It is not clear how exactly g729 pass-through can be enabled. I
have a SIP call off a gateway come into an Asterisk menu, and then I
send the SIP call to another SIP gateway using Dial(). Even though
codec preferences have g729 listed first, it never gets used.
Both gateways have separate
On Wed, 16 Feb 2005 07:08:08 +0800, Leo Ann Boon [EMAIL PROTECTED] wrote:
See my comments in line
From my experience, the ATA is a very solid, dependable piece of
hardware. I was told by a source in the company that OEMs for Cisco, the
units are expensive because of the high quality parts
You can also use the manager.
Take a look at http://www.voip-info.org/wiki-Asterisk+manager+API
- Original Message -
From: Kevin P. Fleming [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, February 16, 2005 6:12
Shaun Ewing wrote:
I'm using the ATA 186 and think it's great. The latest firmware also
changes the web interface - it's similar to the 7905G/7912G phones
now.
Did you have much trouble getting the ATA 186 working?
I have one running Version: v3.1.0 atasip (Build 040211A)
I have it setup and it
Adding in experimental patches willy-nilly, especially ones that have
the potential to cause huge problems, confounds attempts to isolate
bugs and test functionality.
Mark does a pretty good job of keeping the HEAD version solid enough
to use in production, as most of us running it on a daily
Liaan vd Merwe wrote on Tuesday, 15 February 2005 1:37 PM:
http://edgett.bc.ca/simonsays/archives/000228.html
Thank you, but this is not the script.
Sincerely,
Trevor Hammonds
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This is the example script (extracted from that link)
you will need to find a weather page for your region
an then change the urls
and grep statements
chow
L
[weather.sh]
#!/bin/sh
WEATHER=`lynx -nolist -dump
http://weatheroffice.ec.gc.ca/forecast/textforecast_e.html?Bulletin=fpcn11.cwvr
|
On Wed, 2005-02-16 at 09:23 +0800, Stuart Elvish wrote:
What sort of setup is involved for the Cisco as far as config files
etc? I am used to plug and play phones (Zyxel, Grandstream, HOP etc)
which require minimal configuration and have no licensing issues with
them. I know for the Polycom
Hello,
I was interconnecting Asterisk (v1.0) with a strict
router (ie, no ;lr in routes) and I think I found a
bug in the way Asterisk prepare new requests inside a
dialog.
I'm sending some captures (ngrep) along with my
comments.
This is a 200 OK (INVITE) received by Asterisk
Hello !
First time I have instaled Asterisk without problem and working with a SIP
clinet (X-Lite).
Then I try to make the H323 with came with Asterisk.
So, I DL pwlib v.1.5.2 in /root (./configure ; make) no errors
DL openh323 in /root (./configure ; make opt):
hi all
i have created an accoutn with sip.phonehome.co.za
and register it in asterisk this seems to have no problem as
sip show peers displays the connection to the sip proxy
but when i make a call from an extension to the sip number
after dialing the phone starts ringing immidiately
a tcpdump
Liaan vd Merwe wrote on Wednesday, 16 February 2005 2:53 AM:
This is the example script (extracted from that link) you will need
to find a weather page for your region an then change the urls and
grep statements chow L
Once again, this is NOT the script mentioned at Eric Wieling's former
Hi Trevor
This i know
I just send you a other script doing the same task
this will give you a guideline to make you own
- Original Message -
From: Trevor G. Hammonds [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial
Discussion'
asterisk-users@lists.digium.com
Sent:
On Wed, 16 Feb 2005 03:50:28 -0800, Trevor G. Hammonds
[EMAIL PROTECTED] wrote:
Liaan vd Merwe wrote on Wednesday, 16 February 2005 2:53 AM:
This is the example script (extracted from that link) you will need
to find a weather page for your region an then change the urls and
grep
Hi ALL;
I saw several examples of "Dial" app with the
format:
Dial(Local/..)
Anybody knows what the "Local" technology
means?
Regards
Mohamamd
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On Wednesday 16 February 2005 11:40, Nemesis wrote:
Hello !
[...]
ast_load_resource: libpt_linux_x86_r.so.1.5.2: cannot open shared object
file: No such file or directory
Feb 16 13:02:21 WARNING[31879]: loader.c:440 load_modules: Loading module
chan_h323.so failed!
[EMAIL PROTECTED]
On Wed, 16 Feb 2005 03:40:38 +0330, mohammad [EMAIL PROTECTED] wrote:
I saw several examples of Dial app with the format:
Dial(Local/..)
Anybody knows what the Local technology means?
Did you try the WiKi? Or Google?
http://www.google.com/search?q=asterisk+local
--
Peter
I've got a test * server (hppbx) where I install CVS-HEAD as often as
possible, with my extension registered to this, talking through IAX to
our production server which then channels out to the PSTN.
After completing a call just now, the following appeared on the CLI of
hppbx (the
LOL, I'm a dumba$$ please ignore :-)
-Original Message-
From: Matt Schulte
Sent: Tuesday, February 15, 2005 2:41 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Ser 0.9.0 adding a user?
I get this when adding a user in ser (using serctl)
[EMAIL PROTECTED] sbin]#
Matt Schulte wrote:
LOL, I'm a dumba$$ please ignore :-)
Might help to post what you did wrong for the archives...although, I
guess it isn't really Asterisk related.
:)
--
Cheers,
Matt Riddell
___
http://www.sineapps.com/news.php (Daily Asterisk News -
Hi;
As you probably know, SER style of handling an
incoming call is :
1) try to look-up it from registrar DB
2) if not found there, try to do some
thing else
Is there any possibility of doingthe
aboveat "Asterisk Dial-plan"?
Regards
Mohammad
David Uzzell wrote:
Shaun Ewing wrote:
I'm using the ATA 186 and think it's great. The latest firmware also
changes the web interface - it's similar to the 7905G/7912G phones
now.
Did you have much trouble getting the ATA 186 working?
No.
I have one running Version: v3.1.0 atasip (Build 040211A)
At 12:10 16-02-05 +, you wrote:
On Wednesday 16 February 2005 11:40, Nemesis wrote:
Hello !
[...]
ast_load_resource: libpt_linux_x86_r.so.1.5.2: cannot open shared object
file: No such file or directory
Feb 16 13:02:21 WARNING[31879]: loader.c:440 load_modules: Loading module
chan_h323.so
mohammad wrote:
Hi;
As you probably know, SER style of handling an incoming call is :
1) try to look-up it from registrar DB
2) if not found there, try to do some thing else
Is there any possibility of doing the above at Asterisk Dial-plan?
Just forward the call to Asterisk if it has a certain
Mondial Software Limited
020 7043 2795
www.mondialsoftware.com
Click here to view our presentation of Cash Controller showing its
forecasting and automated bank reconciliation features
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter Bowyer
Sent:
when connecting through a sip proxy server outside our network on the net
to connect to a land line.
the call connects to the land line.
but we cannot hear the other party
they can hear us.
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On Wed, 16 Feb 2005 15:42:18 +0200, Nemesis [EMAIL PROTECTED] wrote:
At 12:10 16-02-05 +, you wrote:
On Wednesday 16 February 2005 11:40, Nemesis wrote:
Hello !
[...]
ast_load_resource: libpt_linux_x86_r.so.1.5.2: cannot open shared object
file: No such file or directory
Feb 16
Please file a bug report at bugs.digium.com - thanks!
Cheers, Philipp
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Hi Florian,
If you really are using ulaw, and you do not have extreme packet loss or
jitter, DTMF detection should be very reliable. It is no better in CVS
HEAD because it wasn't broken in the first place.
What does wrong with DTMF detection? Do you realise how DTMF from a GSM
phone works? If
On Tue, Feb 15, 2005 at 07:42:56PM -0500, Nabeel Jafferali said:
4. Scnet.net has 5 pages website (quite a work for ISP), that
any kid could create in 1h
scnet.net is Server Central, a data centre where my host (HostForWeb),
among others, maintains their servers. I do know it is a reliable
What I did wrong was post it to the wrong list, heheh *shame on me*
and no, still no resolution. :-(
-Original Message-
From: Matt Riddell [mailto:[EMAIL PROTECTED]
Sent: Wednesday, February 16, 2005 7:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Hello * users
I've have a rather disturbing problem, which I
don't know how to debug or how to solve, but first a brief description of
the set up.
One Asterisk server with a TE410P card installed (first
line used on this only), and a number of Wellgate 3504A (4 port FXS devices
with
I am running 1.0.5 and can happily blind xfer from extension to
extension, but I can't blind xfer. I have read various snippets about #2
or #8 or other such key combos, but nothing seems to let me do attended
xfer.
From xlite I can blind xfer without problem but no attended xfer.
For
I'm seeing the same problem here, all SIP calls go to the default context.
Kelvin Chua wrote:
this is something i just recently noticed.
have you found any info on how to manage incoming calls through
chan_h323? it doesn't seem to match any entity you define, it always
uses the default context...
On Wed, 16 Feb 2005 22:04:27 +1100, Adam Goryachev
[EMAIL PROTECTED] wrote:
BTW, Polycom *don't* say you can't use their phone in/with any
particular manner/software. All they say is that if the phone breaks,
and it is caused by asterisk, then they won't really help you out.
However, the fact
Are you paying me? Did I ask you to do this? Did you get permission
from all 10,000 to do this?
On Wed, 16 Feb 2005 13:40:41 -, Bill Seddon
[EMAIL PROTECTED] wrote:
Mondial Software Limited
020 7043 2795
www.mondialsoftware.com
Click here to view our presentation of Cash Controller
Matt Riddell wrote:
mohammad wrote:
As you probably know, SER style of handling an incoming call is :
1) try to look-up it from registrar DB
2) if not found there, try to do some thing else
Is there any possibility of doing the above at Asterisk Dial-plan?
Just forward the call to Asterisk if
CVS in a production environment? Is that advisable?
[EMAIL PROTECTED] wrote:
I am running 1.0.5 and can happily blind xfer from extension to
extension, but I can't blind xfer. I have read various snippets about #2
or #8 or other such key combos, but nothing seems to let me do attended
xfer.
Hello * users
Sorry I forgot to send the mail in plain text the first time...
I've have a rather disturbing problem, which I don't know how to debug or
how to solve, but first a brief description of the set up.
One Asterisk server with a TE410P card installed (first line used on this
only), and
I have an 866# number with iax.cc. It works fine. It did take me a
couple of days to get it and they did have some problem the first day it
was active, but I was able to contact support via IM and email. They
resolved the problems and the service is working fine for me. Although,
I still
On Feb 15, 2005, at 1:12 PM, Florian Lefeuvre wrote:
I find the compilation option RADIO_RELAX.
this option change a threshold in DTMF detection (function dtmf_detect
in dsp.c)
I remark an big improvement in the detection of the dtmf over GSM.
have you ever test this option?
RADIO is obscur for
Hi
Has anybody tried using asterisk on an ipv6 internal
network?
If so, any feedback or comments would be very appreciated.
Im not sure, but is ipv6 a real-time protocol
already?
,jm
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On Tue, 15 Feb 2005 14:05:36 +0100, Vledder, Hans wrote:
I've been thinking of making a (mostly) solid-state asterisk pbx.
Take either centos or some other distro, cut it down to bare minimum and
put asterisk + AMP on. Something that could be put onto a usb2.0 flash
stick, bootable.
Modern
On Wed, 16 Feb 2005, Steve Underwood wrote:
If you really are using ulaw, and you do not have extreme packet loss or
jitter, DTMF detection should be very reliable. It is no better in CVS
HEAD because it wasn't broken in the first place.
We have some problems with dtmf detection on our
On Feb 15, 2005, at 2:32 PM, Matthew Boehm wrote:
Stop. The PAP2-NA's have no T38 support. Next time, lets try and read
the
OP's message before responding.
-Matthew
Hah! With over 2-- or 300 messages per day we're supposed to read
everything in them? I find it easier just to respond to multiple
On Wed, 2005-02-16 at 14:11 +, Mark Benson wrote:
As for the alternative to attended xfer, parking calls, I'm guessing
this is just a case of blind xfering calls to a parking extension?
That is correct... if 800 is your parking extension then you dial #800,
you will hear what extension
Hi,
I have installed two X-Lite phones and theyre
able to login successfully. The two phones plus the Asterisk system are all on
the same LAN with private addresses assigned to each of them. When a call
is initiated and is picked up on the other end, there is completely no sound at
all
Does anyone know if the attended transfer in CVS head works with
app_queue (and more importantly, chan_agent ?)
This is the only thing stopping me from deploying the attended transfer
patches.
Cheers,
Ben
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
On Feb 15, 2005, at 6:08 PM, Leo Ann Boon wrote:
From my experience, the ATA is a very solid, dependable piece of
hardware. I was told by a source in the company that OEMs for Cisco,
the units are expensive because of the high quality parts being used.
The web config looks crappy but otherwise
On Feb 16, 2005, at 10:01 AM, Peter Svensson wrote:
On Wed, 16 Feb 2005, Steve Underwood wrote:
If you really are using ulaw, and you do not have extreme packet loss
or
jitter, DTMF detection should be very reliable. It is no better in CVS
HEAD because it wasn't broken in the first place.
We have
Hi Peter,
If that is true, someone must have broken something. Not only does the
DTMF detector I wrote not care about small imperfections, it even
tolerates a dropped packet with the DTMF passes over a VoIP path (this
kind of tolerance was added a couple of years ago).
Regards,
Steve
Peter
Hi Mark,
Mark Eissler wrote:
BTW, Steve, if you're still reading, what is the RADIO_RELAX option
intended to be for in dsp.c?
It is something someone else added to the code to make the detection
criteria in relaxed mode even more relaxed. If setting that helps,
something in your channel must be
Steve:
Given this and the number of recent messages related to DTMF problems
can you add any thoughts on how to improve implementations?
I too have problems correctly handling DTMF in my environment. In the
LAN /IP world DTMF works as expected to allow an IP phone user to
access my Asterisk
Here is my conf files.
sip.conf
register = phone#:sip/[EMAIL PROTECTED]
type=friend
username=phone#
fromuser=phone#
secret=sip/passwd
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
context=from-broadvoice1
dtmfmode=inband
disallow=all
allow=ulaw
canreinvite=no
nat=no
[bv-in-1]
Yes turn off silence suppression.
xlite - Menu - Advanced - audio settings - Silence
Settings - transmite Silence: (change to yes)
- Original Message -
From:
Julius
Kidubuka
To: asterisk-users@lists.digium.com
Sent: Wednesday, February 16, 2005 10:04
AM
OK I have to ask.
Why is it that Asterisk can't cope with silence suppression?
All the clients seem to be able to but not Asterisk.
What would be needed to get it to work with silence suppression?
What is the problem?
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Hi,
I read a lot about US providers that can terminate a PSTN
number for you and offer IAX or SIP connectivity.
Does anyone know such a company in The Netherlands ?
I read about Unet. Anyone with experience with them ?
Any information is welcome.
--
Michiel van Baak
http://lunteren.vanbaak.info
Hello,
I have been attempting to get the Monitor function to
accept a loal variable substitution in order to use
the same filename later in the same context. Monitor
does not appear to like it, as it attempts to use
wav|filename as the recording type, as opposed to just
wav.
Here is what I get
Hello,
I'm attempting to get Asterisk running for the first time in my company. As
I've never used it before, I am creating a small testbed with which to learn
Asterisk and get the kinks worked out before attempting to roll it out.
I have * compiled and running, and built the sample config
OK I have to ask.
Why is it that Asterisk can't cope with silence suppression?
All the clients seem to be able to but not Asterisk.
What would be needed to get it to work with silence suppression?
What is the problem?
Essentially its because * has been architected to send an rtp
packet
-Original Message-
From: Chris Wade [mailto:[EMAIL PROTECTED]
Brian Roy wrote:
I think that my PBX does this too. Is there any way I can get the
Zaptel drivers to disconnect on that tone too? I would love
to replace
my existing voicemail with * but I can't get my PBX to signal
On Wed, 16 Feb 2005, Steve Underwood wrote:
If that is true, someone must have broken something. Not only does the
DTMF detector I wrote not care about small imperfections, it even
tolerates a dropped packet with the DTMF passes over a VoIP path (this
kind of tolerance was added a couple
Hi,
do you need a sending or receiving fax solution ?
Receiving fax via Asterisk and misdn - no problem, but i have no sending
fax solution at this time.
Andreas.
--On Freitag, 11. Februar 2005 15:04 + Anabela Abreu [EMAIL PROTECTED]
wrote:
Was anyone put hylafax working with chan_misdn?
Tom Samplonius wrote:
It is not clear how exactly g729 pass-through can be enabled. I
have a SIP call off a gateway come into an Asterisk menu, and then I
send the SIP call to another SIP gateway using Dial(). Even though
codec preferences have g729 listed first, it never gets used.
Without
On Wed, 16 Feb 2005, Rob Scott wrote:
Why is it that Asterisk can't cope with silence suppression?
All the clients seem to be able to but not Asterisk.
What would be needed to get it to work with silence suppression?
What is the problem?
Asterisk clocks outgoing rtp data to a device from the
Here is a part of a working sip.conf for Asterisk with SNOM190 Phones.
Try using host=dynamic and have a closer look at the configuration in the
snom 190.
Also, try using dtmfmode=rfc2833 .
[general]
realm = hallinux2.gwsnettech.local
port = 5060
bindaddr = 0.0.0.0
context = default
disallow=all
Got two phones here. 1 is Cisco 7960 and other is XTen Pro. Both have 729
capabilities and plenty of licenses on Asterisk. The Cisco phone has and
registers/talks with asterisk on an internal IP (* = 10.0.3.10, phone =
10.0.3.151). The SIP peer for this phone is set to NAT=No and has this Codec
My problem is that the 23 channels are going into asterisk. It seems that
there is no way to pick off a couple of them and use them for faxes.
Basically, I have 23 phone lines and can't use them for faxing.
Thanks,
Brian
I am using a T100P for a 23-channel voice T1. Is it possible to create
Yes i would like to have a solution with asterisk, hylafax
and misdn.
Em Wed, 16 Feb 2005 17:23:03 +0100
Andreas Czerniak [EMAIL PROTECTED] escreveu:
Hi,
do you need a sending or receiving fax solution ?
Receiving fax via Asterisk and misdn - no problem, but i
have no sending fax
At 13:41 16-02-05 +, you wrote:
On Wed, 16 Feb 2005 15:42:18 +0200, Nemesis [EMAIL PROTECTED]
wrote:
At 12:10 16-02-05 +, you wrote:
On Wednesday 16 February 2005 11:40, Nemesis wrote:
Hello !
[...]
ast_load_resource: libpt_linux_x86_r.so.1.5.2: cannot open shared
object
On 2005.02.16 09:00 Brian M. Arlinghaus wrote:
My problem is that the 23 channels are going into asterisk. It seems
that there is no way to pick off a couple of them and use them for
faxes. Basically, I have 23 phone lines and can't use them for faxing.
You could always run another T100P into a
Hi there,
I tried to use Voicemail from a PRI interface but it didn't work because
pressing a key on the ISDN phone just caused Q931_IE_KEYPAD_FACILITY
messages which are normally handled by a bri-stuffed libpri.
Unfortunately a wrong if condition stops keypad messages from being
passed to
On 2005.02.16 09:00 Brian M. Arlinghaus wrote:
My problem is that the 23 channels are going into asterisk. It seems
that there is no way to pick off a couple of them and use them for
faxes. Basically, I have 23 phone lines and can't use them for faxing.
You could always run another T100P
Greetings David,
PerlBox would not be usable for the level of service that is needed by
Asterisk to be viable Speech.
PerlBox is a vocabulary based recognizer, or I as I call it a grunter,
where you grunt something and it then does something cute.
Grunters depend on you creating a vocabulary
Has anybody managed to implement Sphinx in their * system reasonably
painlessly.
if so:
does it cause any problems with normal * operations.
does it place any sort of constant heavy load on the machine.
are there options for simple vs advanced implementations.
all i am looking for is basicaly
David,
Thanks for the reply. Just to clarify, is the register and first
type=friend block all within the [general] section of sip.conf?
Thanks,
Max
Max Clark
max [at] clarksys.com
http://www.clarksys.com
David Shaw wrote:
Here is my conf files.
sip.conf
register = phone#:sip/[EMAIL
On Feb 16, 2005, at 10:34 AM, Steve Underwood wrote:
BTW, Steve, if you're still reading, what is the RADIO_RELAX option
intended to be for in dsp.c?
It is something someone else added to the code to make the detection
criteria in relaxed mode even more relaxed. If setting that helps,
something
Title: Inter-asterisk conferencing delays - IAX2 configuration problem?
Hi
We are having a significant ( 1 sec) delay in a multi-asterisk conference, with IAX2 legs connecting meetme on different boxes.
All the other legs are PSTN (TE410P). The example configuration
Slave box 1 meetme
it only reloads asterisk. in 0.6 it also reloads FOP.
--- David Josephson [EMAIL PROTECTED] wrote:
Yes - the problem was a missing signalling line in
zapata.conf. Now in
and out work.
Also, it was news that reload from the console
doesn't reflect changes
made in zaptel and zapata.conf
AT-320EE
Anyone try these? Do they work? any reviews? I couldn't find jack on
google..
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I was also looking at these this morning but couldnt find any info. I am
interested in an IAX hardphone that works.
Matt Schulte wrote:
AT-320EE
Anyone try these? Do they work? any reviews? I couldn't find jack on
google..
___
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In a production environment, I would not attempt to run Sphinx on
the same computer as Asterisk A few users interacting with
Sphinx could consume all of the server's resources and then some.
Same goes for DMBS servers, One big N-way join could tie up a
CPU for tens of seconds.
--- Mark Kidd
Wang Xiangzhou wrote:
Sun claims that Linux apps can run on Solaris 10 natively. Is there
anyone to run Asterisk on Solaris 10 and what the results are.
Thanks,
William
why not just compile asterisk on sol10?
Ming-Wei
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New features include Festival text to speech and a new
Web Conferencing GUI. There are also numerous small
fixes and enhancements.
http://asteriskathome.sourceforge.net/
__
Do you Yahoo!?
All your favorites on one personal page Try My Yahoo!
Hello,
Does anyone here use any WLAN phones with asterisk? Are there any
posts about problems, security (and prices in germany)?
Bye, Olaf
--
Olaf Klein
Adimus Beratungsges. für System- und Netzwerkadministration mbH
Harpener Hellweg 41
44805 Bochum
Tel. 0234-95015-13
Fax. 0234-95015-29
Sphinx and Festival are good projects. The last I worked with sphinx
I
was told that it would need modifications to make it more grammar
aware,
but that was 2 years ago and things may have improved. If not then
Sphinx people please let me know when you will add grammars natively
or
refer
Yeah, I've been running asterisk 1.0.3 and 1.0-RC1 before that on
Solaris 10. I'm only using it for personal use though. Really I'm just
using SIP to a sipura, broadvoice and freeworlddialup with voice mail
and such. It works fine for my purposes but I can't attest to testing
it well enough for
As for a good way to log him out, you can set
autologout=20
in agents.conf in order to logout agents whose phone rings more than 20
seconds.
Ideally, this should be set to the same value as the timeout on the queue
that the agent is not answering. As for emailing their manager - Not a
built-in
You could always run another T100P into a HylaFAX-run T1 fax modem.
That way you can use your T1 for faxing.
Could you explain a litter further? Thanks.
On 2005.02.16 09:00 Brian M. Arlinghaus wrote:
My problem is that the 23 channels are going into asterisk. It seems
that there is no way to
Lee
So the drop/insert channel bank will pick off a few of the channels and
send the rest to asterisk? Is this some sort of Adtran product? What about
DIDs?
Thanks,
Brian
- Original Message -
From: Jon Pounder [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial
I have a Polycom 400 with H.323 firmware. I know it is not capable of
loading the SIP firmware.
Anybody know if I can get the MGCP firmware (and maybe the bootloader) from
somewhere?
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Essentially
its because * has been architected to send an rtp packet after
receiving a packet. If * never see's and incoming rtp
packet, then it won't send an rtp packet (which usually contains some amount of
audio). Thus choppy audio in one direction.
So why cant * just play
I have a asterisk server with 6 Cisco ATA connected in SIP. My problem is
that one of them is dropping calls an I can't figure out what is the
problem; I had made a SIP DEBUG PEER 1088 that is the peer with the problem.
Any help will be appreciate
Thanks
Erick Weber
VoIP*CLI sip debug peer 1088
On 2005.02.16 11:20 Brian M. Arlinghaus wrote:
You could always run another T100P into a HylaFAX-run T1 fax modem.
That way you can use your T1 for faxing.
Could you explain a litter further? Thanks.
Well, you can do something like this:
T1 -- TE405P(1) -- Asterisk -- TE405P(2) -- Patton 2977
CVS Head 02/02/2005
from the CLI command line, the command
Agent logoff Agent/agentnum soft
does log the agent out, but does not generate any manager events. The
AgentLogoff and AgentCallbacklogin apps do generate such events.
Should the command line agent logoff also generate a manager event ?
I am using mysql sipfriends and can't seem to get the MWI to work. From what
I've read it seems this is not supported with that dynamic system, and
probably never will be.
I was thinking of just setting a cron job or something to check every minute
for voicemail and set our sip NOTIFY messages as
I get a ton of these messages, a pair every 4 or 5 mins.
Is it a problem?
I am wondering where they come from and if they are important.
I have a zaphfc card running in TE mode connected to a PBX.
Feb 16 20:23:04 epbw202 kernel: zaphfc: b channel buffer underrun on
card 0
Feb 16
Hi,
I'm reading a lot of stuff about callerid problems, but couldn't find any
logical explanation of Asterisk behaviour with callerid. When I receive
incoming call, caller info seems ok, but when transferred to local extension
via some macros, callerid gets to 'asterisk'. Does anyone know why and
Hi,
I'm trying to get capiECT working. I'd like to transfer call to another ISDN
router connected extension and free channel from router to Asterisk.
I get incoming call on CAPI and would liek to transfer it to dialed local
extension - 400 in this case:
[outbound-capi-local]
exten =
+++ Robert Augustyn [15/02/05 15:04 -0500]:
May I ask what you did?
robert
I'm sorry if it appears like Quoted text thats cause i cut pasted it rom my
mail to sangoma. but what i did is right there in the mail below Its fixed
and working great. :)
-Original Message-
From: [EMAIL
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