Hi, is anyone else having problems with the reload command crashing
Asterisk 1.6.2.19? I've run a few tests and 1.6.2.18.2 doesn't have
this problem but 1.6.2.19 after a few reloads is just dumping and
restarting.
Thanks
Lee
--
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Seg Faults with 1.6.2.19
On 18 July 2011 12:03, Lee Archer lee.arc...@thebigword.com wrote:
Hi, is anyone else having problems with the reload command crashing
Asterisk 1.6.2.19? I've run a few tests and 1.6.2.18.2
: [asterisk-users] Seg Faults with 1.6.2.19
Sent from my Computer
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee
Archer
Sent: Monday, July 18, 2011 7:04 AM
To: asterisk-users@lists.digium.com
Subject
] On Behalf Of Lee Archer
Sent: 18 July 2011 14:01
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Seg Faults with 1.6.2.19
Hi Eric, are you using ODBC?
Regards
Lee
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk
Hi Kevin, the ticket below was closed as it doesn't happen with 1.8. It
can't be related to my ODBC connections if others are having it. Should
a new ticket be opened?
Regards
Lee
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Hi, can anyone help with this?
Thanks
Lee
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee Archer
Sent: 05 July 2011 16:27
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Recording SIP history
Hi all, can
Hi all, can someone explain what siphistory is supposed to do as it
appears to record nothing to my log files. When I sip show history
callid it's fine but it's not logging anything. My logger.conf has
debug = debug and the debug file grows. Is my understanding correct in
that at the end of the
Hi, I installed the Vestec module to one of my development Asterisk
servers a few months ago but now I need to move the license to another
host. Does anyone know how to do this? I've had a look on my Account
page on the Digium website but it only shows the Language Pack, and I
can't do anything
Hi, does QUEUE_PRIO work the Queues and Asterisk 1.6.2? I've found some
documentation on Google but it looks like it's old Asterisk and not
current.
Thanks
Lee
--
_
-- Bandwidth and Colocation Provided by
Hi, try unloading res_timing_dahdi.so then trying again.
Lee
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Davies
Sent: 07 December 2010 12:54
To: Asterisk Users Mailing List - Non-Commercial
Hi, I am using Asterisk 1.6.2.13 and have an issue but I'm not sure if
it's a bug or not. I am using the cdr_adaptive_odbc logging module and
writing my CDR records to a MS-SQL server. I need to log which end
hangs the call up and have placed the relevant
CDR(myfield)=caller/callee commands
-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Edwards
Sent: 26 August 2010 21:02
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Use of AGISIGHUP
On Thu, 26 Aug 2010, Lee Archer wrote
Hi, I am setting AGISIGHUP=no before running a Perl script via AGI but
it doesn't seem to be doing anything as the script is still exiting on a
hangup and not completing properly. I am using 1.4.35 and have tried
various combinations. Can anyone shed any light on this?
Regards
Lee
--
Should I log this as a bug since it doesn't work?
Regards
Lee
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee Archer
Sent: 20 May 2010 16:28
To: Asterisk Users Mailing List - Non-Commercial Discussion
Try a Cisco ASA. It will rewrite the headers if configured properly.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Motiejus Jakštys
Sent: 26 May 2010 14:17
To: Asterisk Users Mailing List - Non-Commercial
Failed to add '1234,1,NoOp,hello' extension into 'test' context
Regards
Lee
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee Archer
Sent: 19 May 2010 16:54
To: Asterisk Users Mailing List - Non-Commercial
Hi, anyone know?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee Archer
Sent: 17 May 2010 11:29
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Adding a context from the console
Hi
] Adding a context from the console
On Wednesday 19 May 2010 02:28:02 Lee Archer wrote:
Hi, is it possible to add a context from the console using the
dialplan
command?
Yes, just add an extension to it. The context will be created as
needed.
--
Tilghman Lesher
Digium, Inc. | Senior Software
Hi, is it possible to add a context from the console using the dialplan
command?
Thanks
Lee
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar
Hi, I wonder if anyone can help me with a macro issue I have. I need to
set a variable which tells me whether a call has been authenticated
properly. However this authentication is taking place inside of a macro
and I don't want to use a global variable if it will apply to other
channels. I've
- Non-Commercial Discussion
Subject: Re: [asterisk-users] Records sets and ODBC
On Monday 10 May 2010 07:19:34 Lee Archer wrote:
Hi, I have a system using ODBC and connecting to a MS-SQL database.
Does anyone know if it is possible to return a record set consisting
of
several rows from SQL back
at
that location.
Also just F will continue to the next priority on the dialplan.
De: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] En nombre de Lee Archer
Enviado el: lunes, 10 de mayo de 2010 9:36
Para: asterisk-users@lists.digium.com
Asunto: [asterisk
Hi, I have a system using ODBC and connecting to a MS-SQL database.
Does anyone know if it is possible to return a record set consisting of
several rows from SQL back into Asterisk? I have tried using ARRAY but
only the contents of the last row are being stored.
Thanks
Lee
--
Hi, does anyone know if there is an equivalent dial option for the
source channel to the g option? I've had a good look and can't find
one.
g- Proceed with dialplan execution at the current extension if the
destination channel hangs up.
Thanks
Lee
--
.
De: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] En nombre de Lee Archer
Enviado el: lunes, 10 de mayo de 2010 9:36
Para: asterisk-users@lists.digium.com
Asunto: [asterisk-users] Continue dialplan is source channel hangs up
Hi, does anyone
help with a step-by-step?
Thx
Sanjay
On Mon, Mar 15, 2010 at 3:08 PM, Lee Archer lee.arc...@thebigword.com
wrote:
Isn't the use of DNID separate to the userfield? I'd like to have this
working also.
Lee
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Isn't the use of DNID separate to the userfield? I'd like to have this
working also.
Lee
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex
Balashov
Sent: 15 March 2010 08:34
To: Asterisk Users Mailing List
Hi, does anyone run non HWEC Sangoma PRI cards with an increased dahdi
chunk size? I tested it at 2ms and it seemed fine with no noticeable
loss in audio quality, and it reduced the interrupt processing to 50%.
Regards
Lee
--
Hi, does anyone have an info into what could cause
[Nov 28 14:24:48] ERROR[11964] utils.c: write() returned error: Broken
pipe
[Nov 28 14:25:08] ERROR[12540] utils.c: write() returned error: Broken
pipe
[Nov 28 14:25:08] ERROR[12540] utils.c: write() returned error: Broken
pipe
[Nov 28 14:26:23]
I use CentOS, and it works fairly well. But I had to piece together info from
several places. I've tried it several different wants and this way worked, as
long as asterisk is run as root.
Copy asterisk-mib.txt and digium-mib.txt from asterisk_source/doc to
/usr/share/snmp/mibs/
If you are want CDR's to go to MS-SQL try cdr_tds.
Regards
Lee
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Neeraj
Chand
Sent: 06 November 2009 07:04
To: asterisk-users@lists.digium.com
Subject: Re:
Hi, is userfield the only extra CDR field that can be added or can
others?
Regards
Lee
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
-boun...@lists.digium.com] On Behalf Of Lee Archer
Sent: Tuesday, November 03, 2009 10:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Extra CDR fields
Hi, is userfield the only extra CDR field that can be added or can
others?
Regards
Lee
...@lists.digium.com] On Behalf Of Carlos Chavez
Sent: 03 November 2009 16:42
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Extra CDR fields
On Tue, 2009-11-03 at 16:09 +, Lee Archer wrote:
Hi, is userfield the only extra CDR field that can be added or can
] On Behalf Of Lee Archer
Sent: Tuesday, November 03, 2009 10:42 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Extra CDR fields
Do you have any info on multiple userfields as that's exactly what I
would be looking for?
Regards
Lee
From
Zenoss has something that hits the manager port. I run Asterisk 1.4 boxes and
are using SNMP to monitor. Asterisk 1.6 has a couple of extra SNMP OID’s that
show the number of calls processed. It’s a shame 1.4 doesn’t have this OID as
it could be really useful.
Regards
Lee
From:
SPA921 isn't an Aastra phone though is it? I would expect the Linksys
manual to list some of the ones you can use.
Regards
Lee
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle
Dupuis
Sent: 19 October 2009 01:07
To:
For anyone who is interested I've recently created an Aastra IP Phone
config generator. I don't know if one existed but thought I'd create it
anyways. It can be found at http://www.lraweb.pwp.blueyonder.co.uk/.
If you have any problems or stuff you want adding then please contact at
the address
I have a system that has had 5 G729 licenses for over a year and I've
come to install the v31 G729 codec from the Digium ftp server but it
won't see the license. Does anyone know how to get around this problem?
It is registered and I do have newer systems running this v31 version of
the codec but
, Apr 25, 2007 at 08:57:37AM +0100, Lee Archer wrote:
I installed zaptel 1.2.17 and shortly afterwards got a problem of
calls not clearing properly. I ran dmesg which showed
Unable to handle kernel NULL pointer dereference at virtual
address 009c
printing eip:
f8a79fa8
I installed zaptel 1.2.17 and shortly afterwards got a problem of calls not
clearing properly. I ran dmesg which showed
Unable to handle kernel NULL pointer dereference at virtual address
009c
printing eip:
f8a79fa8
*pde =
Oops: [#1]
I used mpg123 to stream air traffic control as a MOH class but I also
found it didn't always work with the shoutcast servers.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
Germann
Sent: 27 February 2007 02:17
To: 'Asterisk Users Mailing List -
Yes check the freepbx website, and in particular trac bug #1610.
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of younss
azzayani
Sent: 16 February 2007 11:23
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users]
I said what to do before.
http://freepbx.org/trac/ticket/1610
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Guillermo
Salas M.
Sent: 16 February 2007 14:51
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users]
I had this problem and in the end it appeared to be slot timing on the mobo. I
had to put the TE110P in the 1st slot - which happened to be a PCI-X slot.
That was using a Supermicro motherboard too.
Regards
Lee
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On
Aren't Aastra due to release new phones and some form of
operator/reception addon? The Aastra user/admin guides are a lot more
up2date that they used to be. I'd like Aastra to add a GSM codec to
their phone and have a more regular firmware release schedule. I agree
with the list below though
Have you tried the #freepbx IRC channel or the freepbx mailing list?
Regards
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Arnilo S. Baluyos (Mailing Lists)
Sent: 23 January 2007 01:57
To: Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [asterisk-users] chan_zap.c: Failed to read gains:
Invalidargument
Lee Archer wrote:
I am running Asterisk 1.2.14 and Zaptel 1.2.12 and using a Digium
TE110P card in E1 mode. I've recently noticed in my logs the
following
Jan 5 01:27:11 VERBOSE[22490] logger.c
] chan_zap.c: Failed to read gains:
Invalidargument
On Fri, Jan 05, 2007 at 07:47:15AM -, Lee Archer wrote:
I am running Asterisk 1.2.14 and Zaptel 1.2.12 and using a Digium
TE110P card in E1 mode. I've recently noticed in my logs the
following
Jan 5 01:27:11 VERBOSE[22490] logger.c
at 10:53:17AM +0200, Tzafrir Cohen wrote:
On Fri, Jan 05, 2007 at 07:47:15AM -, Lee Archer wrote:
I am running Asterisk 1.2.14 and Zaptel 1.2.12 and using a Digium
TE110P card in E1 mode. I've recently noticed in my logs the
following
Jan 5 01:27:11 VERBOSE[22490] logger.c
I am running Asterisk 1.2.14 and Zaptel 1.2.12 and using a Digium TE110P
card in E1 mode. I've recently noticed in my logs the following
Jan 5 01:27:11 VERBOSE[22490] logger.c: [chan_zap.so]Jan 5 01:27:11
VERBOSE[22490] logger.c: [chan_zap.so] = (Zapata Telephony w/PRI)
Jan 5 01:27:11
I wonder if anyone can help me with this. I have 4 sites running
Asterisk and these are linked via IAX trunks and ADSL lines. Calls
coming into any of these sites are received locally and forwarded to a
central operator. E.g. Call comes in on site A and is forwarded to
the operator on
I wonder if anyone can help me with this. I have 4 sites running
asterisk and calls coming into any of these sites are received locally
and forwarded to a central operator. E.g. Call comes in on site A and
is forwarded to the operator on site B. 99/100 the operator will send
the call back to
Are you using freePBX by any chance?
Regards
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nigel
Roberts
Sent: 08 November 2006 08:55
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Queue forks asterisk and then leaves the
Discussion
Subject: Re: [asterisk-users] Queue forks asterisk and then leaves
theextraprocesses lying around
Hi Lee,
On Wed, 08 Nov 2006 at 09:00:27 -, Lee Archer wrote:
Are you using freePBX by any chance?
Yes, version 2.1.1.
___
--Bandwidth
Title: Manager interface
This has probably been discussed before but I need to do a screen pop and I'm looking for ways to do it. I am assuming I need to use the manager interface, which is ok cos I'm using that for calling out but I'm not quite what to pick up on.
Regards
Lee
http://bugs.digium.com/view.php?id=7536
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: 17 July 2006 15:25
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] can no more compile zaptel !!!
Hi all,
I was refreshing
Try make on its own and read what it says. You probably want make linux
Regards
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Marc
Rohlfing
Sent: 13 June 2006 12:09
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Compiling mpg123
Title: Duplicate asterisk processes
I'm still getting duplicate process but the results of gdb are different. Can anyone shed any light onto what is causing this?
(gdb) info threads
1 Thread 1091845040 (LWP 31287) 0xe410 in __kernel_vsyscall ()
(gdb) thread apply all bt
Thread 1
- Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Multiple processes
Temporarily turn off your ODBC CDR stuff and see if the problem is still
there.
Lee Archer wrote:
Can someone shed any light on the following. I have 2 identical
systems, 1 of which seems to spawn multiple processes
Title: Multiple processes
Can someone shed any light on the following. I have 2 identical systems, 1 of which seems to spawn multiple processes which have to be killed manually. It recently kicked up 2 so I ran gdb on them and this is the thread output. I current use FreePBX with these
more than some of the threads Asterisk needs for other services.
If you see as output nptl-version
then I think you should see only one Asterisk process.
Regards
On 5/31/06, Lee Archer [EMAIL PROTECTED] wrote:
Can someone shed any light on the following. I have 2 identical
systems, 1
the associated startup scripts, multiple processes would still
be spawned even if not appropriate. Anthony
On 5/31/06, Lee
Archer [EMAIL PROTECTED]
wrote:
Can someone shed any light on the following.
I have 2 identical systems, 1 of which seems to spawn multiple processes which
have
Can't you use mpg123 as compiled under x86_32? I do on a few servers I
have. I found madplay better process wise than mpg123.
Regards
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Erick
Perez
Sent: 29 May 2006 21:37
To: Asterisk Users Mailing
:29
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] mpg123 or asterisk
Can MAD crash a server like mpg123 can?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lee Archer
Sent: Tuesday, 30 May 2006 5:06 PM
Discussion
Subject: Re: [Asterisk-Users] CallerID
It appears that the PBX sitting between Asterisk and your provider is
not passing on the calling pres flags.
On 5/23/06, Lee Archer [EMAIL PROTECTED] wrote:
I have a problem with BT in the UK. Using setcallerpres I can change
the number shown
Stopping and restarting Asterisk will lose the hints, then
you will have to wait until the phone registers again. With 1.2.7.1 a
reload shouldn't lose anything. Change the register time on the phones to
something less that 60 minutes if it's a big problem. Instead of factory
defaulting the
Are the GXP's configured properly for BLF and
whatdoes show hints print?
Lee
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
asteriskSent: 24 May 2006 11:57To: 'Asterisk Users
Mailing List - Non-Commercial Discussion'Subject: RE:
[Asterisk-Users] GXP2k and BLF problem
I run 1.1.0.13 on my GXP's and after stopping and starting
the server I either wait for the phones to re-reg or I reboot the phones.
After restarting asterisk does rebooting the phones does fix the
problem?
Lee
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
asteriskSent:
Are you using preconfiged scripts? If so what happens
if you manually config the phone then restart asterisk and then the
phone?
Lee
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
asteriskSent: 24 May 2006 12:56To: 'Asterisk Users
Mailing List - Non-Commercial
Think you need to contact Grandstream support then.
I've got the same version of * and GXP fw and I get no problems. Sorry I
can't help you any further.
Lee
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
asteriskSent: 24 May 2006 13:30To: 'Asterisk Users
Mailing List -
I have a problem with BT in the UK. Using setcallerpres I can change
the number shown on the recipents phones to Private or unknown but no
matter what I set my asterisk cid and callerpres to it still displays
the base number of my PRI ddi range.
Regards
Lee
-Original Message-
From:
Title: Future pickup feature
Can anyone say whether call pickup with the ability to transfer the callers details is going to be part of any Asterisk release? I'd like to pick up calls but also know roughly who it is I'm talking.
Regards
Lee
Discussion
Subject: Re: [Asterisk-Users] Asterisk with SuSe 10
On 1/24/06, Lee Archer [EMAIL PROTECTED] wrote:
Thanks, I've got it running on my test box but didn't know if there
was any global objection to using it. I've had a few funnies with it
but that might be down to Supermicro and P4's
Any thoughts as to why only 1 of my boxes has this problem? I'm on a
2.6 kernel so any more ideas?
Regards
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dave
Cotton
Sent: 18 April 2006 09:00
To: Asterisk Users Mailing List - Non-Commercial
2006 09:27
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] multiple asterisk process ?
On Tue, 2006-04-18 at 09:13 +0100, Lee Archer wrote:
Any thoughts as to why only 1 of my boxes has this problem?
Is it really a problem?
I'm on a
2.6 kernel so any
PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dave
Cotton
Sent: 18 April 2006 10:02
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] multiple asterisk process ?
On Tue, 2006-04-18 at 09:33 +0100, Lee Archer wrote:
Yes it is a problem cos after a while
I had this and no one could really answer it. I only get it 1 of my
systems. I've tried a few things, from removing zaptel watchdog - since
I contacted the telco and they said I had a hung channel, to rebuilding
* with different options. Are you configuring * manually or using a
GUI?
Lee
When you find out what's causing it can you let me know as I have 1
system that gets this error and the telco tells me everything is fine
with their equipment.
Regards
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Pimjai
Wesnarat
Sent: 11 April
I found progressinband=no in sip.conf fixed my problem when I had this.
Regards
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of kevin ling
Sent: 10 April 2006 12:24
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE:
PRI fault?
On Tuesday 04 April 2006 10:39, Lee Archer wrote:
I've been looking through the logs of a system trying to figure out
why it sometimes starts extra asterisk processes. In the logs I keep
seeing
Define starts extra asterisk processes.
Apr 4 15:22:18 WARNING[5054] chan_zap.c: Can't
Title: Possible PRI fault?
I've been looking through the logs of a system trying to figure out why it sometimes starts extra asterisk processes. In the logs I keep seeing
Apr 4 15:22:18 WARNING[5054] chan_zap.c: Can't fix up channel from 1 to 2 because 2 is already in use
Apr 4 15:22:18
What's the spec of the box?
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Roth
Sent: 03 April 2006 18:48
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] 1.2.6 doesn't use mpg123?
Matt wrote:
Ok..
I run suse 10 and have an X100p. But I use fxsks=1 in the /etc/zaptel.conf not
/etc/asterisk/zaptel.conf.
Lee
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ali asma
Sent: 04 April 2006 10:13
To: Asterisk Users Mailing List - Non-Commercial Discussion
Discussion
Subject: RE: [Asterisk-Users] Loading module chan_zap.so failed! PLZ help me!
I modified the configuration but I still have the same error.
Please tell me in whach directory should I execute:
modprobe zaptel
modprobe wcfxo
becose it seems that my card not had been detected
Thanks,
--- Lee
] On Behalf Of ali asma
Sent: 04 April 2006 10:45
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Loading module chan_zap.so failed! PLZ help me!
yes I make it but I still have the same error
--- Lee Archer [EMAIL PROTECTED] a écrit :
Just modprobe wcfxo
failed! PLZ help me!
Sorry, now I have this:
linux:~ # ztcfg -vv
Zaptel Configuration
==
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
1 channels configured.
But the same error when running asterisk
--- Lee Archer [EMAIL PROTECTED] a écrit :
Just
Of ali asma
Sent: 04 April 2006 11:05
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Loading module chan_zap.so failed! PLZ help me!
ztcfg is ok, but asterisk still can't load chan_zap.so module
--- Lee Archer [EMAIL PROTECTED] a écrit :
Try signalling
files?
Especially since it has to start a seperate stream for every on hold
person? Seems like in a busy call center.. it would be more efficient
to have 1 stream going to every caller, rather then multiple streams.
On 4/1/06, Lee Archer [EMAIL PROTECTED] wrote:
Check the musiconhold.conf.sample
Asterisk playing the sound files?
Especially since it has to start a seperate stream for every on hold
person? Seems like in a busy call center.. it would be more efficient
to have 1 stream going to every caller, rather then multiple streams.
On 4/1/06, Lee Archer [EMAIL PROTECTED] wrote
I use mpg123 for streaming but I can't get it to compile under SuSe10
and x86_64 CPU. Does anyone have any recommendations for other programs
that allow streaming and will compile on this arch?
Regards
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
given
channel.
[]'s
MM
-Original Message-
From: Lee Archer [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Cc:
Sent: Sat, 1 Apr 2006 10:34:42 +0100
Delivered: Sat, 01 Apr 2006 06:28:16
Subject:[Asterisk-Users] 1.2.6
and you'll have a nice native streaming. You can convert your stuff to
another formats, like sox file.mp3 [-c1] file.gsm or sox file.mp3
[-c1] file.ul and let asterisk decide which one best fits given
channel.
[]'s
MM
-Original Message-
From: Lee Archer [EMAIL PROTECTED
Subject: Re: Re[2]: [Asterisk-Users] 1.2.6 doesn't use mpg123?
How did you switch from native to mpg123 on 1.2.x? That's what I can't
figure out.
On 4/1/06, Lee Archer [EMAIL PROTECTED] wrote:
Has anyone else had a problem with asterisk creating multiple threads?
I'm still testing but I've
Title: Multiple processes
Does anyone have any ideas why my recently updated * 1.2.5 system should spawn multiple * process at seemingly random intervals?
Regards
L:ee
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Could be the same problem I had with my Aastra - progressinband=no
worked for me.
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Julian
Lyndon-Smith
Sent: 15 March 2006 18:10
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Discussion
Subject: Re: [Asterisk-Users] Double-ring tone
That's in the [general] section of sip.conf, yes ?
How would that affect the 7.4 phones ? Wouldn't want to annoy them ;)
Julian.
Lee Archer wrote:
Could be the same problem I had with my Aastra - progressinband=no
worked for me.
Lee
Hi try http://www.grandstream.com/y-downloads.htm
Download the IP Phone Custom Ringtones Generation Tool
Unzip and read the readme
Regards
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dmitry
Ivanov
Sent: 07 March 2006 13:40
To:
07 March 2006 15:49, Lee Archer wrote:
Download the IP Phone Custom Ringtones Generation Tool Unzip and read
the readme
Ringtone != dialtone.
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Title: HDLC error
Can anyone help and point me in a useful direction. I'm using * 1.2.4 with Zaptel 1.2.4. I have a TE110P card and its a Supermicro P8SCT mobo. If I run the PRI card in the PCI-X slot it shares an interupt with eth0 but I don't get problems. I've been trying to move it onto
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