do I do that?
How can asterisk pipe the video into mplayer so as to display the video
image on screen?
Thanks,
john beaman wrote:
I too was curious about this, so I copied the text into Babel Fish, and this is
the result:
I miss of the 2/04/2007 to the 11/04/2007. I will answer your message as of my
return. For any urgency, to contact Emmanuelle Parache Moga or Cédric Buzay.
If this guy is really
Ah, yes. One of the many differences between the US and the rest of the world.
[EMAIL PROTECTED] 4/3/2007 2:52:16 PM
john beaman wrote:
I too was curious about this, so I copied the text into Babel Fish, and this
is the result:
I miss of the 2/04/2007 to the 11/04/2007. I will answer your
Hello Everyone,
I have two never-used, still in the static-bag T400P Cards that I bought a long
while back that I'd like to get rid of.
Before I ever got a chance to use them, I bought Sangoma A400 Cards instead,
and now I definately don't need them any longer.
They have the Dallas DS21Q352
Good evening Armin,
This log below shows no error in parameters, but the problem is still the
same: the fcpci driver doesn't respond and I cannot tell why.
Ok. Thanks for your assistants anyhow. What strikes me as strange ist
the fact, that turning on verbose helps to circumvent the problem.
Charles Ulrich wrote:
I have an Asterisk system deployed at a customer's site. It is connected
to the outside world by a local SIP provider. When someone calls in
through the trunk to leave a voicemail, Asterisk is not sending any RTP
packets back through the trunk after the beep is
Jerry Geis wrote:
Odd question here but if I have asterisk running on PC (and mplayer
installed).
and a video phone calls up the asterisk PC can that video image be
played on mplayer?
Afaik Asterisk does not have any support for video streams
yet.
Regards,
Philipp
--
amooma GmbH -
On Tue, Apr 03, 2007 at 08:50:02AM +0200, Giovanni Maruzzelli wrote:
Ciao Patricio,
have you compiled Asterisk from sources?
At the moment you can only add chan_celliax support if you compiled from
source.
If this is the case, I can give you full instruction.
And if from packages:
Hi,
I recently decided to change my setup from AsteriskNow to plain-asterisk
1.4, which I wanted to set up and configure myself on a server running
Debian Etch 64bit version.
Hardware:
Asrock motherboard, model 775Dual880-Pro, with a Celeron D running at
2.8GHz, 1GB memory, standard Nvidia GF4MX
Hi All!
Maybe a little of topic.
Bout coming from Sweden and needing to call
Lithuania a lot am I wondering if anyone on the
list could recommend a sheep service in Lithuania to connect my Asterisk to.
A local number are not necessary bout preferd for
incoming calls for my contacts.
Regards
On Tue, Apr 03, 2007 at 11:17:57PM +0200, bram kortleven wrote:
Hi,
I recently decided to change my setup from AsteriskNow to plain-asterisk
1.4, which I wanted to set up and configure myself on a server running
Debian Etch 64bit version.
Hardware:
Asrock motherboard, model
Hello Darryl,
* Darryl Dunkin [EMAIL PROTECTED] [03-04-07 12:56]:
November?
It's DD/MM/ in his case, not MM/DD/. Either way, even two days is
more than enough for me.
is the format not?
MM/DD/
DD.MM.
Best regards,
Matthias
--
Programming today is a race between
Is it possible to install a stun server on asterisk?
joe a.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
I'm new to this list, and I apologize if this is an already answered
question, but my Google-fu was not strong enough to find the answer if
it was.
I'm having a problem with DTMF on incoming IAX calls. For the first few
seconds of the call (between maybe 1 and 15, it varies from call to
Je suis absent du 2/04/2007 au 11/04/2007.
Je répondrai à votre message dès mon retour. Pour toute urgence, contacter
Emmanuelle Parache Moga ou Cédric Buzay.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To
I was unaware of that build, thanks Tzafrir!
But it seems old...
You can download the current complete sources with:
svn checkout http://www.celliax.org:8081/svn/celliax/branches/test1 test1
or you can download a livecd from www.celliax.org and test it without
install anything.
For compiling
Hi Everyone,
I am using Zaptel and Asterisk 1.4 and have a Digium card with two FXS
modules. The card works and ztcfg reports that it finds the two
modules.
Howevery when I try and place a call through the gateway I get the
following error message. I have tried to refer to the ZAP device as
Brian McEntire wrote:
Hello -
I've read Asterisk should be able to activate a do not disturb feature
Instead of using 2 extensions, you can get away with just one. Check
the database entry at the start, if it's already set, remove it. If
it's not there, add it.
[dnd]
;
Devraj Mukherjee wrote:
Hi Everyone,
I am using Zaptel and Asterisk 1.4 and have a Digium card with two FXS
modules. The card works and ztcfg reports that it finds the two
modules.
Howevery when I try and place a call through the gateway I get the
following error message. I have tried to refer
Check this out HYPERLINK
javascript:ol('http://www.voip-info.org/wiki-Asterisk+cisco+FXO');http://w
ww.voip-info.org/wiki-Asterisk+cisco+FXO
_
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Joesph
Enviado el: Martes, 03 de Abril de 2007 02:53 p.m.
Para: Asterisk Users
Ok..
I have one box running Asterisk - Box1, and I'm trying to get another setup
out on the internet (Box2) with an IAX2 trunk connecting the two. The calls
flow fine from Box2 to Box1, but when I call Box2 from Box1 the Called
Number always shows up as 's'. Why wont it pass the DID?
I have spandsp, rxfax and asterisk-1.4.2 installed and whenever a fax call
comes in we get this. This isn't good. Any ideas?
[New Thread -1215390800 (LWP 8504)]
-- Accepting call from 'DELETED' to '539' on channel 0/1, span 1
-- Executing [EMAIL PROTECTED]:1] Set(Zap/1-1, DIALEDNUM=539)
Hi Eric,
Thanks for your suggestion
I just reinstalled Asterisk, it still doesn't seem to know anything
about Zaptel. I am using CentOS and installed Asterisk using yum from
ATrpms.
Anything else I can try?
On 4/4/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Devraj Mukherjee wrote:
Not every client supports gsm. Usually it's a good idea to put ulaw as well
or you could get errors when neither side supports the same codec.
disallow=all
allow=gsm
allow=ulaw
--
Salvatore Giudice
[EMAIL PROTECTED]
VoIP Security Training, LLC
From: Devraj Mukherjee [EMAIL PROTECTED]
Date: Wed, 4 Apr 2007 11:46:11 +1000
Hi Eric,
Thanks for your suggestion
I just reinstalled Asterisk, it still doesn't seem to know anything
about Zaptel. I am using CentOS and installed Asterisk using yum from
ATrpms.
Anything else I can try?
Try
1 - 100 of 105 matches
Mail list logo
