On Thu, 2008-10-09 at 12:51 +0800, Darren Murphy wrote:
Hi,
Can anybody point me to an online resource that will assist with
interpreting Asterisk log files?
I note that a similar question was asked in this forum some time ago
On Sun, 12 Oct 2008, sean darcy wrote:
Becasue of all the issues with fax over voip, we want to use pstn for
our fax machine, but not dedicate a line just to fax.
I'm thinking of having asterisk answer the pstn line, check for fax
tones, and route appropriately. In zapata ( chan_dahdi ) set
http://ftp.iq-labs.net/queue_log-
1.4/asterisk_queue_log_realtime_1.4.19.patch
This uses standardized realtime/mysql library from asterisk addons.
For it to support SQL inserts in 1.4, you would also need to apply
both patches from (1 for asterisk, another for asterisk-addons)
2008/10/11 Eric ManxPower Wieling [EMAIL PROTECTED]
snip
Where did you hear that media gateways filter one-way only?
Hi ,
Reading over this reply, it seems to me that having EC working in one
direction is not a so well known fact.
Could we say :
1. EC working in one direction is the
On Mon, Oct 13, 2008 at 09:57:33AM +0800, GNUbie wrote:
Hello Tzafrir,
On Mon, Oct 13, 2008 at 2:12 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
This means Zaptel gets silence from Asterisk.
What codecs are used? What do you see on 'sip show channels'?
I am using the following codecs:
Just a note that deserves to be reminded is
http://lists.digium.com/pipermail/asterisk-dev/2006-January/017774.html
If Generally speaking there is only one direction of echo cancellation
needed is true, EC works in one way ...
___
-- Bandwidth and
This one is also a must-read
http://lists.digium.com/pipermail/asterisk-dev/2007-May/027536.html
except that is the following scheme, I'm wondering if arrows and RX/TX
legends are coherent ...
What is written :
TX * --- *
TDM X Y
RX + --- -
What I would write :
RX
Hello Tzafrir,
On Mon, Oct 13, 2008 at 3:33 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
So the call is not established yet, right?
It is already. The CALLER hears the CALLEE's voice but the CALLEE
cannot hear the CALLER's voices.
This is not a temporary state?
What do you mean?
Regards,
Hi,
I have try again with your method but after that the phone reboot I have on
the screen phone displayed 'upgrading' with MAC address but the reset
process is stopped !
Thanks.
--
Salvatore.
- Original Message -
From: David Gibbons [EMAIL PROTECTED]
To: Asterisk Users
When the 'upgrading' process fails, it means that one or more of the required
files is missing from the TFTP root folder. Check the logs to see which file it
fails on, get that file and you should be good to go.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On
On Mon, Oct 13, 2008 at 9:04 AM, GNUbie [EMAIL PROTECTED] wrote:
Hello Tzafrir,
On Mon, Oct 13, 2008 at 3:33 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
So the call is not established yet, right?
It is already. The CALLER hears the CALLEE's voice but the CALLEE
cannot hear the CALLER's
I cann't view phone log files because, after reboot, the phone is stopped on
this screen ( 'upgrading' with MAC address) !
Regards.
--
Salvatore.
- Original Message -
From: David Gibbons [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
Here's some freaky stuff coming from Areski CDR tool:
101. 2008-10-13 03:41:23 DAHDI/1... 000 unknown 000 BackGround
silence/5 s
ANSWERED 00:20
102. 2008-10-13 03:11:30 DAHDI/1... 000 unknown 000 BackGround
silence/5 s
ANSWERED 00:21
103. 2008-10-13 02:41:23 DAHDI/1...
Hi Salvatore,
I'm talking about the tftp logs on the tftp server:
Something like 'tail -f /var/log/tftp' or 'tail -f /var/log/messages' should do
the trick.
Dave
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sasa
Sent: Monday, October 13, 2008 9:57
Hi,
It is not showing any CLI information even after I have placed that
NoOp(${CALLERID(all)}) function for debugging. Following message was
displayed in debug:
Executing [EMAIL PROTECTED]:3]NoOp(ZAP/6-1,)
What should I do since it is critical to have the callee number.
Please give some more
Hi,
I'm using a 4-port BRI card (b410p) to make and receive calls (via chan_misdn):
http://www.digium.com/en/products/digital/b410p.php
This card supposedly has hardware echo cancellation.
How can I check that echo cancellation is actually ON (taps, etc) on a given
misdn channel (like with the
Quoting Syed Nasruddin [EMAIL PROTECTED]:
Hi,
It is not showing any CLI information even after I have placed that
NoOp(${CALLERID(all)}) function for debugging. Following message was
displayed in debug:
Executing [EMAIL PROTECTED]:3]NoOp(ZAP/6-1,)
What should I do since it is critical
Hello Steve,
On Mon, Oct 13, 2008 at 9:52 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
If you are going to dismiss (the most probable) problem (NAT) without
posting configs, I am not sure how much help you will get, you will
probably be dismissed as well.
What particular configs are you looking
I am running 1.4.10.1 and I am getting garbled MOH from calls within the
same LAN with no firewall. Calls sound fine, but every 5-10 seconds the
MOH gets garbled. I am using the stock MOH files. Any ideas where/how
this could occur? There is no debug showing any issue with MOH. Thanks.
On Mon, Oct 13, 2008 at 10:49 AM, GNUbie [EMAIL PROTECTED] wrote:
Hello Steve,
On Mon, Oct 13, 2008 at 9:52 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
If you are going to dismiss (the most probable) problem (NAT) without
posting configs, I am not sure how much help you will get, you will
Hi
I am trying to get H323 to run on Asterisk, basically I had Asterisk running
so I followed this tutorial
http://astrecipes.net/index.php?n=286
and got h323 to run on my first server on the second server it is just
throwing the error:
ERROR:Failed to create H323 listener
The whole error is :
On Oct 13, 2008, at 9:29 AM, [EMAIL PROTECTED]
wrote:
IME: One-way audio problems are almost always casued by NAT gateways
and/or incorrect NAT settings in sip.conf and/or incorrect IP
address or
extenal proxy settings in the SIP phone.
And reinvite issues in particular. Those have been
How do we adjust zaptel and asterisk for CLI??. Is there some variable
to be set??.. kindly explain keeping in view your country settings this
will give me some hint.thanks
Syed Nasruddin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Phil
Reynolds
I am running asterisk 1.2.27 and it dead today. The following is the backtrace
of core file. Can anybody help me to identify what is the possible cause of
crash?
It seems the mysql connection causing problem in Thread 2. But I can not tell
what exactly happened.
This asterisk is using as ACD
Quoting Syed Nasruddin [EMAIL PROTECTED]:
How do we adjust zaptel and asterisk for CLI??. Is there some variable
to be set??.. kindly explain keeping in view your country settings this
will give me some hint.thanks
Bearing in mind that these settings are for UK BT (even on other
providers
Is the IP 650 sidecar compatible with asterisk?
If I pair it with the IP 650 phone, can I have more than 6 lines registered
w/ the server?
Jeremy Mann
Director of IT
Texas Health Management Group
Direct Line: 817-310-4956
Main Line: 817-310-4999
Fax: 817-310-4990
Email: [EMAIL PROTECTED]
Hi John,
On Mon, Oct 13, 2008 at 9:51 AM, Lee, John (Sydney)
[EMAIL PROTECTED] wrote:
http://ftp.iq-labs.net/queue_log-
1.4/asterisk_queue_log_realtime_1.4.19.patch
Haven't you forgotten this one? ;)
if you have applied everything correctly - queue_log file shoudln't
have any more lines
On Mon, Oct 13, 2008 at 10:19 AM, Jeremy Mann [EMAIL PROTECTED] wrote:
Is the IP 650 sidecar compatible with asterisk?
Yes. Our attendant phone is a 650 with three expansion modules
(sidecars). Asterisk can't really tell the difference. The sidecar
just gives the phones more buttons for lines or
On Monday 13 October 2008 10:29:17 gary wrote:
I am running asterisk 1.2.27 and it dead today. The following is the
backtrace of core file. Can anybody help me to identify what is the
possible cause of crash? It seems the mysql connection causing problem in
Thread 2. But I can not tell what
Hello Norman,
On Mon, Oct 13, 2008 at 11:02 PM, Norman Franke [EMAIL PROTECTED] wrote:
And reinvite issues in particular. Those have been the only one-way
audio problems I've experienced. Setting reinvite=no fixed everything
for me.
You mean, canreinvite=no? I already have done line on my
Hello Steve,
On Mon, Oct 13, 2008 at 10:59 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
First, drop firewall/iptables/selinux and try again.
I already turned off the firewall and I don't have SELinux on my
system and the problem is still there.
Regards,
GNUbie
On Monday 13 October 2008 01:25:12 am Gordon Henderson wrote:
On Sun, 12 Oct 2008, sean darcy wrote:
Becasue of all the issues with fax over voip, we want to use pstn for
our fax machine, but not dedicate a line just to fax.
I'm thinking of having asterisk answer the pstn line, check for
Hello,
Thanks for your replies.
We checked our sip.conf and we have canreinvite=no already. I agree it
could be a firmware issue. I will get another vendors phone hooked up to
the pbx before going crazy with support.
Thanks,
Neal
On Sun, Oct 12, 2008 at 6:14 AM, Vieri [EMAIL PROTECTED]
A packet trace will probably show exactly what is happening. Try:
tcpdump -nlXs 8192 -i eth0 port 5060
You should be able to see the SIP information going back and forth and
will probably show you that your NAT rules are applying when they
shouldn't. I agree with first turning off your
Might be a stretch, but does the Asterisk log show that the call was
answered? I had this problem when interfacing * with an NEC system to
do call parking pickup. The NEC would never give a dialtone (nor did
it give answer supervision) so * never knew the call got picked up so
audio only
- Original Message -
From: Tilghman Lesher [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, October 13, 2008 11:54 AM
Subject: Re: [asterisk-users] Need help for debuging
On Monday 13 October 2008 10:29:17
Hi,
Am Montag, den 13.10.2008, 10:00 +0800 schrieb GNUbie:
Hello Gordon,
On Mon, Oct 13, 2008 at 2:22 AM, Gordon Henderson
[EMAIL PROTECTED] wrote:
You mention the SIP phone being inside the LAN. Where is the Asterisk box?
It is the main gateway of the IP phones and my laptop to the
gary schrieb:
For some reason, I never received your reply nor my original post. That is
why I repost again. Can you repost your reply here?
http://lists.digium.com/pipermail/asterisk-users/2008-October/219985.html
Philipp Kempgen
--
http://www.das-asterisk-buch.de -
On Monday 13 October 2008 11:19:35 gary wrote:
- Original Message -
From: Tilghman Lesher [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, October 13, 2008 11:54 AM
Subject: Re: [asterisk-users] Need help for
On Oct 11, 2008, at 8:28 PM, Rob Hillis wrote:
Eric Chamberlain wrote:
Is there a particular reason you /can't/ register? It would seem
that
registration would provide the functionality you require, even if
you're
only making outbound calls.
In the case of a server like Asterisk,
On Mon, 13 Oct 2008, Anthony Messina wrote:
On Monday 13 October 2008 01:25:12 am Gordon Henderson wrote:
On Sun, 12 Oct 2008, sean darcy wrote:
Becasue of all the issues with fax over voip, we want to use pstn for
our fax machine, but not dedicate a line just to fax.
I'm thinking of having
Hi Eric,
Am Sonntag, den 12.10.2008, 18:06 -0700 schrieb Eric Fort:
resolve.conf and dns is working. The problem persists.
/var/log/asterisk/messages shows a few notices and warnings on
res_smdi.c, res_musiconhold.c, and usbradio.c. when I disable loading
of these in modules.conf asterisk
Hello Friends,
Can any one help me installing the CDR TOOL to integrate with freeradius
on freebsd? Any help will be highly appreciated.
Thanks
Mohit
DISCLAIMER: The information contained in this message (including any
attachments) is confidential and may be privileged. If
On Mon, Oct 13, 2008 at 12:37 PM, Eric Chamberlain [EMAIL PROTECTED] wrote:
We're developing the client and don't have control over the server,
which may or may not be Asterisk. Adding extra extensions isn't
possible.
Can OPTION packets be used to verify authentication?
--
Eric
When I sent Hangup using AGI application, Asterisk always return -1:
AGI Rx EXEC HANGUP
-- AGI Script Executing Application: (HANGUP) Options: ((null))
AGI Tx 200 result=-1
But only send Bye ( Sip message ) when the script is fisnish, not when
I send EXEC HANGUP. Why?
Thx
Hi to all, I have a project for Customer Relationship Management
interfaced with asterisk, I need send the CallerID to my application (via
http or tcp/ip), When the phone rings I need to launch a pop-up windows to
the Call Center Agent to display customer info, do you know a
framework/libraries
Can somebody please give a pointer to a complete neophyte (like me) on
text messaging, what product can I use to send and automatic text message to
a cell phone from within the asterisk dialplan? (the part of the dialplan I
have down, the part of the text message no)
Thanks
C. Savinovich
On Mon, Oct 13, 2008 at 12:50 PM, Gordon Henderson
[EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
On Mon, 13 Oct 2008, Anthony Messina wrote:
On Monday 13 October 2008 01:25:12 am Gordon Henderson wrote:
On Sun, 12 Oct 2008, sean darcy wrote:
Becasue of all the issues with fax over voip, we
Tim Panton from Phone From Here was able to implement this functionality
when he was at Mexuar so I would check with him.
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).
C. Savinovich wrote:
Can somebody please give a pointer to a complete neophyte (like me) on
text messaging, what product can I use to send and automatic text message to
a cell phone from within the asterisk dialplan? (the part of the dialplan I
have down, the part of the text message no)
Any thoughts?
http://www.websitemagazine.com/content/blogs/posts/archive/2008/10/13/go
ogle-analytics-track-phone-calls.aspx
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).
I mean is if someone know of an sms server or service that allows me to
send outgoing text messaging.
CS
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Pavel Jezek
Sent: Monday, October 13, 2008 2:11 PM
To: Asterisk Users Mailing List -
Sorry to hijack this thread...but I'm still looking for a G.722 capable
soft phone other than Eyebeam. I only need a handful of licenses. There
are a couple others available but only to OEMs in 1000 unit quantities.
Any ideas?
Michael
--Original Message Text---
From: Dean Collins
Date: Mon, 13
Hi,
i had the same problem with an 7970g. You should open one of the files,
tht end with .loads and compare the files on you tftp-server with the
list at the end of this file. Pay attention on case sensitive writing of
filenames.
I had to changed the jarXYZ.sbn to JarXYZ.sbn an the upgrade was
On Oct 13, 2008, at 11:21 AM, Dean Collins wrote:
Any thoughts?
http://www.websitemagazine.com/content/blogs/posts/archive/2008/10/13/google-analytics-track-phone-calls.aspx
It wouldn't be to hard to duplicate this with Asterisk. One could
export the entire call path through an IVR or
On Oct 13, 2008, at 11:28 AM, C. Savinovich wrote:
I mean is if someone know of an sms server or service that allows
me to
send outgoing text messaging.
Are you sending SMS to known users or to any mobile phone user?
If you are sending to a fixed user base, track down the email to SMS
Ricardo Melendez wrote:
Hi to all, I have a project for Customer Relationship Management
interfaced with asterisk, I need send the CallerID to my application
(via http or tcp/ip), When the phone rings I need to launch a pop-up
windows to the Call Center Agent to display customer info, do
I use the 'generic' file in Postfix to map an email address that is not in
use to someone's text messaging address. It'd be [EMAIL PROTECTED]
ie: [EMAIL PROTECTED] Then, any email that gets sent to
[EMAIL PROTECTED], will get automatically sent to that person's phone.
On Mon, Oct 13, 2008 at 3:14
I am trying to send text messaging to one caller, maybe about 40 of those
per day, whose phone service have expired. All these callers are calling
from their cell phones, and I have their caller ids. I will like to send
each of them an individual text message (not an email) saying you
Are you sending SMS to known users or to any mobile phone user?
If you are sending to a fixed user base, track down the email to SMS
gateways for their carriers. Then sending an SMS is no different than
sending an e-mail.
If it's for something really important this might not be the
Gordon Henderson wrote:
On Sat, 11 Oct 2008, Jorge Mendoza wrote:
I founded this behaviour in the past. When the CO provides reversal
polarity and the FXO port is configured to ignore polarity events, then
a reversal polarity could be detected as ringing if the
hardware/software is not
Hi,
I'm in the process of setting up Asterisk in a SOHO environment using ISDN for
trunking. More specifically a BRI 2B+D circuit where one SPID is used for the
business and the other is used for personal. The circuit already exists, but
is presently being interfaced to POTS phones via a TA.
I had considered something like this as well, but was convinced to go
another direction.
I wrote something up about it at the time.
http://www.smallnetbuilder.com/content/view/30444/84/
Michael
--Original Message Text---
From: Wilton Helm
Date: Mon, 13 Oct 2008 14:44:26 -0600
Hi,
I'm in
On Mon, 13 Oct 2008, Jorge Mendoza wrote:
Gordon Henderson wrote:
On Sat, 11 Oct 2008, Jorge Mendoza wrote:
I founded this behaviour in the past. When the CO provides reversal
polarity and the FXO port is configured to ignore polarity events, then
a reversal polarity could be detected as
On Sat, 2008-10-11 at 10:09 +0200, Benny Amorsen wrote:
Tilghman Lesher [EMAIL PROTECTED] writes:
exten = [0-9*#+].,...
If that does not work, that is a bug and needs to be reported as such.
Sadly that matches *james and 9foo...
It would be nice if you could use normal regexes (e.g.
On Sunday 12 October 2008 04:15:02 Olivier wrote:
2008/10/11 Eric ManxPower Wieling [EMAIL PROTECTED]
Handsets use a 4-wire connection. Handsets with the the volume turned
up could cause a form of echo as the microphone picks up the ear piece
audio (I call this acoustic echo). Everything
I need to monitor the states of my T1/PRI Zap channels. Specifically, I
need to be able to programmatically determine whether a channel is
unused, carrying an inbound call, or carrying an outbound call.
Using the manager interface, I can easily tell whether a Zap channel is
used or not by
I'm trying to get VoiceMailMain() to work properly, but it refuses. : (
I am using IMAP_STORAGE, which is functioning fine now... My
voicemail.conf user line:
6000 = 1234,Brendan's Mailbox,,,[EMAIL PROTECTED]|
imappassword=password
6000 = d,Brendan Martens
My voicemail extension in
I have done this. Why BRIs exist in the US is beyond me. If you can, don't
go with BRI.
Who is the carrier. There is someone on the list that will tell you it is
impossible unless you use his code, which is not true.
Thanks,
Steve Totaro
On Mon, Oct 13, 2008 at 4:57 PM, Michael Graves [EMAIL
On Mon, Oct 13, 2008 at 5:22 PM, Kevin DeGraaf [EMAIL PROTECTED] wrote:
I need to monitor the states of my T1/PRI Zap channels. Specifically, I
need to be able to programmatically determine whether a channel is
unused, carrying an inbound call, or carrying an outbound call.
Using the
On Mon, 2008-10-13 at 17:37 -0400, Steve Totaro wrote:
I have done this. Why BRIs exist in the US is beyond me.
Much of the idea's behind ISDN are hopelesly outdated, except for one:
With POTS, the analogue/Digital conversion is done some miles away, in
the your local number exchange, and the
I had converations with both Pika and Xorcom wherein the thought that
it should be possible using their interface hardware. There might be
some software changes to be made in their drivers, but BRI should be
usable in the US.
I abandoned the idea for being more expensive when all costs are
On Mon, Oct 13, 2008 at 5:53 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
On Mon, Oct 13, 2008 at 5:22 PM, Kevin DeGraaf [EMAIL PROTECTED] wrote:
I need to monitor the states of my T1/PRI Zap channels. Specifically, I
need to be able to programmatically determine whether a channel is
Hi,
I am new user on asterisk (for that matter linux) and i have lot of embedded
programming experience. We have a new project from our client, to design a
box that takes the telelphone line as input and route the line to the
respective user with different ring tones. The box should be programmed
The documentation is in my head, two solid days worth.
The issue is the SPID code that Marcin Pyco claimed he had the only code,
and way to make it work in the US..
You may need this code if you are using SPIDs to route calls. In my
situation, they were just a hunt group, two BRIs, and I was
I'm in the process of setting up Asterisk in a SOHO environment using =
ISDN for trunking. More specifically a BRI 2B+D circuit where one SPID =
is used for the business and the other is used for personal. The =
circuit already exists, but is presently being interfaced to POTS phones =
via
On Sun, Oct 12, 2008 at 11:18:40AM +0100, Gordon Henderson wrote:
On Thu, 9 Oct 2008, Mike wrote:
I'm guessing this lamp is on an ordinary analogue phone you have?
Yeah, this is a bog standard 9 quid analogue phone.
OK. A bit convoluted this as I'm not local to the PBX, but an IAX trunk,
Steve Murphy [EMAIL PROTECTED] wrote:
People have voiced this before; but the cut-down version of RE's that
the matching algorithms allow are fairly fast, both in the new and
the old pattern matching algorithms.
Steve
Your explanation is clear and it seems like a good design choice to
On Mon, Oct 13, 2008 at 04:55:01PM -0500, Michael Graves wrote:
I had converations with both Pika and Xorcom wherein the thought that
it should be possible using their interface hardware. There might be
some software changes to be made in their drivers, but BRI should be
usable in the US.
Or
exten = +13129842314,1,Noop(Happy match!)
or
exten = _+1NXXNXX,1,Noop(Happier match!)
Karl Fife wrote:
Steve Murphy [EMAIL PROTECTED] wrote:
People have voiced this before; but the cut-down version of RE's that
the matching algorithms allow are fairly fast, both in the new and
the old
On Mon, Oct 13, 2008 at 06:51:47PM +0200, Karsten Wemheuer wrote:
Hi Eric,
Am Sonntag, den 12.10.2008, 18:06 -0700 schrieb Eric Fort:
resolve.conf and dns is working. The problem persists.
/var/log/asterisk/messages shows a few notices and warnings on
res_smdi.c, res_musiconhold.c,
Yes, I certainly applied the patch in
http://ftp.iq-labs.net/queue_log-1.4/asterisk_queue_log_realtime_1.4.19.
patch
Just to double-check, there is only one patch in this URL which is
main/logger.c
By the way, did you see anything wrong with my config files?
/etc/asterisk/extconfig.conf
if you have applied everything correctly - queue_log file shoudln't
have any more lines (except init when restarting asterisk).
Thanks Atis.
I see what you are saying. In the patch for logger.c,
The code to write to mysql is there except that we need to perform
ast_check_realtime(queue_log).
Im a 3-days-asterisk-newbie. In 3 weeks, I must have a PBX installed in
a new office of ours: Panasonic or Asterisk. Asterisk would be, if I can
emulate some Panasonic functions on Asterisk fast, to convince the
executives.
What I have done until now: Bought 1 Linksys pap2 (2 FXS), 1 Linksys
With ISDN, the conversion is done in your phone
Exactly. Or in the case of Asterisk, it is a 4 wire digital right into the
switch--no degradation. Even converting back and forth between analog and
digital multiple times compromises quality. Try doing a dial-up modem across
such a path. The
I have done this.
Good
Why BRIs exist in the US is beyond me.
I'm not sure why you say that. It is the only way I know of two get two
digital voice grade circuits at prices competitive with POTS. The better
question is why the LECs used such poor judgment when they introduced this.
Most
The card I have has no name but is based on the Winbond W6692CF chip and ships
with RVS, which I think is for Windows and of no use to me. I'm not sure about
whether it is supported by DAHDI or not.
Wilton___
-- Bandwidth and Colocation Provided by
Rodolfo Alcazar Portillo wrote:
Im a 3-days-asterisk-newbie. In 3 weeks, I must have a PBX installed in
a new office of ours: Panasonic or Asterisk. Asterisk would be, if I can
emulate some Panasonic functions on Asterisk fast, to convince the
executives.
Asterisk is more featured than
Can somebody please give a pointer to a complete neophyte (like me) on
text messaging, what product can I use to send and automatic text message to
a cell phone from within the asterisk dialplan? (the part of the dialplan I
have down, the part of the text message no)
Thanks
C. Savinovich
1)
At 9:37 AM -0700 2008/10/13, Eric Chamberlain wrote:
On Oct 11, 2008, at 8:28 PM, Rob Hillis wrote:
Eric Chamberlain wrote:
Is there a particular reason you /can't/ register? It would seem
that
registration would provide the functionality you require, even if
you're
only making
Paul Douglas Franklin wrote:
When calling out to another phone, they always identify themselves
correctly. But sometimes they will respond to the wrong incoming
calls. (By respond, I mean that the phone rings and if someone picks up
the receiver, the call then goes thru.) For example,
Tom Moore wrote:
Hi guys,
Does the Aastra line of phones work with dns srv records?
I'm trying to get my 8133i to do this and in the settings it asks for ip
addresses of registration and proxy servers.
Does this mean that it will not just let me put the domain name in like
other devices I
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