Hi friends,
I am facing a problem with my asterisk 1.2 PBX. The problem is because of
the CLI message Remote UNIX Connection. After 2 days of a server reboot,
this message starts coming. After it starts coming it still works well for
few more hours, but then the asterisk hangs. During this
Dear Asterisk community!
I am having trouble with a project concerning the 183 Session Progress SIP
messages. Asterisk seems to only accept these when there is also a Session
Description (SDP) included in the message.
I also verified this by looking at the code.
However for a project we
On 27 Jun 2009, at 10:06, Olivier wrote:
Hi,
As many remember, almost one year this Skype for Asterisk extension
program was announced.
Has anyone tried it ?
Is there any available pricelist ?
I've just had a talk on Skype for Asterisk accepted at Astricon (www.astricon.net
), so
if
Hi there,
I'm having some fairly serious asterisk problems, which seem to be spread
quite liberally across all asterisk versions, I've tried 1.4.2, 1.6.0.10,
1.6.2beta4 and still had exactly the same problem with static and echo on
the line when using the PRI interface.
A little background:
Hello
Can I configure one h323 endpoint to register to asterisk?
Which asterisk version can support this?
Regards
Bilal
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On Tue, Jun 30, 2009 at 10:38:25AM +0400, Shanavaz E A wrote:
Hi friends,
I am facing a problem with my asterisk 1.2 PBX.
What version, exactly?
The problem is because of
the CLI message Remote UNIX Connection.
Read: 'asterisk -r' connecting.
After 2 days of a server reboot,
Hello!
I am trying to set up a dialplan that uses the Local channel type:
[default]
exten = s,1,dial(local/2...@dialplan/n)
[dailplan]
exten = 220,1,saydigits(123)
exten = 220,2,dial(SIP/120||m)
The calling party does not hear any of the digits nor the music on
hold. What should be done so
Hi all
After a long iam back to forum
back to my own topic and several readings done on this forum
how people doing same kind of setup what iam trying to achive
so here i have done some good developements
for testing iam doing all in one Server
Step1 :
Installed in Fresh BOX with Debian
Hello,
Oddly enough, sound is sent to original caller if it is a registered
SIP device on the server. If the caller is remote, than nothing is
passed back.
Any help will be greatly appreciated,
Elliot
On Tue, Jun 30, 2009 at 12:13 PM, Elliot Murdockmurdo...@gmail.com wrote:
Hello!
I am
What version do you mean.. 1.6?
Upgrading might be a option, but we cant loose any functionality/stability
- Original Message -
From: Paul Hales
[mailto:pdha...@optusnet.com.au]
To: Asterisk Users Mailing List -
Non-Commercial Discussion [mailto:asterisk-us...@lists.digium.com]
Sent:
Hello users.
i am planning to implement the dundi protocol among 3 servers
where the real channels residing in 2 servers and the remaining one
is only for routing purpose..
here is how my config files
#Routing_server
routing server -192.168.1.11
node1-192.168.1.21
node2
Hello!
I needed to answer the local call for any sound to pass through:
[default]
exten = s,1,dial(local/2...@dialplan/n)
[dailplan]
exten = 220,1,answer()
exten = 220,2,saydigits(123)
exten = 220,3,dial(SIP/120||m)
From my understanding, the answer command only answers the local call,
but
Hi there,
I'm having some fairly serious asterisk problems, which seem to be spread
quite liberally across all asterisk versions, I've tried 1.4.2, 1.6.0.10,
1.6.2beta4 and still had exactly the same problem with static and echo on
the line when using the PRI interface.
A little background:
Originally posted on asterisk-dev with no response for 5 days, so posting it
to the wider audience now.
Asterisk Release 1.6.1.1
Scenario:-
1. 2 SIP peers (Zoiper softphone, if it matters) registered as 901 and
902
2. Using AMI, 901 is Originated
3. When 901 answers, it is
Hi,
I'm aware you didn't get a response. But please only post once, or at
least leave a day or two.
Is the PRI a known-good? i.e. tested with other stuff and error free?
Steve
On 30 Jun 2009, at 11:45, Tom O'Connor wrote:
Hi there,
I'm having some fairly serious asterisk problems, which
On Tue, Jun 30, 2009 at 5:02 AM, Tom O'Connor t...@twinhelix.org wrote:
Hi there,
I'm having some fairly serious asterisk problems, which seem to be spread
quite liberally across all asterisk versions, I've tried 1.4.2, 1.6.0.10,
1.6.2beta4 and still had exactly the same problem with static
Hi Ram,
Does your OpenSIPS get any SIP reply from Asterisk? or the INVITE is
simply discarded by Asterisk?
Regards,
Bogdan
Hi all
After a long iam back to forum
back to my own topic and several readings done on this forum
how people doing same kind of setup what iam trying to achive
so
On Tue, Jun 30, 2009 at 10:02:29AM +0100, Tom O'Connor wrote:
Asterisk works fine for SIP calls, as long as they don't touch the outside
world via the PRI card.
This pastebin contains the console log from asterisk
-vcg
http://pastebin.com/f780c591e
There are lots
On Tue, Jun 30, 2009 at 1:31 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Tue, Jun 30, 2009 at 10:02:29AM +0100, Tom O'Connor wrote:
Asterisk works fine for SIP calls, as long as they don't touch the
outside
world via the PRI card.
This pastebin contains the console log from
Hello,
i have just installed asterisk 1.6.0.10 on debian 5.0 like:
./configure;make menuselect; make;make install
There are no erorrs, but folder /usr/lib/asterisk/modules is empty.
What am i doing wrong? Where are modules?
p.s. Doing the same on Slackware, i ve got all selected modules at
On Tue, Jun 30, 2009 at 6:45 AM, Tom O'Connortom.bio...@gmail.com wrote:
I'm having some fairly serious asterisk problems, which seem to be spread
quite liberally across all asterisk versions, I've tried 1.4.2, 1.6.0.10,
1.6.2beta4 and still had exactly the same problem with static and echo on
On Tue, Jun 30, 2009 at 04:57:29PM +0400, M C wrote:
Hello,
i have just installed asterisk 1.6.0.10 on debian 5.0 like:
./configure;make menuselect; make;make install
There are no erorrs, but folder /usr/lib/asterisk/modules is empty.
What am i doing wrong? Where are modules?
ls -l
I've set up an outbound .call system for customer callbacks and the
like. Calls are going out over analog lines and I'm trying to use the
WaitForSilence routine to make sure the phone has stopped ringing before
starting message playback. The problem is that if I set the first
argument of
On Tue, Jun 30, 2009 at 2:16 PM, David Backeberg dbackeb...@gmail.comwrote:
On Tue, Jun 30, 2009 at 6:45 AM, Tom O'Connortom.bio...@gmail.com wrote:
I'm having some fairly serious asterisk problems, which seem to be spread
quite liberally across all asterisk versions, I've tried 1.4.2,
2009/6/30 Tzafrir Cohen tzafrir.co...@xorcom.com
On Tue, Jun 30, 2009 at 04:57:29PM +0400, M C wrote:
Hello,
i have just installed asterisk 1.6.0.10 on debian 5.0 like:
./configure;make menuselect; make;make install
There are no erorrs, but folder /usr/lib/asterisk/modules is
There are no erorrs, but folder /usr/lib/asterisk/modules is empty.
What am i doing wrong? Where are modules?
Yes, i ve embeded all modules in menuselect
Uh..
___
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asterisk-users
Hi,
I am using a basic VOIP phone (find here:
http://www.tootoo.com/d-rp20207560-VoIP_phone/) on an Asterisk
1.2.26-BRIstuffed-0.3.0-PRE-1y-q version of asterisk. I am running a C-based
prepaid application based on MySQL that accepts dtmf events from the phone
to authenticate. When asterisk is
On Tuesday 30 June 2009 08:24:29 Tom O'Connor wrote:
I'm currently
pointing fingers at either the hardware (someone on #asterisk said it could
be a cruddy chipset, but it's an HP Server.. so should be kosher.. ), I
Is it an HP server from the HP server line, or is it an HP server from the old
cdr_mysql doesn't set the userfield when it's set inside a macro
called from a feature (1.4.25, addons 1.4.8).
I have a feature code:
autorecord = *1,self,Macro,apprecord
The apprecord macro looks like:
[macro-apprecord]
exten = s,1,Playback(beep)
exten =
I run a phone in a remote office using the IAX2 protocol. It mostly works
fine; except that every 5 mins it loses connection with Asterisk, before
reconnecting 30 seconds later; rinse repeat.
Using the IAX2 debugging, I'm seeing this a lot:
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000
Elliot Murdock murdo...@gmail.com writes:
I needed to answer the local call for any sound to pass through:
[default]
exten = s,1,dial(local/2...@dialplan/n)
[dailplan]
exten = 220,1,answer()
exten = 220,2,saydigits(123)
exten = 220,3,dial(SIP/120||m)
From my understanding, the answer
On Tue, Jun 30, 2009 at 3:12 PM, Tilghman Lesher
tilgh...@mail.jeffandtilghman.com wrote:
On Tuesday 30 June 2009 08:24:29 Tom O'Connor wrote:
I'm currently
pointing fingers at either the hardware (someone on #asterisk said it
could
be a cruddy chipset, but it's an HP Server.. so should
Check this issue, seems related
https://issues.asterisk.org/view.php?id=14662
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Russell Brown
Sent: martes, 30 de junio de 2009 11:33 a.m.
To:
Hi,
Am Montag, den 29.06.2009, 10:35 -0400 schrieb Kayton Sapale:
That's the strange thing. Nothing shows when monitoring the service
in debug. On the phone, however, I do see a connection time-out
error. I guess this might indicate that the device is attempting to
connect to the service
Tom O'Connor wrote:
On Tue, Jun 30, 2009 at 3:12 PM, Tilghman Lesher
tilgh...@mail.jeffandtilghman.com
mailto:tilgh...@mail.jeffandtilghman.com wrote:
On Tuesday 30 June 2009 08:24:29 Tom O'Connor wrote:
I'm currently
pointing fingers at either the hardware (someone on
Floimair Florian schrieb:
I am having trouble with a project concerning the 183 Session Progress SIP
messages. Asterisk seems to only accept these when there is also a Session
Description (SDP) included in the message.
I also verified this by looking at the code.
Which version of
Hi all,
we face a problem with SMS reception sended to _landlines_, at least in
France.
Normally operators -tested with France Telecom and SFR- are sending
voice SMSs from a particular CID number, so no problem. But today we
discover that -at least SFR- send from time to time voice SMSs with
This is not a clean or efficient solution, but you could use an AGI or .call
file to sent the SMS as a separate call.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Administrator
TOOTAI
Sent: Tuesday, June 30,
Hello!
I am looking for a way to dynamically redirect a call while it is
ringing to another device. Basically, if a person is far away from
his desk, he should have the option to use another phone and pick up
the call.
Thanks for any suggestions,
Elliot
If it is configured and working correctly, *8 picks up the ringing line from
any eligible phone.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Elliot Murdock
Sent: Tuesday, June 30, 2009 2:21 PM
To: Asterisk
Elliot Murdock schrieb:
I am looking for a way to dynamically redirect a call while it is
ringing to another device. Basically, if a person is far away from
his desk, he should have the option to use another phone and pick up
the call.
Pickup() application?
Philipp Kempgen
--
AMOOMA
I'm in the process of converting our current hybrid key system to
Asterisk and Aastra 57i phones. One of the features that seems to be a
show stopper for almost everyone in the office is the inability to see
who is on the phone. Can someone point in the right direction to setup
an XML app on the
On Tue, Jun 30, 2009 at 4:17 PM, Jeremy Winder jwin...@logicalsi.comwrote:
I'm in the process of converting our current hybrid key system to
Asterisk and Aastra 57i phones. One of the features that seems to be a
show stopper for almost everyone in the office is the inability to see
who is on
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Jeremy Winder wrote:
I'm in the process of converting our current hybrid key system to
Asterisk and Aastra 57i phones. One of the features that seems to be a
show stopper for almost everyone in the office is the inability to see
who is on the
Hello, all. I must be brain cramping badly on our Asterisk 1.6.1.1
installation. Our MeetMe macros are working fine except they do not
prompt for a PIN. So I made a very simple conference room:
exten = ,1,MeetMe(123456,cMaAsx,123456)
Shouldn't this prompt the user who dials to enter
Has anyone ever gotten an Adit 600 to work with Asterisk1.4 via MGCP.
Asterisk keeps giving me the following error in the LOGs:
[Jun 30 08:32:59] NOTICE[26785]: chan_mgcp.c:1726
find_subchannel_and_lock: Gateway 10.0.0.245' (and thus its endpoint
'*') does not exist
MGCP Config:
On Tue, Jun 30, 2009 at 3:17 PM, Jeremy Winderjwin...@logicalsi.com wrote:
I'm in the process of converting our current hybrid key system to
Asterisk and Aastra 57i phones. One of the features that seems to be a
show stopper for almost everyone in the office is the inability to see
who is on
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Barry L. Kline wrote:
If you'd like a more generalized approach you can install an Openfile
server and use the Asterisk plugin. That'll give you an internal IM
server which will show the status you seek.
Sorry, not 'openfile' but 'openfire'.
No
That says:
Join Conference 123456
The PIN for the Conference is 123456
What you need to do is setup the conference ID, guest PIN, admin PIN in
the meetme.conf
and then use
exten = ,1,MeetMe(123456,cMaAsxp,)
John A. Sullivan III wrote:
Hello, all. I must be brain cramping badly
On Tue, 2009-06-30 at 16:17 -0400, Jeremy Winder wrote:
I'm in the process of converting our current hybrid key system to
Asterisk and Aastra 57i phones. One of the features that seems to be a
show stopper for almost everyone in the office is the inability to see
who is on the phone. Can
I've got an issue where I am trying to route calls between Asterisk
Servers. I can route calls inbound to a server but seem to have an
authentication issue going out over the same sip account. It appears
that my server isn't sending the second invite after proxy
authentication request. I
Hi All,
I have a problem with my Asterisk Server that the logs aren't giving me
any clue to what's going on.
The server is running 1.6.1.1 and connected to a Grandstream GXP2000
phone. At 3:58 minutes the call cuts off with no indication in the
log. This is random and is only localized to
Thanks again Jim. I seem to be successful in using this method but now I get
the following after the call completes. It seems that asterisk doesn't know
what to do with the first channel. Would this indicate I am missing a
Hangup() somewhere?
Thx.
:
[Jun 30 18:31:30] WARNING[26484]: pbx.c:3907
I have had weird issues with that model, too. Have you tried reseting the
phone to factory defaults and then reconfigure?
The directions to reset can be found here:
http://sipx-wiki.calivia.com/index.php/HowTo_configure_Grandstream_SIP_phone
_with_sipX
hope this helps
-Original Message-
Try upgrading the firmware on it. They have all sorts of goofy bugs.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Todd Reese
Sent: Tuesday, June 30, 2009 4:56 PM
To: Asterisk Users Mailing List -
I did the upgrade to the phone. And the problem continued. Currently,
as per the previous poster, I have reset the phone to the factory
default and have started setup again.
Peder wrote:
Try upgrading the firmware on it. They have all sorts of goofy bugs.
-Original Message-
Hi,
I am using Originate in testing and also using call files in testing. I also
needed to capture DIALSTATUS and update my CDRs accordingly. My original
attempts at using Originate (or call files) did not report {DIALSTATUS} if
the call could not be connected (e.g. bad phone number like
When I login to the asterisk, I just hear the HALF of the welcome message :
You are currently the instead of You are currently the only person in
the conference
Thats also, I hear it after 60 secs or so..
Asterisk 1.2.27
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I have gotten around the issue by adding the following to the dialplan
before sending to MeetMe:
exten = XXX,1,Playback(/var/lib/asterisk/sounds/silence/1)
It seems to be a bug in Asterisk as far as I can tell. Hope that helps!
- Josh
David @ULC wrote:
When I login to the asterisk, I
On Tue, Jun 30, 2009 at 04:57:29PM +0400, M C wrote:
Hello,
i have just installed asterisk 1.6.0.10 on debian 5.0 like:
./configure;make menuselect; make;make install
any reason to not use the deb files ?
There are no erorrs, but folder /usr/lib/asterisk/modules is empty.
What am i
Thanks for the Reply,
I was waiting online for someone to reply : -)
Here is my Extension file : [ Where should I enter those line ? ]
exten = 8600099,1,Meetme(8600099)
exten = 8600100,1,Meetme(8600100)
exten = 8601,1,Meetme(8601)
exten = h,1,DeadAGI(agi://127.0.0.1:4577/call_log)
exten =
What do you do if you find things sharing interrupts (IRQ 11) in my case
with my X100P card. I believe there is some sort of internal audio card
in my cheap slow PC.
Alejandro Kauffmann wrote:
Tom O'Connor wrote:
On Tue, Jun 30, 2009 at 3:12 PM, Tilghman Lesher
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