hi,
i have installed a2billing , when i open /admin web pages. errors as follow:
Fatal error: Call to undefined function bindtextdomain() in
/usr/local/src/a2billing/common/lib/languageSettings.php on line 130
do you know what's wrong?
--
Thanks,
Sucan
On Mon, Dec 28, 2009 at 10:39 PM, Zhang Shukun bit...@gmail.com wrote:
hi,
i have installed a2billing , when i open /admin web pages. errors as
follow:
Fatal error: Call to undefined function bindtextdomain() in
/usr/local/src/a2billing/common/lib/languageSettings.php on line 130
do you
OK. Thanks
2009/12/29 ram talk2...@gmail.com:
On Mon, Dec 28, 2009 at 10:39 PM, Zhang Shukun bit...@gmail.com wrote:
hi,
i have installed a2billing , when i open /admin web pages. errors as
follow:
Fatal error: Call to undefined function bindtextdomain() in
2009/12/29 Zhang Shukun bit...@gmail.com:
OK. Thanks
2009/12/29 ram talk2...@gmail.com:
On Mon, Dec 28, 2009 at 10:39 PM, Zhang Shukun bit...@gmail.com wrote:
hi,
i have installed a2billing , when i open /admin web pages. errors as
follow:
Fatal error: Call to undefined function
Hi all,
*I have problem about sip : SIP/2.0 401 Unauthorized
*domain = rajnikant.net ( its ipaddress is 172.18.100.74 - kamailio server )
when i have call from 1...@rajnikant.net user to 2...@rajnikant.net this error
occured *SIP/2.0 401 Unauthorized*
Asterisk server on 172.18.100.65
sip.conf
Hi Friends,
need to help.
*I have problem about sip : SIP/2.0 401 Unauthorized*
Is it require to nathelper module in kamailio ?
*what can i write kamailio.cfg file when kamailio and Asterisk on same
network?*
Scenario is like as :
-
1) kamailio server on
Hi
I have a problems with a new Asterisk Server,
when i want call, i have:
[Nov 14 09:12:38] NOTICE[31992]: chan_sip.c:18160
handle_request_invite: Call from 'PHISIP01' to extension
'00420225352184' rejected because extension not found.
but into my extensions.conf:
exten =
On Sat, 14 Nov 2009, Phibee Network Operation Center wrote:
when i want call, i have:
[Nov 14 09:12:38] NOTICE[31992]: chan_sip.c:18160
handle_request_invite: Call from 'PHISIP01' to extension
'00420225352184' rejected because extension not found.
(You don't say what version of
Phibee Network Operation Center wrote:
Hi
I have a problems with a new Asterisk Server,
when i want call, i have:
[Nov 14 09:12:38] NOTICE[31992]: chan_sip.c:18160
handle_request_invite: Call from 'PHISIP01' to extension
'00420225352184' rejected because extension not found.
Hi
when i use MeetMe, i have this errors:
app_meetme.c: Unable to open pseudo device
Where is the problems ?
i have too warning and error into my logs:
[Nov 1 07:26:17] WARNING[18544] res_musiconhold.c: Unable to open
pseudo channel for timing... Sound may be choppy.
[Nov 1 07:26:17]
You need to have the dadhi_dummy driver loaded, or have a digium (or
similar) card plugged in. Meetme needs a timing source. dahdi_dummy is
used as the timing source in case you dont have a card.
sam!!
Phibee Network Operation Center wrote:
Hi
when i use MeetMe, i have this errors:
while compiling zaptel drivers for my yeaster TDM800 hardware, I get this error;
make[3]: Leaving directory `/usr/src/zaptel-1.4.12/menuselect/mxml'
gcc -o menuselect menuselect.o strcompat.o menuselect_curses.o mxml/libmxml.a
mxml/libmxml.a -lncurses
make[2]: Leaving directory
Hi
Just download tar.gz of your kernel version and extract into
/usr/src/kernels/ directory
!
--
Regards,
Chandrakant Solanki
On Wed, Oct 21, 2009 at 1:34 PM, PATRICK KANGETHE patricemb...@yahoo.comwrote:
while compiling zaptel drivers for my yeaster TDM800 hardware, I get this
Thanks solanki it worked fine.
From: Chandrakant Solanki solanki.chandrak...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wed, October 21, 2009 1:45:42 PM
Subject: Re: [asterisk-users] error
Hello list !
SETUP :
Grandstream --sip-- Local Asterisk (NSLU) --iax-- Hosted Asterisk
(VirtualDedicatedServer) --sip-- SIPprovider -- my CellPhone
PROBLEM :
I've noticed that when I put down the horn of my Grandstream it still
takes a while for my GSM/CellPhone to stop ringing.
INFORMATION :
Discussion
Subject: Re: [asterisk-users] Error When Using Postgresql Schema
WithRealtime Sip
On Thursday 24 September 2009 05:06:02 stephen.hindma...@bt.com wrote:
I have investigated further and found that it is a bug in ODBC, not
Asterisk. The SQLColumns function, which asterisk uses
I have investigated further and found that it is a bug in ODBC, not
Asterisk. The SQLColumns function, which asterisk uses to describe the
table, does not return any columns when the table name includes the
schema specification. You can show this by using isql to do help table
which returns info
On Thursday 24 September 2009 05:06:02 stephen.hindma...@bt.com wrote:
I have investigated further and found that it is a bug in ODBC, not
Asterisk. The SQLColumns function, which asterisk uses to describe the
table, does not return any columns when the table name includes the
schema
I am using asterisk 1.6.1.6 and have been setting up a system to use a
Postgresql database as the realtime DB via the ODBC route. I have got
extensions and voicemail working but am having trouble with SIP
The problem seems to be with using a schema. If I put the table sip in
the schema foo then I
On Wednesday 23 September 2009 05:49:54 stephen.hindma...@bt.com wrote:
I am using asterisk 1.6.1.6 and have been setting up a system to use a
Postgresql database as the realtime DB via the ODBC route. I have got
extensions and voicemail working but am having trouble with SIP
The problem
Tilghman Lesher wrote:
On Wednesday 23 September 2009 05:49:54 stephen.hindma...@bt.com wrote:
I am using asterisk 1.6.1.6 and have been setting up a system to use a
Postgresql database as the realtime DB via the ODBC route. I have got
extensions and voicemail working but am having trouble
Hi,Excellent! Thank you so much for the tip. I updated /etc/asterisk/modules.conf and un-commented the following lines: preload = res_odbc.so preload = res_config_odbc.soAnd I was good to go. Much appreciated.Todd
Original Message
Subject: Re: [asterisk-users] Error loading
Hi,
I'm getting the following at asterisk startup. OCBC was working with
1.4 no problem, but now under 1.6 (I've tried 1.6.1, 1.6.2-beta3/beta4)
I can't seem to get rid of this anyone?
WARNING[32664]: loader.c:385 load_dynamic_module: Error loading
module 'res_config_odbc.so':
On Friday 28 August 2009 18:46:16 Todd Fulton wrote:
Hi,
I'm getting the following at asterisk startup. OCBC was working with
1.4 no problem, but now under 1.6 (I've tried 1.6.1, 1.6.2-beta3/beta4)
I can't seem to get rid of this anyone?
WARNING[32664]: loader.c:385
On about 25% of inbound calls to a ring group, picking up any one
extension as it rings results in dead air.
Some details regarding my VoIP network to make the following logs more
readable:
192.168.7.130 resolves to the trixbox host.
192.168.7.135 resolves to endpoint 812.
192.168.7.137
Does anyone have any light to shed on:
c_avpair_new: unknown attribute
sip02 asterisk[7796]: rc_avpair_new: unknown attribute 1490026597
We are getting congestion errors on a Pri to telco, and not sure what is
going on.
Thanks
Cary Fitch
___
--
Cary Fitch wrote:
Does anyone have any light to shed on:
c_avpair_new: unknown attribute
sip02 asterisk[7796]: rc_avpair_new: unknown attribute 1490026597
We are getting congestion errors on a Pri to telco, and not sure what is
going on.
Doing a google search gave an indication that
: Tuesday, July 14, 2009 11:34 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Error
Cary Fitch wrote:
Does anyone have any light to shed on:
c_avpair_new: unknown attribute
sip02 asterisk[7796]: rc_avpair_new: unknown attribute 1490026597
We
Hello,
I am trying to create a simple IVR for testing..
What i did is to create a voice file from asterisk-gui.
And i saw it created that under /var/lib/asterisk/sounds/record/ as
deneme.gsm
Then i tried to make a IVR menu and play that file.
I tried
exten=s,4,Playback(/record/deneme.gsm)
Hi
Drop the '.gsm' from the filename.
Ish
Oguzhan Kayhan wrote:
Hello,
I am trying to create a simple IVR for testing..
What i did is to create a voice file from asterisk-gui.
And i saw it created that under /var/lib/asterisk/sounds/record/ as
deneme.gsm
Then i tried to make a IVR menu
hello,
you should try it with the following:
exten=s,4,Playback(record/deneme)
the .gsm is not necessary
best regards
steve
Oguzhan Kayhan schrieb:
Hello,
I am trying to create a simple IVR for testing..
What i did is to create a voice file from asterisk-gui.
And i saw it created that
exten=s,4,Playback(/record/deneme.gsm)
should be
exten=s,4,Playback(/record/deneme)
so without a format.
On Tue, 2009-06-23 at 11:31 +0300, Oguzhan Kayhan wrote:
Hello,
I am trying to create a simple IVR for testing..
What i did is to create a voice file from asterisk-gui.
And i saw it
exten=s,4,Playback(/record/deneme.gsm)
should be
exten=s,4,Playback(/record/deneme)
so without a format.
Thank you.
That worked :)
On Tue, 2009-06-23 at 11:31 +0300, Oguzhan Kayhan wrote:
Hello,
I am trying to create a simple IVR for testing..
What i did is to create a voice
On Tue, Jun 23, 2009 at 11:31:54AM +0300, Oguzhan Kayhan wrote:
Hello,
I am trying to create a simple IVR for testing..
What i did is to create a voice file from asterisk-gui.
And i saw it created that under /var/lib/asterisk/sounds/record/ as
deneme.gsm
Then i tried to make a IVR menu and
Hello,
I am trying to do :
Exten =_X.,n,Dial(SIP/ser_sei0/1130,L(10208400:61000:1))
But it return that error:
[Jun 2 10:04:44] WARNING[18920]: app_dial.c:1623 dial_exec_full: Invalid
timeout specified: 'L(10208400:61000:1)'
Why?
I forgot something ?
Thank you
BERGANZ François schrieb:
Exten =_X.,n,Dial(SIP/ser_sei0/1130,L(10208400:61000:1))
But it return that error:
[Jun 2 10:04:44] WARNING[18920]: app_dial.c:1623 dial_exec_full: Invalid
timeout specified: 'L(10208400:61000:1)'
Syntax:
François
Sent: 2009 m. birželio 2 d. 11:07
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] error with dial timeout
Hello,
I am trying to do :
Exten =_X.,n,Dial(SIP/ser_sei0/1130,L(10208400:61000:1))
But it return that error:
[Jun 2 10:04:44] WARNING[18920]: app_dial.c
[mailto:asterisk-users-boun...@lists.digium.com] De la part de Philipp Kempgen
Envoyé : mardi 2 juin 2009 10:37
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [asterisk-users] error with dial timeout
BERGANZ François schrieb:
Exten =_X.,n,Dial(SIP/ser_sei0/1130,L(10208400:61000
hi
i got TOS and retranssmission error on receiving SIP call
chan_sip.c:2794 retrans_pkt: Maximum retries exceeded on transmission
10caed68-0f1d-df82-da1e-a76c1cb3d...@172.18.100.72 for seqno 43156 (Critical
Response) -- See doc/sip-retransmit.txt.
[May 22 13:42:44] WARNING[18021]:
On Fri, 2009-05-22 at 13:57 +0530, DHAVAL INDRODIYA wrote:
i got TOS and retranssmission error on receiving SIP call
chan_sip.c:2794 retrans_pkt: Maximum retries exceeded on transmission
10caed68-0f1d-df82-da1e-a76c1cb3d...@172.18.100.72 for seqno 43156
(Critical Response) -- See
I've seen that message when then endpoint is not available.
Cary Fitch wrote:
[Apr 26 10:47:01] NOTICE[32151]: chan_sip.c:16223 sip_poke_noanswer: Peer
'3516533812' is now UNREACHABLE! Last qualify: 86
[Apr 26 10:47:11] NOTICE[32151]: chan_sip.c:12723 handle_response_peerpoke:
Peer
[Apr 26 10:47:01] NOTICE[32151]: chan_sip.c:16223 sip_poke_noanswer: Peer
'3516533812' is now UNREACHABLE! Last qualify: 86
[Apr 26 10:47:11] NOTICE[32151]: chan_sip.c:12723 handle_response_peerpoke:
Peer '3516533812' is now Reachable. (98ms / 2000ms)
[Apr 26 12:08:49] WARNING[32273]:
Hi David,
Thanks for the answer!
By using the h extension now I'm able to check that the Faxes are sent
successfully.
Best regards,
Santi
On Fri, Mar 27, 2009 at 4:42 PM, David Backeberg dbackeb...@gmail.com wrote:
On Tue, Mar 24, 2009 at 1:57 PM, Santiago Gimeno
santiago.gim...@gmail.com
On Tue, Mar 24, 2009 at 1:57 PM, Santiago Gimeno
santiago.gim...@gmail.com wrote:
Hello,
The NoOp output was not displayed at all. I'm assuming because of the
failure in the ReceiveFax application. In fact, the verbose output
Try changing
[fax-in]
exten =
Hello,
In my scenario, the asterisk machine is installed behind a CISCO
mediaGW in order to be able communicate with the PSTN. Asterisk is
configured to use T.38 to send and receive faxes.
I'm trying to receive a fax from a fax machine located in the PSTN.
Apparently everything goes well: the
On Tue, Mar 24, 2009 at 9:21 AM, Santiago Gimeno
santiago.gim...@gmail.com wrote:
WARNING[12229]: app_fax.c:650 in transmit: Transmission error
and the ReceiveFax function ends abruptly.
That doesn't really help, other than that it seems your arrangement
defaulted to voice rather than using
Sorry about that, I forgot to post them:
-extension.conf:
[fax-in]
exten = 9,1,Set(INCOMING_FAXFILE=/root/santi/fax/incoming.tif)
exten = 9,n,Answer()
exten = 9,n,Wait(3)
exten = 9,n,ReceiveFax(${INCOMING_FAXFILE})
exten = 9,n,NoOp(FAXSTATUS: ${FAXSTATUS}, FAXERROR:
On Tue, Mar 24, 2009 at 11:33 AM, Santiago Gimeno
santiago.gim...@gmail.com wrote:
Sorry about that, I forgot to post them:
That all looks pretty good.
So in your original post, you clipped it off before you got all the
useful no-op output at the end.
I'm also assuming your file was empty?
Hello,
The NoOp output was not displayed at all. I'm assuming because of the
failure in the ReceiveFax application. In fact, the verbose output
was:
== Using SIP RTP CoS mark 5
== Using UDPTL CoS mark 5
-- Executing [99...@demo:1] Set(SIP/192.168.0.253-b7a96b70,
Hello,
I am having problems getting one xlite clients to communicate through
asterisk. I am getting an error message:
chan_sip.c:15593 handle_request_register: Registration from 'chinmay
chakrabortysip:1...@10.44.32.193 sip%3a1...@10.44.32.193' failed
for '10.44.32.193' - No matching peer found
chinmay chakraborty schrieb:
I am having problems getting one xlite clients to communicate through
asterisk. I am getting an error message:
chan_sip.c:15593 handle_request_register: Registration from 'chinmay
chakrabortysip:1...@10.44.32.193 sip%3a1...@10.44.32.193' failed
for '10.44.32.193'
Here you go http://tinyurl.com/a7tkkz
2009/1/12 chinmay chakraborty chinmay.chakrabo...@gmail.com
Hello,
I am having problems getting one xlite clients to communicate through
asterisk. I am getting an error message:
chan_sip.c:15593 handle_request_register: Registration from 'chinmay
I am calling into an asterisk machine using console/dsp
initially I head the voice just fine on the speaker, after about 30
seconds or so
I get the error:
[Dec 31 09:51:36] NOTICE[11724]: chan_alsa.c:642 alsa_write: Error -11
on write
Hangup on console
I am running alsa 1.4.22 , dahdi 2.0.0
On Wednesday 31 December 2008 08:54:56 Jerry Geis wrote:
I am calling into an asterisk machine using console/dsp
initially I head the voice just fine on the speaker, after about 30
seconds or so
I get the error:
[Dec 31 09:51:36] NOTICE[11724]: chan_alsa.c:642 alsa_write: Error -11
on write
Error 11 is Resource temporarily unavailable, so it sounds like something
else is accessing the sound port. Check the other services running on this
machine. For example, are you running a desktop environment that might
send sounds to the speaker?
Tilghman,
I am not even running the
On Wednesday 31 December 2008 10:07:01 Jerry Geis wrote:
Error 11 is Resource temporarily unavailable, so it sounds like
something else is accessing the sound port. Check the other services
running on this machine. For example, are you running a desktop
environment that might send sounds
I spent some time looking at this and I may have identified a possible
problem. Please try the patch located here:
http://bugs.digium.com/view.php?id=14153
Tilghman,
I downloaded the patch. cd'd to my asterisk directory.
tried to do patch -p1 20081231__bug14153.diff.txt
and it says:
Jerry Geis wrote:
I spent some time looking at this and I may have identified a possible
problem. Please try the patch located here:
http://bugs.digium.com/view.php?id=14153
Tilghman,
I downloaded the patch. cd'd to my asterisk directory.
tried to do patch -p1
On Wednesday 31 December 2008 13:06:03 Jerry Geis wrote:
Jerry Geis wrote:
I spent some time looking at this and I may have identified a possible
problem. Please try the patch located here:
http://bugs.digium.com/view.php?id=14153
Tilghman,
I downloaded the patch. cd'd to my
On Wed, Dec 31, 2008 at 01:58:21PM -0500, Jerry Geis wrote:
I spent some time looking at this and I may have identified a possible
problem. Please try the patch located here:
http://bugs.digium.com/view.php?id=14153
Tilghman,
I downloaded the patch. cd'd to my asterisk
Hello,
we are using an EICON/DIALOGIC DIVA Server 4BRI together with asterisk
1.4.21.2 and chan-capi-head (20-11-2008)
From time to time error messages like the following appear several
times in /var/log/asterisk/messages:
ERROR[31152] chan_capi.c: Could not write to pipe for ISDN4#02
Is
Hi. I am getting the following error while trying to compile the
dahdi-tools-trunk from svn this morning.
gcc -g -O2 -I. -O2 -g -fPIC -Wall -DBUILDING_TONEZONE -MD -MT
sethdlc.o -MF .sethdlc.o.d -MP -c -o sethdlc.o sethdlc.c
sethdlc.c: In function 'set_iface':
sethdlc.c:205: error: 'union
On Sun, Sep 07, 2008 at 09:22:57AM -0400, John covici wrote:
Hi. I am getting the following error while trying to compile the
dahdi-tools-trunk from svn this morning.
gcc -g -O2 -I. -O2 -g -fPIC -Wall -DBUILDING_TONEZONE -MD -MT
sethdlc.o -MF .sethdlc.o.d -MP -c -o sethdlc.o sethdlc.c
on Sunday 09/07/2008 Tzafrir Cohen([EMAIL PROTECTED]) wrote
On Sun, Sep 07, 2008 at 09:22:57AM -0400, John covici wrote:
Hi. I am getting the following error while trying to compile the
dahdi-tools-trunk from svn this morning.
gcc -g -O2 -I. -O2 -g -fPIC -Wall -DBUILDING_TONEZONE
I'm just wondering if anyone has seen this/fixed this before.
I'm trying to compile Zaptel 1.4.11 in Sabayon Linux. It keeps failing out
when I attempt to make after running configure. It looks like it might be a
problem with the kernel sources, but they are there and the symlink is pointing
Christopher Hoff wrote:
I'm trying to compile Zaptel 1.4.11 in Sabayon Linux. It keeps failing out
when I attempt to make after running configure. It looks like it might be
a problem with the kernel sources, but they are there and the symlink is
pointing to the correct directory. Does
Hi,
I got some errors about not being able to create subdir [already
existing] on a 'make update' in my zaptel 1.4.
I removed the directory and did a new svn co of zaptel 1.4
[ svn co http://svn.digium.com/svn/zaptel/branches/1.4 zaptel ]
now I get:
/usr/bin/install -c -D -m 644
Hi,
I got some errors about not being able to create subdir [already
existing] on a 'make update' in my zaptel 1.4.
I removed the directory and did a new svn co of zaptel 1.4
[ svn co http://svn.digium.com/svn/zaptel/branches/1.4 zaptel ]
now I get:
/usr/bin/install -c -D -m
Hello All,
This is my third freshly installed and updated CentOS 5.1 with installed
Digium 4-port Analog card and while compiling Zaptel I am getting this
error. If I run ./install_preq test and ./install_preq install it
says Install Successfully.
Error
=
CC [M]
On Wed, Jun 18, 2008 at 02:02:28PM -0700, Robert McNaught wrote:
Trying to install zaptel-1.4.11 on a Supermicro SuperServer with Centos
5, and getting the following error trail on make. Googling the issue
has found one user who tried:
seems that commenting out typedef int bool; in
On Wed, Jun 18, 2008 at 02:02:28PM -0700, Robert McNaught wrote:
Trying to install zaptel-1.4.11 on a Supermicro SuperServer with Centos
5, and getting the following error trail on make. Googling the issue
has found one user who tried:
seems that commenting out typedef int bool; in
Robert McNaught wrote:
Hi All,
Trying to install zaptel-1.4.11 on a Supermicro SuperServer with Centos
5, and getting the following error trail on make. Googling the issue
has found one user who tried:
seems that commenting out typedef int bool; in xpp/xdefs.h on line 93
works
that out,
Hi All,
I am getting following error when i start AsterFax:
Please help me to solve this issue:
[EMAIL PROTECTED] asterfax]# ./asterfax.sh
log4j: Threshold =null.
log4j: Retreiving an instance of org.apache.log4j.Logger.
log4j: Setting [au.com.noojee.asterfax] additivity to
You really should discuss this at the Asterfax forums:
http://forums.asteriskit.com.au/
later,
PaulH
On Wed, 2008-06-04 at 14:17 +0530, Sukhbir Singh wrote:
Hi All,
I am getting following error when i start AsterFax:
Please help me to solve this issue:
[EMAIL
scenario is incoming calls on ZAP (TE410p euro isdn) to SIP (or any
other channel) and call is answered.
When I hangup on the ISDN side on the 1.2 then the SIP hangs up to
immidiatly so everything is fine (se short pri debug below).
When I do the same on 1.4.20 then it take more than 30
Hi I better answer my own post.
I went to the code and the issue is in q931.c
/* wait for a RELEASE so that sufficient time has passed
for the inband audio to be heard */
if (c-progressmask PRI_PROG_INBAND_AVAILABLE)
break;
Changing this line
On Wed, May 21, 2008 at 04:09:33PM -0500, Lyle Giese wrote:
Around here, the telco locks that cabinet and there is a user id/password
required to use that craft interface. (I have a key, but then I worked for
the telco for 23 years and have a few insider tools laying around)
They are
On Wed, May 21, 2008 at 05:20:27PM -0400, Alex Balashov wrote:
Not only that, but Jay's suggestion is going to give you the performance
monitoring data off the smartjack itself, not off of any portion of the
span beyond the smartjack.
So, if you are receiving errors on the part that runs
On Tue, May 20, 2008 at 07:03:06PM -0500, Lyle Giese wrote:
Is there a way to see error counts on the T1 of a PRI? Hooked up to
asterisk via a digium TE122. Looking for something to make sure I'm not
getting any CRC, framing or other errors on the T1.
Go on ebay and buy an
Hello,
In Asterisk you can type zap show status to at least show you some
basic error information:
CLI zap show status
Description Alarms IRQbpviol CRC4
wanpipe1 card 0 OK 0 0 0
wanpipe2 card 1
Joe Pukepail wrote:
Is there a way to see error counts on the T1 of a PRI?
Hooked up to asterisk via a digium TE122. Looking for
something to make sure I'm not getting any CRC, framing or
other errors on the T1.
Many moons ago I use to used a program called zttool. It
Don Pobanz wrote:
Joe Pukepail wrote:
Is there a way to see error counts on the T1 of a PRI?
Hooked up to asterisk via a digium TE122. Looking for
something to make sure I'm not getting any CRC, framing or
other errors on the T1.
Many moons ago I use to used a
Jay R. Ashworth wrote:
On Tue, May 20, 2008 at 07:03:06PM -0500, Lyle Giese wrote:
Is there a way to see error counts on the T1 of a PRI? Hooked up to
asterisk via a digium TE122. Looking for something to make sure I'm not
getting any CRC, framing or other errors on the T1.
Lyle Giese wrote:
Jay R. Ashworth wrote:
On Tue, May 20, 2008 at 07:03:06PM -0500, Lyle Giese wrote:
Is there a way to see error counts on the T1 of a PRI? Hooked up to
asterisk via a digium TE122. Looking for something to make sure I'm
not
getting any CRC, framing or other
Is there a way to see error counts on the T1 of a PRI? Hooked up to
asterisk via a digium TE122. Looking for something to make sure I'm not
getting any CRC, framing or other errors on the T1.
Using asterisk 1.4.19 and zaptel 1.4.10
___
-- Bandwidth
Joe Pukepail wrote:
Is there a way to see error counts on the T1 of a PRI? Hooked up to
asterisk via a digium TE122. Looking for something to make sure I'm
not getting any CRC, framing or other errors on the T1.
Using asterisk 1.4.19 and zaptel 1.4.10
Go on ebay and buy an ADC Kentrox
Using Asterisk 1.2, still.
We are issuing a callback. User rejects the first two calls, but answers the
third. For some reason, the Manager Interface outputs a CDR with disposition
'NO ANSWER' for all three attempts, eventhough the 3rd call worked.
Is this an asterisk 1.2 bug?
Doug.
Hi there dear users and dear developers of Asterisk!
I've got a maybe strange idea, let's see if you laugh or think it's reasonable J
I'm using Asterisk with Digium TDM800P cards with 24 lines (as an answering
machine).
Each analog line can be reached through a phonenumber, since they are each
On Thu, Feb 14, 2008 at 01:17:45PM +0100, Johan Sandgren wrote:
Hi there dear users and dear developers of Asterisk!
I've got a maybe strange idea, let's see if you laugh or think it's
reasonable J
I'm using Asterisk with Digium TDM800P cards with 24 lines (as an answering
machine).
hi,
When i load module under my linux system
debian OS kernel 2.6.18.-6-xen-amd64
i get the folloeing error:
Can You give me any hint what can be wrong?
Know my asterisk server generates no sounds in applications such as echo or
voicemail
apata Telephony Interface Registered on major 196
Zaptel
Hi,
I encountered the following problem:
My asterisk works as a gateway between two sip networks external (public
internet) and internal (local lan)
From public side asterisk is registered UA in external network.
Internal sip UAs are registered in local SIP Proxy.
When there is an incoming call
Hi, my name is Cristian, i am Argentina.
I Have asterisk 1.4.11 with libs and patchs for unicall from
http://www.moythreads.com/astunicall/. I work with mfcr2 and my configuration
is:
zaptel.conf
span=1,1,0,cas,hdb3
cas=1-15:1101
cas=17-31:1101
loadzone=us
defaultzone=us
unicall.conf
Try playing with the options.
protocolvariant=ar,10,4,7
And please post debug output of unicall.
unicall.conf
loglevel=255
- Moy
On Nov 21, 2007 2:04 PM, [EMAIL PROTECTED] wrote:
Hi, my name is Cristian, i am Argentina.
I Have asterisk 1.4.11 with libs and patchs for unicall from
My calls provider has suspended my account, because he says that I am
send bad formating call strings.
According to the email he sent me a line return is beign inserted
after the number.
One string he sent me is the following:
BADCALL,101339,0115712550727
,reseller,cesar reategui
I have Asterisk 1.2.13 installed as a Debian package and I edit only
sip.conf and extensions.conf in this way:
sip.conf:
[general]
realm=work.com.ar ; Realm for digest
authentication
bindport=5060 ; UDP Port to bind to (SIP standard port
is
This line gives you the clue:
Oct 9 12:52:41 WARNING[3478]: app_dial.c:1024 dial_exec_full: Dial
argument takes format (technology/[device:]number1
Your dialplan should have Dial(SIP/user1) rather than Dial
(SIP,user1) / instead of ,
Give that a try.
--
Aubrey Wells
Senior
Ciao Jody,
hi there :)
i get this error in the asterisk CLI:
Sep 21 20:31:49 ERROR[3774]: cdr_addon_mysql.c:159 mysql_log: cdr_mysql:
Unknown connection error:
(2006) MySQL server has gone away
when I run 'cdr mysql status' i get:
Connected to [EMAIL PROTECTED], port 3306 using table
hi there :)
i get this error in the asterisk CLI:
Sep 21 20:31:49 ERROR[3774]: cdr_addon_mysql.c:159 mysql_log: cdr_mysql:
Unknown connection error:
(2006) MySQL server has gone away
when I run 'cdr mysql status' i get:
Connected to [EMAIL PROTECTED], port 3306 using table cdr for 18 minutes, 24
It would surprise me to see mySQL configured incorrectly, but it's always a
possibility. Look at the mysql server var called 'wait_timeout'.
phpMyAdmin shows it under system vars.
hi,
I am using debian 4.0 with version 2.6.18-4-686
I have downloaded the required files form site
asterisk-1.2.24.tar.gz
libmfcr2-0.0.3-1.4.tar.bz2
libsupertone-0.0.2.tar.gz
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