Douglas Garstang schrieb:
Is it just me or is the voip-info web site down right now?
I was experiencing problems accessing voip-info, too. But i guess the
problems derived from accessing http://www.google-analytics.com, because
i could see that voip-info was resolved rather quickly and after
Hi Gidean,
look at http://soft-switch.org/ and
http://www.voip-info.org/wiki/view/Asterisk+fax .
Bye
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von Gidean Chan
Gesendet: Mittwoch, 15. März 2006 08:32
An: asterisk-users@lists.digium.com
Benny Amorsen schrieb:
MG == Michael George [EMAIL PROTECTED] writes:
MG I may be way behind here, but I see that digium redesigned their
MG site. I cannot find the mailing list search screen.
MG I have found the mailman list page, but that doesn't have have a
MG nice search ability.
Tobias Wolf wrote:
Have taken a look at it and it looks really nice :) But because of the
issue of the Original Poster I looked at
http://lists.digium.com/pipermail/asterisk-users/ and saw that
Asterisk Users are indead ahead of time :)
Archive View by: Downloadable version
On Tue, Mar 14, 2006 at 11:26:14PM -0800, Ira wrote:
At 08:51 PM 03/14/2006, you wrote:
In my humble opinion, EVERYONE (unless you have your own in
a different voice/language) that uses Asterisk should be using
these prompts. How about a direct link this time:
For what it's
Hello,
Step by Step instructions on installing the card with Asterisk can be
found here:
http://www.eicon.com/support/helpweb/slnxen/asterisk.asp
Let me know if this helps
David
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Kennedy
Sent: 14
A user of mine has discovered that when you call into asterisk and get the IVR
menu with options 1-5 available, if you
dial 1 then immediatly dial 2 it will connect you to 2 and not 1. I expect
this is due to the digit timeouts and
response timeout. Is there a way to force an immediate action
14 mar 2006 kl. 15.38 skrev Matt:
The jitterbuffer branch is based on svn trunk (the same as the old
CVS HEAD)
The jitterbuffer-1.2 branch is based on the 1.2 branch HEAD
(meaning latest 1.2 version code).
Ok... so if I pull the 'jitterbuffer' branch I would get 'svn trunk
code'.
But if
On Wed, Mar 15, 2006 at 02:01:50PM +1100, James Harper wrote:
This is more an isdn question than an asterisk specific one, but is
there any end to end signalling channel available during call setup? Eg
if AParty dials BParty, can any information be conveyed (in both
directions preferably, and
Thanks!
- Original Message -
From: Philipp Dreimann [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Wednesday, March 15, 2006 4:18 PM
Subject: AW: [Asterisk-Users] Asterisk to receive fax
Hi Gidean,
look at
14 mar 2006 kl. 19.00 skrev Robert Webb:
On Tue, 14 Mar 2006 14:32:02 +0100
Olle E Johansson [EMAIL PROTECTED] wrote:
14 mar 2006 kl. 13.35 skrev Matt:
Right saw that. But I'm trying to get away from using CVS-HEAD :)
We all are. Every developer have switched from CVS to Subversion :-)
Hi,
we have an asterisk 1.2.1 box and 2 polycom SIP phones. We'd like to
show the called name on the calling polycom display instead of his /her
extensions as I do with the caller name on the called polycom.
Is it possible? If yes, how?
TIA
Giorgio Incantalupo
Hi,
Are you using Trunked IAX?
Currently we do not use trunking.
How many calls at a time?
All the test we have performed so far were with only one active call.
What codecs are you using?
We have set the bandwith=low, so I think that G.723.1, GSM, and LPC10 are in
the play.
What is the
Hello all! I want to assign one of the PSTN lines to a specific extension only.Expecting an earlier response. Thanks a lot.Faisal
Yahoo! Travel
Find
great deals to the top 10 hottest destinations!___
--Bandwidth and Colocation
MCC - Billing solution for Asterisk PBX
Current version: 1.3 + 1.3.1 Patch
MCC is a web-based, user (and admin) friendly billing interface for Asterisk
and VOIP.
MCC is open source software licensed under the GPL
Some features of MCC:
Unlimited SIP, IAX and Mobile/PSTN devices assigned to
Hi,
Just downloaded the latest cvs from zaptel on my sparking new Athlon64
Centos4.2 system, but hitting a stumbling block... (sorry for the long post)
#make
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64 -DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o gendigits.o gendigits.c
cc
I'm having problem powering Cisco phone's (7940 and 7905) on Linksys SRW224P
switch (with PoE functionality). I have tested three phone's, one is working
(7905) and two aren't (7905 and 7940). I have plugged all three phones on same
switch port with same cable!
Do I need to change anything in
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
I'm having problem powering Cisco phone's (7940 and 7905) on Linksys SRW224P
switch (with PoE functionality). I have tested three phone's, one is working
(7905) and two aren't (7905 and 7940). I have plugged all three phones on
same
Hi,
Im using frequently the perl api within asterisk.
Now Im looking for documentation for the perl
commands.
Some perl commands I found on this URL: http://www.voip-info.org/wiki/view/Asterisk+PHP
Does anybody got more documentation or where I can found
some more documentation
On Wed, 2006-03-15 at 17:49 +0800, Walter Klomp wrote:
Hi,
Just downloaded the latest cvs from zaptel on my sparking new Athlon64
Centos4.2 system, but hitting a stumbling block... (sorry for the long post)
Kernel source installed?
--
Dave Cotton [EMAIL PROTECTED]
Hi ,
I have been using, Monitor (wav, ${CALLFILENAME}) to records calls from extensions.conf and soxmix software to compiles calls. The problem I am facing that for long calls more that 2 minutes there is disturbance in sequence of calls, calls from both ends are not in sequence and there is
Hi ,
I have been using, Monitor (wav, ${CALLFILENAME}) to records calls from extensions.conf and soxmix software to compiles calls. The problem I am facing that for long calls more that 2 minutes there is disturbance in sequence of calls, calls from both ends are not in sequence and there is
I've a problem. I've some spa3000 and spa2100. Asterisk 1.2.4.
Prefered codec g711u in both. Calleng from a fxs of spa2100 to the fxo
of spa3000, all works ok. Then I call from a sip phone configured for
using g729, to the fxo of spa3000, it also works ok.
The problem is that after this, when,
2006/3/15, Gidean Chan [EMAIL PROTECTED]:
Can anyone tell me how to configure my system so that fax can be received
and forward to email account?
You can install iaxfax. It acts as a software modem that connects to
asterisk as a iax phone. It creates a device that can be accesed as a
faxmodem.
On 3/15/2006, Douglas Garstang [EMAIL PROTECTED] wrote:
Boy, am I stuck...
[snip]
My brain hurts.
Doug,
Whenever I have gotten to this point in a project, I use two rules for
handling the situation.
Rule 1. Booze
Rule 2. Throw money at it.
Rule 1 makes me feel better.
Rule 2 takes care of the
One more thing. Cisco 7905 phone that is working is 74-3092-04 Rev.F0.
Cisco 7905 phone that is not working is 74-3092-08 Rev.A0.
Anybody know about any hardware issue with this revisions?
Nothing for sure, and you may already know this, but some early Cisco
phones only knew how to speak
I have strange peaks of machine load on my asterisk servers, looking at
top the load is very high even if cpu usage is low and no swap memory is
used.
This happens on all the machines, some of them have asterisk, mysql, agi
and digium cards on them, so I thought I was only asking too much,
I've noticed this as well from pre 1.0 versions through to 1.2.5
across 12 separate Asterisk servers. The severity seems to be random
mostly. I still haven't figured out what is causing it.
MATT---
On 3/15/06, Simone Cittadini [EMAIL PROTECTED] wrote:
I have strange peaks of machine load on my
The SPA-2100 is the only one to support T.38 at the moment though.
SPA-2002 has the ability to support t.38 (i.e. it has the processing
power required) but the firmware support isn't there yet.
C F wrote:
On 3/11/06, Kristian Kielhofner [EMAIL PROTECTED] wrote:
Olle E Johansson
Hello Asterisk community,
I have written a document that covers an Asterisk implementation I am
building.
I want to place it on the lists so USERS can view and make comments on,
the ideas contained within.
I think it is an important issue to develop a standardised Dialplan for
applications,
This called hot line or batphone (as it's like the phone the
commissioner used to have in Batman that went straight through to Bruce
Wayne).
Set the dialplan to this:
(S0:#)
where is the number/SIP address you want to dial. Note, that's
a zero after the S.
Anton Krall
On Wed, 2006-03-15 at 14:01 +1100, James Harper wrote:
This is more an isdn question than an asterisk specific one, but is
there any end to end signalling channel available during call setup? Eg
if AParty dials BParty, can any information be conveyed (in both
directions preferably, and in
Walter Klomp wrote:
Just downloaded the latest cvs from zaptel on my sparking new Athlon64
Centos4.2 system, but hitting a stumbling block... (sorry for the long post)
Yup, there's a typo in the latest CentOS/RHEL kernel (confirmed by
Redhat). The fix is to edit the Zaptel Makefile (fix
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Nothing for sure, and you may already know this, but some early Cisco
phones only knew how to speak Cisco PoE, not the 802 standard which was
defined a bit later. The Cisco web site should tell you which phone
talks which protocol though.
On 15 Mar 2006, at 09:02, Stojan Sljivic - GDS wrote:
Hi,
Are you using Trunked IAX?
Currently we do not use trunking.
How many calls at a time?
All the test we have performed so far were with only one active call.
What codecs are you using?
We have set the bandwith=low, so I think
I dont have this cisco-check-cfg exten command in my asterisk...
Did you installed some extra module or channel?
Thanks
Joao Pereira
Aaron Daniel wrote:
It really depends on the number of phones you're wanting to reboot.
Whenever we do a reconfiguration of our phones, I have a script that
Hi Tim,
Just to check, can you get decent call quality between 2 IAX
clients on
the same
(local server)?
I have never tested that since we have no IAX phones.
We use SIP phones and IAX is used for connecting two Asterisk servers.
Regards,
Stojan Sljivic
-Original Message-
hi
in my callcenter i start asterisk on server with asterisk_safe
command, after 4 days i can see that it is crashed 12 times, reporting
segmentation fault error...each time asterisk is correctly restarted
without loss of services but, is it normal?
thanks
Hi,
I get this error in the log file when I call from my mobile to the Asterisk
server, but hang up the mobile before anyone picks up.
Normally I would not worry about it, but I have been having some bad
experiences (only recently, after about 9 months of good operation) with
asterisk, although
Olle E Johansson wrote:
11 mar 2006 kl. 23.44 skrev George Pajari:
*** ITU T.38 -- Fax over VoIP
It's not clear from the bug tracker if the problem with a T.38
endpoint (say ATA) behind NAT is working yet (with sip.conf
specifying nat=yes/qualify=yes). Is this working or do both T.38
I take care of a system that has Premisys IMACS units in it. They are
setup in pairs and each pair uses a single T1 circuit connected to their
WAN Dual 8010 cards. In an effort to create a redundat T1 link I could
really use some help in configuring the units to use the second T1 port
when the
7905/12, 7940/60 are NOT 802.3af compatible
ONLY 7911, 7941/61/70 (and their gigabiteth variants), are 802.3af
compliant
PJ
Tomislav Parčina wrote:
Hi James!
It seams that you are right. I have another phone (7940 - I have bough it the same time I bough 7905 that works) that works.
I have
I'm getting a strange error on one of the two controllers on an AVM C2
card under chan_capi-cm-0.6.3.
I have two ISDN controllers defined, both in the same group, both
connections are UK ISDN2e Point to Point:
On the third outbound call (both of the first two calls are handled by
the second
I am trying to use misdn insted of zaphfc to drive two billion isdn cards
zaphfc is ok, but the problem with cdr and the fact tha you always have to
wait the bristuffed version of asterisk took me to
try another way.
so I downloaded the misdn installation script from beronet for the last
version (
Hi there
I am using asterisk version 1.2.4. I have clients based on the iax client
library dialling into meetme sessions. I am experiencing echo in the case
where one or more users has speakers instead of headphones. So the audio
from me is fed from the other participant's speakers into their mic
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
7905/12, 7940/60 are NOT 802.3af compatible
ONLY 7911, 7941/61/70 (and their gigabiteth variants), are 802.3af
compliant
Totally not true!
I have 7905 phone that IS 802.3af compatible. Its on my table, right to my
laptop from which I'm
[EMAIL PROTECTED] wrote on 15.03.2006 14:37:27:
I am trying to use misdn insted of zaphfc to drive two billion isdn
cards
zaphfc is ok, but the problem with cdr and the fact tha you always have
to
wait the bristuffed version of asterisk took me to
try another way.
so I downloaded the misdn
On Tue, 2006-03-14 at 21:02 -0700, Douglas Garstang wrote:
Boy, am I stuck...
[snip]
Why don't you just hire a consultant/company to implement this on a no
cure no pay basis?
Regards,
Patrick
___
--Bandwidth and Colocation provided by Easynews.com --
Can Asterisk @ home receive incoming
call using a external modem?
Thanks
Gidean
Chan
___
--Bandwidth and Colocation provided by Easynews.com --
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Message: 20
Date: Wed, 15 Mar 2006 15:23:08 +0500
From: Mazhar Hussain [EMAIL PROTECTED]
Subject: [Asterisk-Users] there is lack behind
I am not sure but there may be a bug in the
Meetme() application.
The flag p (allow user to exit the conference by
pressing #) does not work when the flag m (sets monitor-only mode ) is also set.
I am unable to exit a conference when in monitor
only mode.
Can anyone tell me if
Tomislav, please look at:
http://powerdsine.com/Support/Certification/company_All.asp?company=Cisco
also on ci$co site you can't find info, that old phones are 802.3af
compliant, but pre-standard...
old ci$co phones can work with some poe equipment, but you can't be
sure, that will be working
This was non-trivial for me also. I prefer to right-click-copy the link
on the website, switch over to putty type in my wget (right-click), and
download the file directly to the box. The link I tried on the sounds
page happily downloaded index.html (if memory serves).
I did go ahead and get
I would recommend this particular method as well. It's quite a project,
but the end result seems to be a very solid, configurable solution.
Bob McDowell
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alejandro
Vargas
Sent: Wednesday, March 15, 2006
Steven Langley wrote:
Hi there
I am using asterisk version 1.2.4. I have clients based on the iax client
library dialling into meetme sessions. I am experiencing echo in the case
where one or more users has speakers instead of headphones. So the audio
from me is fed from the other participant's
Hi,
I have downloaded an IAX softphone and tested the connection locally.
The sound is perfect.
How should I troubleshoot this IAX connection between these two Asterisk
servers?
Is there some tool that can help in determining the cause of the choppy
sound?
Regards,
Stojan Sljivic
Hi, (sorry for my mistake in not deleting the rest of the message just now)
The problem seems to be here in zaptel.c (and torisa.c)
#ifdef DEFINE_SPINLOCK
static DEFINE_SPINLOCK(zaptimerlock);
static DEFINE_SPINLOCK(bigzaplock);
#else
static spinlock_t zaptimerlock = SPIN_LOCK_UNLOCKED;
static
Hi,
I bought a Digium
Quad E1 card model TE406P. Till now, I can't make it
work...
I mean, I have red
alarm when I configure one E1. The provider is in France (France Télécom)
and I use the following zaptel config :
span=1,1,0,ccs,hdb3,crc4bchan=1-15,17-31dchan=16
I'm using Linux
On Wed, 2006-03-15 at 11:26 +, Paul Hayes wrote:
The SPA-2100 is the only one to support T.38 at the moment though.
SPA-2002 has the ability to support t.38 (i.e. it has the processing
power required) but the firmware support isn't there yet.
Any info on the SPA-3000 and t.38 support?
Hi,
Using 1.2.5 with attended transfers, we are finding that dialling the
transferee is timing out after only 3 rings, after which the original
caller is transferred back.
I have searched high and low but cannot find anywhere to increase the
timeout for dialling the transferee.
This is a
Hi,
I run an asterisk server. The configuration is very basic.
Here is my question :
When someone calls my phone line, which is connected to an FXO card,
asterisk is answering using the context :
; Incoming calls goes to this default context :
[incoming-rtc]
include = postes-sip
;
exten =
I found a bug in the latest T38 passthrough patches, the effect
is that a non-SIP call after being put on hold is then lost, no
resume is possible.
The fix is to be applied in the chan_sip.c file:
} else {
My PocketPC with ppcIAX and/or SJPhone behaves in exactly the same way.
The only resolution is to use an earbud... I'm guessing that the
server's echo cancelling is intended to cancel minor echo introduced by
the path, but doesn't handle 'real' echo caused by looping sound. Is
that right?
Bob
Hi everyone,
Been reading up on Asterisk, and very interested in learning more. I've
googled and read the archives and haven't found anything definitive on
support for this phone system. We have a fairly large investment in the
system itself and the phones, but would love to get away from the
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Tomislav, please look at:
http://powerdsine.com/Support/Certification/company_All.asp?company=Cisco
also on ci$co site you can't find info, that old phones are 802.3af
compliant, but pre-standard...
old ci$co phones can work with some
Douglas Garstang [EMAIL PROTECTED] wrote:
Boy, am I stuck...
I'm officially ready to toss Asterisk out the window. I have to admit it isn't
necessarily all the fault of Asterisk either. It just seems that every option I
turn to suddenly ends in failure. I don't know if it's me that's bitten
I just set up an Aastra 480i CT with separate registrations on my Asterisk
server. The way I set it up is Line 1 on the phone is registered to 101 on
the server and Line 3 is registered to 103.
If Line 1 is being used and a call comes in on 101, it rings to Line 2. But,
if Line 3 is being used
Whoops ! sorry - wrong release ...
chan_capi-cm-0.6.4 !
John Daragon wrote:
I'm getting a strange error on one of the two controllers on an AVM C2
card under chan_capi-cm-0.6.3.
I have two ISDN controllers defined, both in the same group, both
connections are UK ISDN2e Point to Point:
Thank you for your answer.
I tried that syntax with misdn/g:TEports/${EXTEN}, but nothing changes.
what should I write in the /etc/capi.conf ?
If I had a Fritz, I would have
#SuSEconfig.isdn generated
# card fileproto io irq mem cardnr options
fcpci - - -
Hi
You need to use a cross-over E1 cable (not an ethernet cross-over one)
Good luck
Jose Manuel Cortes David
X Semestre Ingenieria Electronica
PONTIFICIA UNIVERSIDAD JAVERIANA
De: [EMAIL PROTECTED] en nombre de David Masure
Enviado el: Mié 15/03/2006
Hi
Im developing an ip telephony project and i need
some help in order to choose the better PCI card, the options at the moment are
digium, sangoma and voicetronix, the strongest ones are digium and sangoma but i
dont know how justify the election
Best regards
Jose Manuel Cortes
David
Hi,
We're using Asterisk to develop a specialized IVR system for our
employees and someone is telling us there is some OSHA requirement that
you have to always be able to reach a live human on such systems. I've
never heard of that and google didn't turn up anything in my searches.
This is
as I known, all ci$co switches supports pre-standard ci$co phones and
mostly all todays switches also supports new 802.3af phones (and also
pre-standard phones)
PJ
Tomislav Parčina wrote:
Anther thing, is there any Cisco switch that supports even oldest Cisco VoIP
phones (7905 and
Hi
again,
Can
you specify the pin order for each end ?
thanks
-Message d'origine-De: JOSE MANUEL CORTES
DAVID [mailto:[EMAIL PROTECTED]De la part de
JOSE MANUEL CORTES DAVIDEnvoyé: mercredi 15 mars 2006
16:28À: Asterisk Users Mailing List - Non-Commercial
Bob McDowell wrote:
My PocketPC with ppcIAX and/or SJPhone behaves in exactly the same way.
The only resolution is to use an earbud... I'm guessing that the
server's echo cancelling is intended to cancel minor echo introduced by
the path, but doesn't handle 'real' echo caused by looping sound.
Hi,
I am using a cisco 7912. I setup the phone at my first location. I
edited the gkMAC.txt
file setup the proxy and UID etc... values. generated the gkMAC file
and booted
the phone and it worked...
I then mailed it to its final destination. this place has other 7912
phones working there..
On 3/15/06, Charles Marcus [EMAIL PROTECTED] wrote:
Can anyone provide any feedback on using this system with Asterisk? Am Iwasting my time even thinking about it?
I run Asterisk partnered up to the 280 (424 for us). We have a 6 cabinet installation of the Toshiba so I understand you dilemma.
On Wed, 15 Mar 2006, John Daragon wrote:
Whoops ! sorry - wrong release ...
chan_capi-cm-0.6.4 !
There were problems with 0.6.3 and AVM cards, but should be fixed in 0.6.4.
Can you please create a full debug log (set verbose 5; capi debug) for such
a case ?
Armin
John Daragon wrote:
On 3/15/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote:
Hi,
we have an asterisk 1.2.1 box and 2 polycom SIP phones. We'd like to
show the called name on the calling polycom display instead of his /her
extensions as I do with the caller name on the called polycom.
Is it possible? If yes, how?
##
# mISDN (experimental) #
##
#avmfritz - - - - - -
#hfcpci - - - - - -
#hfcsusb- - - - - -
Forgive me if this question has been asked/answered in another post.
And let me reiterate what other users have frequently said - Asterisk is great,
and I really appreciate all the work you folks have put into it.
How have some of you gone about integrating Asterisk with a legacy office PBX,
I wanted to investigate this myself, so I called OSHA, got VoiceMail!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Scott Plante
Sent: Wednesday, March 15, 2006 10:37 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] OSHA
People love to blame things on acronyms. Usually it's 'HIPPA' (which,
by the way is a clear indicator that they've never studied HIPAA),
sometimes OSHA, etc.
If it really is OSHA then it should be pretty easy to find out. If (and
check first) your organization is on the up and up, call them
Armin Schindler wrote:
On Wed, 15 Mar 2006, John Daragon wrote:
Whoops ! sorry - wrong release ...
chan_capi-cm-0.6.4 !
There were problems with 0.6.3 and AVM cards, but should be fixed in 0.6.4.
Can you please create a full debug log (set verbose 5; capi debug) for such
a case ?
(top posting for brevity, but original post included below, as it was
over seven weeks ago)
I've at last updated the patches for both trunk and 1.2, and posted them
to Mantis at http://bugs.digium.com/view.php?id=6731
Cheers
Tony
In article [EMAIL PROTECTED],
Tony Mountifield [EMAIL PROTECTED]
Scott Plante wrote:
Hi,
We're using Asterisk to develop a specialized IVR system for our
employees and someone is telling us there is some OSHA requirement
that you have to always be able to reach a live human on such
systems. I've never heard of that and google didn't turn up anything
in
Hi Scott -
We're using Asterisk to develop a specialized IVR system for our
employees and someone is telling us there is some OSHA requirement that
you have to always be able to reach a live human on such systems. I've
never heard of that and google didn't turn up anything in my searches.
On 3/15/06, Scott Plante [EMAIL PROTECTED] wrote:
Hi,
We're using Asterisk to develop a specialized IVR system for our
employees and someone is telling us there is some OSHA requirement that
you have to always be able to reach a live human on such systems. I've
never heard of that and google
Hi Giorgio -
we have an asterisk 1.2.1 box and 2 polycom SIP phones. We'd like to
show the called name on the calling polycom display instead of his /her
extensions as I do with the caller name on the called polycom.
Is it possible? If yes, how?
If this is possible, it would be quite
I just download and compile asterisk-addons. But whne I tried to start Asterisk and I go t error as below:
[res_config_mysql.so]Mar 15 09:32:24 WARNING[10597]: loader.c:325
__load_resource: /usr/lib/asterisk/modules/res_config_mysql.so: cannot open
shared object file: No such file or
Are you selling it TO osha? If so, maybe they have an internal requirement..
If not, I've never heard of that. Granted, I haven't sold a LOT of phone
systems, but I've been involved with a couple into public works departments of
local governments as well as private corps, and nobody has ever
Dear All,I am currently have this problem in which I am sending call out from the Zaptel TE405 to a VoIP gateway. But the problem that the call over to the VoIP Gateway will always have a fake ring tone. Can you please give some pointer how to fix this problem? This problem is existing in my
But, ...your call is very important to us... :)
Alexander Lopez wrote:
I wanted to investigate this myself, so I called OSHA, got VoiceMail!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Scott Plante
Sent: Wednesday, March 15, 2006 10:37 AM
To:
Dear All,It seems that there is a bug on the ooh323 while using registering with gatekeeper. The gatekeeper is GnuGK and the problem is when the Asterisk recieves a call from the Gatekeeper and routes it back out to an SIP Phone. The call would be connected and immediately dropped after 1-2
In the sip_notify.conf file, there's a couple different events that will
cause different phones to reboot. One of them is cisco-check-cfg. In
the asterisk cli, if you run sip notify cisco-check-cfg exten with
that file in your tftpboot directory, you'll send the phone a reboot
command.
Kenige Ho wrote:
Dear All,
I am currently have this problem in which I am sending call out from the
Zaptel TE405 to a VoIP gateway. But the problem that the call over to the
VoIP Gateway will always have a fake ring tone. Can you please give some
pointer how to fix this problem?
Don't use
I upgraded my Cisco7960 to SIP 8-2 from 7-4. Everything seems ok, works
fine. Except that when I make an outbound call, I get a double-ring
sound. I also found that if the target number is engaged, I get a ring
sound and at the same time get a busy sound.
If I revert back to 7-4, there is no
On Wednesday 15 March 2006 11:20, Steve Jones wrote:
Are you selling it TO osha? If so, maybe they have an internal
requirement.. If not, I've never heard of that. Granted, I haven't sold a
LOT of phone systems, but I've been involved with a couple into public
works departments of local
On Wed, 15 Mar 2006, John Daragon wrote:
Armin Schindler wrote:
On Wed, 15 Mar 2006, John Daragon wrote:
Whoops ! sorry - wrong release ...
chan_capi-cm-0.6.4 !
There were problems with 0.6.3 and AVM cards, but should be fixed in 0.6.4.
Can you please create a full debug log (set
I was looking for this exactly as well
Any ideas?
- Gabe
- Original Message -
From: Giorgio Incantalupo [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, March 15, 2006 12:52 AM
Subject:
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