From: Lane
Sent: Wednesday, December 29, 2004 6:00 PM
I subscribed to this list for about two months before I began posting, so
I've
got a buttload of email to sift through ... I'm doing this BEFORE I flood
the
list with my inane questions ...
But here goes:
I read a reply from one
Yup...Check out twisted's patch
http://bugs.digium.com/bug_view_page.php?bug_id=0002955 . It does almost
exactly what you're looking for.
Don't forget to reply to the bug if the patch works for you
-Brian
On Wed, 29 Dec 2004, Paul Rodan wrote:
Is there a way to monitor a call mid-stream?
Thnx!
--
Andrew McRory - President/CTO
Linux Systems Engineers, Inc. - http://www.linuxsys.com
Located in beautiful Tallahassee, Florida
Office 850-224-5737
Office 850-575-7213
Mobile 850-294-7567
___
Asterisk-Users mailing list
Accounts by itself would be useful.
-Original Message-
From: David Boyd [mailto:[EMAIL PROTECTED]
Sent: 30 December 2004 00:52
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Final call for departments
HOw about :
development
Dave
On Wed,
On December 29, 2004 22:35 pm, Andrew Kohlsmith wrote:
Well it is hard to go back to a specific configuration since I have
used the system to test the rpm packages I compile.
Yikes.
Yep. But there is only one way to know for sure that a new package is
working. I have had much success in
Since Friday is the last day I can accept new requests fro this run, I
wanted to post to the list what I have as of about 1:30am Pacific Time
30 Dec. This way people have Thursday to make any additions /
suggestions and then Friday, I will send what I have on.
The list is getting longer so
Hallo Thomas Niesel
On Wed, 29 Dec 2004 22:03:05 +0100 you wrote:
Hi Folks, hi Steve
I get following error on loading app_rx/txfax.so:
...WARNING[10458]: loader.c:258 ast_load_resource:
/usr/lib/asterisk/modules/app_rxfax.so: symbol errno,
version GLIBC_2.0 not defined in file libc.so.6
For threeway calling (analog phone) I just hit the
flash button get a dial tone, dial the number and hit
the flash key again.
It doesn't work for me when I'm using asterisk. No problems without it. So
is my hardware broken or my dialplan? When you hit the flash key is anything
I've been thinking about taking steps to make my * server more
reliable. In particular I'd like to have it automatically start * after
a power loss. Can anyone here provide some guidance as to how to
accomplish this. Keep in mind that I have a TDM400p that needs a couple
of modprobe commands
I have installed Digium TDM04B with the latest CVS. However I have
encountered following problems:
1. When it dials out, many times the digits are not properly recognized
by telco as I hear the announcement please check the number and dial
again although I see on the screen that the dialed
Hi all,
I have some doubts concerning the way asterisk records
calls using the Monitor command.
I ´ve done some jitter and packet loss tests in a such
way that, from asterisk 1, I send a file to asterisk 2
and record this file in asterisk 2 using the Monitor
command. To simulate the jitter and
Bruno Hertz is believed to have said:
Hi Aldo
don't know about Suse, but I have a working setup with asterisk 1-0
stable, chan_capi 0.3.5 and fcpci-suse9.1-3.11-02 on Debian Sarge,
though not prepackaged but all self compiled.
Looking at your log messages, chan_capi obviously is installed, but
Welcome to the Asterisk users community!
Asterisk.org is a fast moving project. New code is added every day.
Asterisk is the leading Open Source Telephony platform,
with support both for classical telephony and IP telephony.
Our community is also growing
In extensions.conf
[smsdial]
exten = _X.,1,SMS(${CALLERIDNUM},,${EXTEN},${CALLERIDNAME})
exten = _X.,2,SMS(${CALLERIDNUM})
exten = _X.,3,Hangup
[local]
exten = 07,1,wait(1)
exten = 07,2,Answer
exten = 07,3,GotoIf($[foo${CALLERIDNUM} = foo]?12:4)
exten =
I have more FXO ports on TDM400's than I have PSTN lines available for
testing. When all the lines were used up (the FXO ports are all in zap
group 2), and I did a Dial(Zap/G2/${phone},5) it thinks the Dial succeeded
even though there is neither line voltage nor dial tone. Can at least the
Hi Eric,
Thanks every body that answered about this problem.
About change de default SIP port (5060), I tried it at first and the UAC
could authenticate but when I made a call and another side pick the phone up
DSLink 200E freeze again.
ie. there wasn't any port 5060 on transactions.
I will have
Paul Fielding wrote:
I've just picked up a pair of IAXy devices. They work fine except
that they keep going offline. As in, I plug it in, it connects to
Asterisk, I can dial and phone and all is dandy. Then, maybe 12h
later, maybe 24, maybe 36, maybe 48, I'll either try to phone the
device
Hi *,
Would anyone know about solutions that let you use a VoDSL connection
without using and IAD? VoDSL is starting to come from many vendors now
in The Netherlands and it seems silly to have an IAD that turns VoDSL
into POTS/ISDN to connect it to a card in the Asterisk box that turns it
into
On Thu, 30 Dec 2004, Guild Jackson wrote:
Hearing the sent file with a handset, without
recording, I listen a deteriorated file different from
the recorded one.
My question is:
Is asterisk able to detect the packet loss and modify
the file recorded in a such way that compensate this
1. When it dials out, many times the digits are not properly recognized
by telco as I hear the announcement please check the number and dial
again although I see on the screen that the dialed number is correct.
Had the same problem with an older central office and the 'w' fixed it.
I
On Thu, 30 Dec 2004, Gary wrote:
On Thu, 30 Dec 2004 00:12:51 -0700, Paul Fielding wrote:
I've just picked up a pair of IAXy devices. They work fine except that they
keep going offline. As in, I plug it in, it connects to Asterisk, I can
dial and phone and all is dandy. Then, maybe
Up until last night, I could run:
asterisk -vvvr
as root to connect to a running * session and have the verbosity set to 3.
Last night, however, I updated to CVS-v1-0-12/29/04-16:47:20 and the behavior
is different. Now the -v flags don't seem to make a difference, I have to
issue:
Of course I try canreinvite=yes
- Original Message -
From: Matthew Boehm [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, December 29, 2004 6:46 PM
Subject: Re: [Asterisk-Users] What happened with the
from voip-info wiki
Asterisk automatic daily restart
To automatically restart Asterisk you can add something like this to cron
# Restart Asterisk PBX once a day to prevent any problems from piling up
10 7 * * * root /usr/sbin/asterisk -rx restart now /dev/null 21
or
10 7 * * * root
On Wed, 29 Dec 2004 01:51:16 -0800, Alspach Family
[EMAIL PROTECTED] wrote:
I am getting ready to submit a list of department names to be recorded.
This is what I have so far:
QA or Quality Assurance.
-Chuji
___
Asterisk-Users mailing list
Hi all,
Diax version 0.9.9f is ready to be tested by the interested people.
You can download it for the moment from the following location only:
http://www.geocities.com/tdanro/diax/diax099f.zip
Please do not use older config files with 0.9.9f !!!
You have some command line options now for
Greg - Cirelle Enterprises wrote:
from voip-info wiki
Asterisk automatic daily restart
To automatically restart Asterisk you can add something like this to cron
# Restart Asterisk PBX once a day to prevent any problems from piling up
10 7 * * * root /usr/sbin/asterisk -rx restart now /dev/null 21
Hello,
I'm actually trying
to connect an asterisk PBX with 2 E100P card to an alcatel 440, but I'm facing
some problems.
In fact, i had one
E100P connected to the public PSTN and the other one connected to the
Alcatel.
I can receive call
from the PSTN without any problems but I can't
At 09:19 AM 12/30/04, you wrote:
Greg - Cirelle Enterprises wrote:
from voip-info wiki
Asterisk automatic daily restart
To automatically restart Asterisk you can add something like this to cron
# Restart Asterisk PBX once a day to prevent any problems from piling up
10 7 * * * root
No.. it's not that unstable. Some people are just paranoid. With my
X100p's I do notice that caller id gives me trouble after about a
week. Could just be in my head though.
On Thu, 30 Dec 2004 08:52:15 -0500, Greg - Cirelle Enterprises
[EMAIL PROTECTED] wrote:
from voip-info wiki
Asterisk
On Thu, 2004-12-30 at 08:52 -0500, Greg - Cirelle Enterprises wrote:
from voip-info wiki
Asterisk automatic daily restart
To automatically restart Asterisk you can add something like this to cron
# Restart Asterisk PBX once a day to prevent any problems from piling up
10 7 * * * root
I do about 500 calls per day on average volume and about 750 on heavy
volume and find it necessary to run a logger rotate every other day...
other then that I can go on for a couple weeks until I need a full
reboot.
___
Asterisk-Users mailing list
This is the age old difference between Microsoft environments and
Unix/Novell environments.
I like to joke that Microsoft uptime is measured in hours
Unix/Novell is always in years,months, and days.
Although, I have to admit that Win 2k (server) and XP have substantially
improved uptime and
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Steven Critchfield
Sent: Thursday, December 30, 2004 7:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Is asterisk that unstable
On Thu,
On Thu, 2004-12-30 at 09:50 -0500, Luke Catranis wrote:
I do about 500 calls per day on average volume and about 750 on heavy
volume and find it necessary to run a logger rotate every other day...
other then that I can go on for a couple weeks until I need a full
reboot.
Oddly enough, My logs
On Thu, 2004-12-30 at 07:58 -0700, Damon Estep wrote:
\ Right now this is the uptime from my main PBX.
phone*CLI show uptime
System uptime: 21 weeks, 21 hours, 16 minutes, 50 seconds
Last reload: 1 week, 1 day, 15 hours, 53 minutes, 40 seconds
As of this message, we have run about
I just make it a habit, the only issues I run into are after an IAX2
gridlock and my log files get filled up quickly...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Critchfield
Sent: Thursday, December 30, 2004 10:03 AM
To: Asterisk Users
Hi all,
I was wondering how the easiest way to restrict the users ability to set
caller ID would be ?
I have some users that uses IAX to connect with me. multiple numers via
iax.
on outgoing calls I would like the user to only be able to set his
range of numbers on the outgoing calls.
Is
ALL,
As a community is there anything we can do to help with communications
to the Tsunami area ? we all sit on top of a welth of knowledge on
communications can we use this to help these area's in any way? IE
free sip calls , maybe there are * users in the area that we can send
US calls to ?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Steven Critchfield
___
Any analog FXO or FXS interfaces in that box?
Of course not. FXO and FXS interfaces are for small
deployments. We only
- Original Message -
From: Helder Rogério
[MICROREDE]
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Thursday, December 30, 2004 3:13 PM
Subject: Open ports on router in front of asterisk
Hi,
what are the ports that I must have open to
Asterisk work correctly
Use a separate context for the outbound calls for that customer, check
the caller ID in the dialplan before completing an outbound call using a
PATTERN MATCH, and IF the pattern does not match the pattern of the
customers numbers GOTO a step that sets the caller ID to the customers
main phone
[EMAIL PROTECTED] wrote:
I like to joke that Microsoft uptime is measured in hours
Unix/Novell is always in years,months, and days.
It's not just you. A while back Microsoft was running a TV ad
where a server was bragging that it was so reliable that it hadn't
even seen the sysadmin for DAYS.
Hi,
I've been loosing my mind with NAT and read that
IAX doesn't have problems about nat.
Does anyone knows about hadware (routers and etc)
support IAX?
Best regards
helder
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
I do about 500 calls per day on average volume and about 750 on heavy
volume and find it necessary to run a logger rotate every other day...
other then that I can go on for a couple weeks until I need a full
reboot.
How do you rotate your logs?
--
No virus found in this outgoing message.
Hello
I have done the following test-network:
IP-Phone = ASTERISK == ISDN PSTN Phone
(SIP) +
SER
When I'm calling from the PSTN phone to the IP (SIP) phone:
I cannot get ANY DTMF from PSTN, they
I have a Sipura 3000, apparently configured correctly, when incoming
calls arrive on the telco port they arrive properly on the Asterisk
system - however they don't get routed properly. The Asterisk message:
Dec 30 07:47:16 NOTICE[2745]: chan_sip.c:7486 handle_request: Failed to
authenticate
Does anyone have some decent Nagios scripts out there that do more than
monitor the proc itself? Rather than reinvite the wheel, figured I'd
ask. I already saw the one on the wiki.
Matt
___
Asterisk-Users mailing list
The device may also be doing RTP fixup, I guess. SIP uses RTP for the
audio.
Rodrigo P. Telles wrote:
Hi Eric,
Thanks every body that answered about this problem.
About change de default SIP port (5060), I tried it at first and the UAC
could authenticate but when I made a call and another side
Hi Randy,
Randy MacKay wrote:
I do about 500 calls per day on average volume and about 750 on heavy
volume and find it necessary to run a logger rotate every other day...
other then that I can go on for a couple weeks until I need a full
reboot.
How do you rotate your logs?
I have made a script
Damon Estep wrote:
Use a separate context for the outbound calls for that customer,
check the caller ID in the dialplan before completing an outbound
call using a PATTERN MATCH, and IF the pattern does not match the
pattern of the customers numbers GOTO a step that sets the caller ID
to the
System uptime: 6 weeks, 1 day, 22 hours, 37 minutes, 55 seconds
Last reload: 48 seconds
Verbosity is atleast 3
System uptime: 7 weeks, 19 hours, 19 minutes, 48 seconds
Last reload: 41 seconds
Verbosity is atleast 3
System uptime: 7 weeks, 4 days, 9 hours, 25 minutes, 33 seconds
Last reload: 36
Nicolas FOURNIL wrote:
Hello
I have done the following test-network:
IP-Phone = ASTERISK == ISDN PSTN Phone
(SIP) +
SER
When I'm calling from the PSTN phone to the IP (SIP) phone:
I cannot get ANY
This file, which was attached to the message titled Asterisk-Users Digest, Vol
5, Issue 407 by [EMAIL PROTECTED] and was quarantined on 12/30/2004 11:01
AM, has been released.
NOTE: If AutoProtect is enabled, then this restored attachment will be
rescanned during the restore. If the
--Original Message Text---
From: Helder Rogério [MICROREDE]
Date: Thu, 30 Dec 2004 15:32:59 -
Hi,
I've been loosing my mind with NAT and read that IAX doesn't have problems about nat.
Does anyone knows about hadware (routers and etc) support IAX?
Best regards
helder
Steven P. Donegan wrote:
I have a Sipura 3000, apparently configured correctly, when incoming
calls arrive on the telco port they arrive properly on the Asterisk
system - however they don't get routed properly. The Asterisk message:
Dec 30 07:47:16 NOTICE[2745]: chan_sip.c:7486 handle_request:
Logger rotate from cli
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Randy
MacKay
Sent: Thursday, December 30, 2004 10:49 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Is asterisk that unstable
I
Hi,
I've been loosing my mind with NAT and read that IAX doesn't have problems
about nat.
Does anyone knows about hadware (routers and etc) support IAX?
Best regards
helder
Well, in fact, IAX doesn't needs an ALG (application level gateway) unlike SIP,
IRC or
FTP.
It uses a one normal
On Thu, 2004-12-30 at 08:29 -0700, Damon Estep wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Steven Critchfield
___
Any analog FXO or FXS interfaces in that box?
Of course not.
what was wrong with logrotate?
On Thu, 2004-12-30 at 10:57 -0500, Matt Gibson wrote:
Hi Randy,
Randy MacKay wrote:
I do about 500 calls per day on average volume and about 750 on heavy
volume and find it necessary to run a logger rotate every other day...
other then that I can go on for a
Here's some advice to myself. Why don't I check out the documentation before
I post. I think I'll bear that in mind in the future. Thanks me.
http://www.voip-info.org/wiki-Asterisk+auto-dial+out+deliver+message
- Original Message -
From: Gary Ruddock (Swift Drinks) [EMAIL PROTECTED]
To:
At 11:00 AM 12/30/04, you wrote:
I wouldn't say it's unstable... these boxes all run res_perl and reload
100's of times a day. It all depends on if you know what the hell you're
doing.
bkw
why are they reloading 100's of times a day??
greg
___
Steven Critchfield wrote:
Does any business outside of a ISP still use analog modems? I would
think internet connections and good encryption would be the norm for
those needs than an analog modem.
Funny story, and not really related, but I was talking to a guy who
works upstairs from our office
Hello, anytime I make an IAX2 call to another peer,
I am collecting CDR records which are divided into small files, one for each
accountholder customer that makes the calls.
I have records of this nature:
""123456","1234567890","IAX2/[EMAIL PROTECTED]/5","2004-12-30
22:17:07","2004-12-30
I have about 40 of these in production with Asterisk, send me an email off
list with your sip.conf file and you extensions.conf file and I will help:)
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Deepak
Malhotra
Sent: Wednesday,
Hello!
Am Mittwoch, 29. Dezember 2004 23:46 schrieb Alvaro Parres:
I want to know if there is any console o secretarial hardphone that
works with asterisks.
I mean a phone in witch i can see the state of the extensions, the
phone lineas, etc. Can hold o transfer easly a call, etc.
Kristian Kielhofner wrote:
Steven P. Donegan wrote:
I have a Sipura 3000, apparently configured correctly, when incoming
calls arrive on the telco port they arrive properly on the Asterisk
system - however they don't get routed properly. The Asterisk message:
Dec 30 07:47:16 NOTICE[2745]:
Justin Carlson wrote:
what was wrong with logrotate?
nothing, i just like doing things my own way :)
this makes use of the asterisk rotate feature, and my own daily log
rotating. meh. to each their own :)
matt
___
Asterisk-Users mailing list
My PSTN line doesn't allways hang up properly after it goes to voicemail.
The problem occurs when a caller hangs up during the initial greeting.
Even though the hangup accured, voicemail continues to record, usually a
fast busy and/or a teleco generated please hangup now message. After the
For efficiency reliability, when SIP transmits DTMF as non-audio data, it uses RFC2833 or INFO.
My question is - (not knowing much about IAX2) - when IAX2 transmits DTMF as non-audio data - is it also using RFC2833 and/or INFO, or it it using some other IAX2-specific mechanism with its own
My incoming PSTN line is configured to ring multiple extensions and
eventually fall trough to voicemail if the call goes unanswered. If a SIP
phone gets picked up just before voicemail should kick in, the call quite
often goes to the phone but voicemail happens as well, the greeting plays
and the
On Thu, 30 Dec 2004 [EMAIL PROTECTED] wrote:
I was wondering how the easiest way to restrict the users ability to set
caller ID would be ?
I have some users that uses IAX to connect with me. multiple numers via
iax.
on outgoing calls I would like the user to only be able to set his
Matt Gibson wrote:
nothing, i just like doing things my own way :)
this makes use of the asterisk rotate feature, and my own daily log
rotating. meh. to each their own :)
matt
Know you can make your own wheel before you drive someone else's car.
This sums up the way I live - kind of goes along
Steven Critchfield wrote:
On Thu, 2004-12-30 at 08:29 -0700, Damon Estep wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Steven Critchfield
___
Any analog FXO or FXS interfaces in that box?
Of course
On Thu, Dec 30, 2004 at 01:38:43PM +1100, Adam Goryachev wrote:
On Thu, 2004-12-30 at 01:48, Steve Underwood wrote:
Hi Adam,
You must be using a prehistoric GCC. Before 3.0, GCC didn't understand
this C99 construct.
Hmmm, well I have:
gcc version 2.96 2731 (Red Hat Linux 7.3
On Thu, 30 Dec 2004 09:04:32 -0800, Steven P. Donegan wrote:
Kristian Kielhofner wrote:
Steven P. Donegan wrote:
I have a Sipura 3000, apparently configured correctly, when incoming
calls arrive on the telco port they arrive properly on the Asterisk
system - however they don't get routed
Is your X100P set for loop start or Kewl Start? I am betting loop start,
try changing to ks instead.
Lyle
- Original Message -
From: Adi Linden [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, December 30, 2004 11:08 AM
Subject: [Asterisk-Users] Voicemail and
Hi Matt,
Thanks for the information. I didn't mean for you to get beat up on this;-)
I'm still learning linux, so your information is very helpful and I'm now
going to try and figure it out. It will be a good challenge.
I have been able to locate very little information about logs, so your
Steven P. Donegan wrote:
Kristian Kielhofner wrote:
Steven P. Donegan wrote:
I have a Sipura 3000, apparently configured correctly, when incoming
calls arrive on the telco port they arrive properly on the Asterisk
system - however they don't get routed properly. The Asterisk message:
Dec 30
Michael Graves wrote:
On Thu, 30 Dec 2004 09:04:32 -0800, Steven P. Donegan wrote:
Kristian Kielhofner wrote:
Steven P. Donegan wrote:
I have a Sipura 3000, apparently configured correctly, when incoming
calls arrive on the telco port they arrive properly on the Asterisk
system -
Kristian Kielhofner wrote:
Steven P. Donegan wrote:
Kristian Kielhofner wrote:
Steven P. Donegan wrote:
I have a Sipura 3000, apparently configured correctly, when
incoming calls arrive on the telco port they arrive properly on the
Asterisk system - however they don't get routed properly. The
On Thu, 30 Dec 2004, Lyle Giese wrote:
Is your X100P set for loop start or Kewl Start? I am betting loop start,
try changing to ks instead.
Lyle
This is what I have in /etc/asterisk/zapata.conf so it should be Kewl
Start.
[channels]
; X100P
signalling=fxs_ks
echocancel=yes ; You
Hmmm I could certainly see that being the issue. If it is the issue,
though, then I think it's something that needs to be addressed.
In my opinion, Digium needs to address it, as well as the whole provisioning
via cli thing. I know Asterisk itself is a CLI oriented piece of software,
but
Well I am about to reserve a small padded room so I can bounce off the
walls without inflicting tooo much damage... Nothing is making sense at
this point. I tried several releases last night before settling on the
latest CVS (seemed to work the best). Asterisk was running GREAT for the
first few
Steven P. Donegan wrote:
The Sipura has registration entries in sip.conf for both lines - and
from my earlier post appears to register just fine. I'm still clueless
on the failure originally reported.
Steven,
So, of the 1001, 1002, 1003, etc. one of those in the PSTN line?
Confusing at best.
I think this is a great idea...I have up to 5000 minutes I could donate, but
unfortunetly my SIP service only allows calls to/from US and Canada.
Gabe
- Original Message -
From: Jason p [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, December 30, 2004 7:18 AM
Hello,
Hope this isn't TOO much of a newb question...
I just created a new WBEL server with a fresh install of asterisk.
When I try to load asterisk, it dies with some cryptic error messages.
I've googled for them, but haven't found anything helpful.
If anyone can point me in the right
Hi,
Tzafrir Cohen schrieb:
spandsp builds fine on Sarge. Anybody needs debs?
It does?
I ITPed it a while ago, but placed it somewhat lower on my list when I
saw it needed libtiff internals. I have debs for sarge that depend on
libtiff3g, however I could not get it to work reliably with the more
Hi All,
Channels 25-28 on a customers PBX are regular Zaptel FXO cards that
are hooked into 4 incomming phone lines. They are all in a group to do
automatic rollover for outgoing calls (if channel 25 is being used,
dial on channel 26, etc.).
Sometimes when a user is dialing a number, instead of
On Thu, 2004-12-30 at 10:22 -0700, Michael Welter wrote:
Steven Critchfield wrote:
On Thu, 2004-12-30 at 08:29 -0700, Damon Estep wrote:
Only for small deployments? How do you interface with your fax machines?
analog alarm systems? pc modems?
I think most alarm companies continuously
Anyone? :-)
- Original Message -
From: Matt Klein [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, December 29, 2004 7:37 PM
Subject: Re: [Asterisk-Users] Cisco 7690 Voicemail Problem
a faint scratching sound
Brent Goran wrote:
For efficiency reliability, when SIP transmits DTMF as non-audio data,
it uses RFC2833 or INFO.
My question is - (not knowing much about IAX2) - when IAX2 transmits
DTMF as non-audio data - is it also using RFC2833 and/or INFO, or it it
using some other IAX2-specific mechanism
Paul Fielding wrote:
Hmmm I could certainly see that being the issue. If it is the
issue, though, then I think it's something that needs to be addressed.
In my opinion, Digium needs to address it, as well as the whole
provisioning via cli thing. I know Asterisk itself is a CLI oriented
I can also Donate minutes ,, please contact if in that area are
Asterisk users with Satellite, to interconnect.
also if someone needs help I am available as far I can.
regards
Humberto
On Thu, 30 Dec 2004 10:56:50 -0800, Gabriel Afana [EMAIL PROTECTED] wrote:
I think this is a great idea...I
On 30/12/2004 19:01, Paul A Brown wrote:
Anyone? :-)
If you turn down the volume on the phone slightly (Just one or two
units) it goes away.
I assume the output volume is overloading the phone and the DSP isn't
clever enough to clip it. A longer term solution would be to boost the
gain of
On Thu, 30 Dec 2004, Brent Goran wrote:
My question is - (not knowing much about IAX2) - when IAX2 transmits
DTMF as non-audio data - is it also using RFC2833 and/or INFO, or it it
using some other IAX2-specific mechanism with its own name?
Yep - IAX's protocol is quite different from
[EMAIL PROTECTED] wrote:
Paul Fielding wrote:
Hmmm I could certainly see that being the issue. If it is the
issue, though, then I think it's something that needs to be
addressed.
In my opinion, Digium needs to address it, as well as the whole
provisioning via cli thing. I know
/snip
Hello everyone.
I can place outgoing calls no problem with my IP500 (using teliax as our
provider). Thing is, when a call comes in, 90% of the time when I pick
up the handset it drops the call immediately. I turned on SIP debug, and
have listed my extension config from sip.conf. Any help
Has there been any consideration of having asterisk save to a file the
state of which agents are logged in such that after a restart (or crash)
all agents don't have to manually re-login (after eventually realizing
they're no longer logged in and not receiving calls :) ?
Andrew McRory wrote:
Well I am about to reserve a small padded room so I can bounce off the
walls without inflicting tooo much damage... Nothing is making sense at
this point. I tried several releases last night before settling on the
latest CVS (seemed to work the best). Asterisk was running
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