Odd you should have this problem as I had exactly the same. In my case
it was a slow DHCP server. Around 7 seconds in the phones tries to time
sync. If the phone hasn't got an IP address then this time sync fails
but it doesn't retry. I emailed Grandstream about it but got nowhere.
I changed
Title: Asterisk with SuSe 10
Has anyone had any experience with the Asterisk on a SuSe 10 platform? I'm currently using FC3 but because we use SuSe within other parts of the business I'm being pushed to changed the OS.
Regards
Lee
###This message
] On Behalf Of Ben Klang
Sent: 24 January 2006 15:49
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk with SuSe 10
On Tuesday 24 January 2006 09:26, Lee Archer wrote:
Has anyone had any experience with the Asterisk on a SuSe 10 platform?
I'm
I had a problem with the scripts you can bulk generate, they are linked
to the MAC address you initially put in, so if the phone packs in you
can't just rename the file.
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Phil
Blundell
Sent: 30 January
Title: Streaming MOH
Hi, I'm having some problems getting this to work with Asterisk 1.2.4. Does it work for anyone? Does anyone have a site I can test this with?
Regards
Lee
___
--Bandwidth and Colocation provided by Easynews.com --
I have this problem in the UK on BT too.
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Garth van
Sittert
Sent: 02 February 2006 11:37
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Outbound Caller ID number
I've got it working now but the playback through the handset is
sloow. I can tell it's music but you couldn't sing along to it...
Still maybe it's about the right speed for a hangover.
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Title: Musiconhold in zapata.conf
I've been trying to change the musiconhold= in the zapata.conf to use something other than default. However it doesn't seem to do it. I know the other musiconhold source works but whatever I set it to in the zapata.conf file it always plays whatever is
Title: Double ring
Can anyone shed any light on to why I get a double ring when calling external numbers? When calling out I hear the actually ring-ring of the called phone and the asterisk ring tone. I'm using the same config I used with 1.0.10 but have now upgraded to 1.2.4.
Regards
Lee
February 2006 12:27
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Double ring
I was getting something very similar with my Aastra test phones until I
change 'callprogress=' to 'no'.
Thanks,
Bob
On Fri, 10 Feb 2006 12:13:47 -
Lee Archer [EMAIL
: [Asterisk-Users] Double ring
'callprogress', in zapata.conf:
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zapata.conf
Thanks,
Bob McDowell
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lee Archer
Sent: Friday, February 10, 2006 7:04 AM
Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lee Archer
Sent: 10 February 2006 13:50
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Double ring
Sorry I meant callprogress. I've tried it set to yes and no with no
difference
Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Double ring
I have the similar problem with thomson sip voip cable modems:
http://bugs.digium.com/view.php?id=6083
On Fri, 2006-02-10 at 12:13 +, Lee Archer wrote:
Can anyone shed any light on to why I get a double ring
Subject: Re: [Asterisk-Users] Double ring
I noticed the issue today and came looking for confirmation when I came
upon this thread. My Grandstream does not have this problem.
SPA-941, Snom 320 and Aastra 480i all demonstrate this issue.
I'm going to
Lee Archer wrote:
Am I the only one
Title: Firmware version 1.3.1 released for Aastra IP phones
There is no release note, just a text file that says
AASTRA TELECOM INC.
February 2006
FC-46-01-07.st - 9133i Generic SIP Firmware 1.3.1.1095
for customer release.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
-16 at 13:28 +, Lee Archer wrote:
There is no release note, just a text file that says
AASTRA TELECOM INC.
February 2006
FC-46-01-07.st - 9133i Generic SIP Firmware 1.3.1.1095 for
customer release.
http://www.aastra.com/support/show_manuals.asp?p=241
Worked for me
Title: Firmware version 1.3.1 released for Aastra IP phones
Any chance of getting a config option in that allows you
set headset/speaker in the audio menu?
Lee
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gareth
OwenSent: 15 February 2006 02:00To:
/headset
3 = headset/speaker
Gareth
Lee Archer wrote:
Any chance of getting a config option in that allows you set
headset/speaker in the audio menu?
Lee
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
Worth saying that the Aastra 9133i with the 1.3.1 firmware is a pretty
good phone. I used to run GXP-2000's, still have 10 new in a box and
another 20 in demo/test circulation, but I also run a few dozen 9133i,
480i and 9112i phones and I think Aastra are getting their now. Biggest
problem I had
all the config
options.
Gareth
-Original Message-
From: [EMAIL PROTECTED] on behalf of Lee Archer
Sent: Fri 2/17/2006 10:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] RE: Firmware version 1.3.1 released for
AastraIPphones
Nice one
Yes this is quite an issue. The POE converter is 'optional'. I bought
a 480i a while back and after waiting a few days had to order the POE
cos the dealer hadn't told me it was actually required!
Regards
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
I spent a days or two on this and in the end did
Musiconhold.conf
[livestream1]
mode=custom
application=/usr/local/bin/mpg123 -s --mono -y -f 8192 -r 8000 -@
/etc/asterisk/stream.playlist
Then in stream.playlist I just put the links from Shoutcast I wanted to
use
I used madplay with * 1.0 and moved to native for playing mp3's with 1.2 with
no problems. Depends what you want to play, doesn't native stop when there is
no one to play to then restart when there is someone to play to? Might be a
problem if you want to plays ads and don't have many callers,
Check out the musiconhold.conf.sample in the asterisksource/configs
folder.
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Faris
Raouf
Sent: 23 February 2006 18:36
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
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
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.
###
This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange.
For more information, connect
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]
, 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 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
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
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
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 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 -
Title: Running commands from dialplans
Hi, is it possible to run a command like system but from outside of a dial plan?
E.g.
; include extension contexts generated from AMP
#include extensions_additional.conf
In extensions.conf but I need to run a command.
Regards
Lee
--
No
Also NTL don't drop the leading 0 on incoming numbers like BT do.
Regards
Lee
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Stenton
Sent: 18 July 2005 11:42
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
I have all my GXP-2000's set to dynamic with no problems. You need to make
sure they have the latest firmware, as this fixed a few issues and improves the
overall usage of the phone. Hopefully they will make the useless LED's work so
we can line monitor etc...
Regards
Lee
-Original
Title: Dropping call
Hi, after upgrading from 1.0.7 to 1.0.9 I now seem to have a call drop problem. It mostly happens after about 1min 30 secs but also happens are random intervals. Everything was fine with 1.0.9 and I'm using the same config files. Could it be a zaptel problem? Does anyone
On a different note using Fedora Core 3 I get
CC [M] /usr/src/zaptel/zaptel.o
/usr/src/zaptel/zaptel.c: In function `zt_chan_write':
/usr/src/zaptel/zaptel.c:1745: warning: ignoring return value of
`copy_from_user', declared with attribute warn_unused_result
/usr/src/zaptel/zaptel.c: In
Title: SATA
Has anyone had any problems with SATA, either on board or 3rd party setup? I've currently got a problem where an AMD non SATA FC2 system is working fine but an Intel system with a 3Ware SATA card and FC3 is radomly not syncing with the ISDN30. It allows and receives calls but at
I noticed this, but then I moved to madplay which only uses 1 process.
Regards
Lee
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of MF Hulber
Sent: 27 July 2005 03:38
To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users
Mailing
I had a problem with this card and 2.6.11 kernel. I
am using FC3 but sticking with the 2.6.9 kernel. I had a lot of make
warnings on the zaptel build and the card played up. It also wouldn't do a
modprobe -r without crashing the system. With 2.6.9 zaptel compiles fine
and I can unload the
I tend to make it pause for 10 secs when loading the module as I have had a few
occurances of loading before /dev/zap has been populated. Wouldn't trust the
2.6.12 kernel as far as I could virutally throw it.
Has anyone had any problems with PCI-X systems? In particular call dropping?
I have been trying to get faxing working with stable but I have had no luck
since cvs 1.0.4. I've tried 3 versions of SpanDSP and the system answers the
fax but looks like it isn't training properly.
Lee
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf
But where do can you get this later firmware from? I'm still on 1.0.0.78 on my
480i.
Regards
Lee
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Passchier
Sent: 05 August 2005 00:04
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users]
This phone works fine, however the initial firmware it came with was
awkward. Once updated no problems, even NTP works!
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis
Sent: 11 August 2005 01:23
To: asterisk-users@lists.digium.com
Title: CRM software
Can anyone recommend CRM software with a link into Asterisk? I would like a pop up on caller ID if possible. I've played with the FOP and SugarCRM but can't get them working together.
Regards
Lee
___
Asterisk-Users mailing
Title: Faxing help
Hi, I have still had no luck with faxing and am getting a couple of pages of the following debug message
Changed from phase 1 to 4
DIS:
Prefer 256 octet blocks
Can receive fax
Supported data signalling rates: V.27ter and V.29
R8x7.7lines/mm and/or
I found that only the kernel is installed. I'd avoid
2.6.12 for now as I had problem with the zaptel driver and stay with
2.6.9.
Regards
Lee
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Damon
EstepSent: 24 August 2005 22:33To:
asterisk-users@lists.digium.comSubject:
Hi, do you have an on-site NTP server? I found that
after the firmware update NTP from the * server stopped
working.
Regards
Lee
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jesus
MogollonSent: 24 August 2005 22:11To:
asterisk-users@lists.digium.comSubject:
: Asterisk Users
Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users]
GXP 2000 Firmware 1.0.1.2
Hi Lee: NTP is working as expected, but it does take a
couple of minutes (!) to get the date from the serverJesus
Mogollon
2005/8/25, Lee Archer [EMAIL PROTECTED]:
Hi, do you
, Lee Archer [EMAIL PROTECTED]:
Hi, do you
have an on-site NTP server? I found that after the firmware update NTP
from the * server stopped working.
Regards
Lee
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of
Jesus MogollonSent: 24 August 2005 22:11To
What's the best way to get 1.0.8? I've downloaded the latest from CVS but when
I compile it it says 1.0.6!! Is that right?
Regards
Lee
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West
Sent: 23 June 2005 16:45
To: Asterisk Users Mailing
Title: Spinlock with ZAPTEL
Hi, I'm using Fedora Core 3 and the 1.0.8 version of ZAPTEL. Why do I get spinlocks when I modprobe -r it but the HEAD version unloads fine?
Regards
Lee
--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.323 / Virus Database:
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
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
###This message has been scanned by F-Secure Anti-Virus for Microsoft
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
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
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
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
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
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:
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 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
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
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
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
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:
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 -
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
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
- 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: 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
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: Installing beta2
Once built no matter whether I do make install or make clean I get the same output
[EMAIL PROTECTED] asterisk]# make clean
build_tools/make_version_h include/asterisk/version.h.tmp
if cmp -s include/asterisk/version.h.tmp include/asterisk/version.h ; then echo;
]
Installing beta2
Are you installing over a previous source tree? If so, please
rm -rf the previous source tree and install the new source tree from scratch.
On 11/2/05, Lee
Archer [EMAIL PROTECTED]
wrote:
Once built no matter whether I do make install or
make clean I get the same output
beta2
Can you issue the following command on FC3 and let us know the results?
rpm -q kernel-source zlib zlib-devel openssl openssl-devel
On 11/2/05, Lee Archer [EMAIL PROTECTED] wrote:
Hi, I had removed all old versions before starting and downloaded from
CVS.
Regards
Lee
I get an error when patching the makefile, seems the order is different.
Had the same problem with rc1 and 2.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton
Krall
Sent: 13 November 2005 17:34
To: 'Asterisk Users Mailing List - Non-Commercial
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