Just have a lok at this config :
[general]
Disallow=all
Allow=g729
Allow=ulaw
[pstn]
Disallow=all
Allow=g729
[zap]
Disallow=all
Allow=ulaw
In extensions.conf, I change the context for each call, Asterisk doesn't
care of codecs in contexts, it uses the order of general...
Could be good to have
Ian White wrote:
Make sure you have a recent copy of the firmware. There was a bug
preventing registrations from succeeding until Nov 08 2005 and newer
firmwares.
Where can I find the firmware?
--
Best regards,
Bartosz Piec
___
--Bandwidth and
Philip Edelbrock wrote:
We've got a Toshiba DK system w/ analog ports that went to a voicemail
server. I swapped in an Asterisk box with a Digium 4-port fxo card.
It /almost/ worked perfectly.
The problem is that Zap channels never hang up. They have to time out.
I set up MeetMe, but
Thank you very much for your attention;
Here is what you asked for:
***
asteriskge03*CLI set verbose 15
Verbosity is at least 15
asteriskge03*CLI capi debug
CAPI Debugging Enabled
asteriskge03*CLI capi info
Contr1: 2 B
Carlos Alperin a écrit :
No,
I never said that. I'm only not joking with another people believes.
Well, I *am*. Believe it or not, it wasn't even disrespectful.
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users
I would like to develop a video file player tool inside Asterisk. When
calling to an extension answer and Play a video file (H264). With the
applications PlayBack is not possible to give a video extension (only sound
file extension). is it posible?
How do u start in this development? With AGI
Mark Phillips a écrit :
It has to be said that Eid is a funny and possibly suspect celebration
though.
As I understand it (from one of my Muslim underlings) 3 Mad Mulahs
have to look for a particular phase of the moon. When they see this
phase they declare the start of Eid. They apparently
In article [EMAIL PROTECTED],
Steve Totaro [EMAIL PROTECTED] wrote:
Sorry if this is slightly off topic but it does pertain to Asterisk
Users as well as the biz list. Also, sorry if it is a double post but
the first one never made it to the list for some reason.
Please test it out and let
There is no 'sending-complete'/'setup' info-element, please use
immediate=yes in capi.conf
Armin
On Wed, 11 Jan 2006 [EMAIL PROTECTED] wrote:
Thank you very much for your attention;
Here is what you asked for:
Hi List,
Is there a way to have Asterisk remember which agents are registered
to it using a MySQL database rather than in memory? It would help with
high availability / clustering scenarios. It also means you could
restart the server without loosing this information...
Cheers,
Jean-Michel.
Title: Transfer to meetme on different server
Hi there
I am using IAX2 based phones and am wondering if the following is possible:
1. User registers with Server 1
2. User dials an extension on Server 1
3. Extension transfers call to an extension on Server 2, which transfers the call to a
|IN) in
new stack
-- Executing GotoIf(CAPI/BRI1/104695467-1, 0 0?2:4) in new stack
-- Goto (macro-record-enable,s,4)
-- Executing AGI(CAPI/BRI1/104695467-1,
recordingcheck|20060111-103127|1136971887.1) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
In article [EMAIL PROTECTED]
ny.censys.net, [EMAIL PROTECTED] says...
Something to think about is this too, when completed scheduling, ask
would you like to notify another extension, so if the first does not
answer in two attempts, ring a cell phone or such.
But I cannot complain, I use the
In article [EMAIL PROTECTED], [EMAIL PROTECTED]
says...
How can you convert mp3 to gsm? mencoder? Do you have an example?
You can use this page.
http://www.asteriskguru.com/tools/audio_conversion.php
--
Tomislav Parcina
[EMAIL PROTECTED]
___
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Figured it out :)
Basically, you have to have a file called syncinfo.xml in the tftp root
directory, with the following contents:
SYNCINFO
IMAGE VERSION=* SYNC=1/
/SYNCINFO
Also, in SIPDefault.cnf or the phone's configuration
hi,
What is the typical delay (latency and latency variance) in Asterisk
when you use rtp/rtcp between 2 endpoint's? Has anyone measured this?
Also, how much better is the TDMoE on this?
Jan
___
--Bandwidth and Colocation provided by Easynews.com
Hi Kokmeng,
Unfortunately that's wasn't it. WaitExten was executed but then I still
get the timeout error -
Timeout, but no rule 't' in context 'incomingpstn'
I am totally stuck...I have been googling and searching the archives and
testing different things for days to no avail. I thought at
Just another bit of info which might help solve this:
Looking at the Asterisk log messages - I notice when I start up
Asterisk, I see the error:
pbx_config.c: Can't use 'next' priority on the first entry!
Could I be right that its something got to do with priorities? I changed
the incomingpstn
El Jueves, 5 de Enero de 2006 01:12, Alexander Lopez escribió:
Asterisk dows not currently support MultiCast.
You may want to look at some applications that where written for Mbone
http://ntrg.cs.tcd.ie/undergrad/4ba2/multicast/bryan/index.html
If you can incorporate them into an Asterisk
James Harper wrote:
Okay then... next question... if I were to come up with a driver for
asterisk (either as hack in chan_capi, an extension to libcapi20, or a
driver for the kernel) to use the rcapi functionality of the cisco (and
other) isdn ta's, would anyone care to try it?
Thanks
James
Well,
We built a site that runs about 30 E1 PRIs. Heavy load, about a million
call attempts per day.
We built it using 10 Asterisk servers. Integration is achieved through
the application design.
Steve
___
--Bandwidth and Colocation provided by
In article [EMAIL PROTECTED],
[EMAIL PROTECTED] says...
In extensions.conf, I change the context for each call, Asterisk doesn't
care of codecs in contexts, it uses the order of general...
Could be good to have Ssterisk making a match between codecs in General and
the context used to make a
Hi,
On 1/11/06, Aisling [EMAIL PROTECTED] wrote:
Hi Kokmeng,
Unfortunately that's wasn't it. WaitExten was executed but then I still
get the timeout error -
Timeout, but no rule 't' in context 'incomingpstn'
You are still in context 'incomingpstn', this indicates that the
Goto has not
When I try to make attendend transfer (*2) this what hapends.
I press *2 other person goes on hold and I hear transfer. I press
extension number and that extension starts to ring but I don't hear
anything. If nobody picks up that phone call in few seconds I get back
to the person I was talking
I Have a lot of 8xx ciscos too, i would try it too.
ISDN-DCP, which looked pretty straightforward at first glance, isn't.
Rather than a simple wrapper around the CAPI messages it seems to
provide a similar but not even closely compatible message structure,
such that my libcapi20 code is
hi,
Thanks - I was hoping someone who had done this would pop-in.
Do you treat each Asterisk server as a separate entity or do you have a
sentralized Asterisk that perform call-control for all etc? How do you
make them behave as one, or is this not needed?
Also, do you switch voice from
On Wed, January 11, 2006 12:46, Tomislav Parcina said:
When I try to make attendend transfer (*2) this what hapends.
I press *2 other person goes on hold and I hear transfer. I press
extension number and that extension starts to ring but I don't hear
anything. If nobody picks up that phone
Tomislav Parcina wrote:
When I try to make attendend transfer (*2) this what hapends.
I press *2 other person goes on hold and I hear transfer. I press
extension number and that extension starts to ring but I don't hear
anything. If nobody picks up that phone call in few seconds I get back
Hi All
Apologises if this has been disussed and I missed it.
My SetUp
I have a sip phone registered to an asterisk box (a1) in one location 1.
This phone dials an extension which is in another location, so a1 passes the
call via IAX to the other asterisk (a2) in location 2 which then dials the
Rich Adamson wrote:
No. The reason is that if the phones are the only thing on this, the
size of the sip packets will never be greater then 214 bytes.
Given your table below, there are other devices on your network and
6% of those are sending packets of in the 512 to 1023 byte range.
I have posted this to the Asterisk Forums, but
got no response yet. Sorry if you are reading
this for the second time.
What fax hardware do I need for a T1? Ideally, I
will switch my T1 to a digital PRI (not CAS I'm
told, which is not as good) coming into the
building. My CLEC said I can do this
add to iax.conf on server1
register = username:[EMAIL PROTECTED]
on server1
lets say extension 1001 on server1 will transfer the call to extension 1002
on server2
exten = 1001,1,Dial(IAX2/[EMAIL PROTECTED]) ; replace server2 with ip/domain of
server2
on server 2 extension 1002 will join a
On Tuesday 10 January 2006 17:28, Geoff Manning wrote:
Just as an update, the users used to be on two 2mb down/512 up ADSL lines
(PPPoE) (4 users on each) and they never reported a problem. Now that they
are on one SDSL (PPPoA) line (2mb) is when they report the issues.
Threre are *plenty* of
On Wed, January 11, 2006 7:52, scott said:
Hi All
Apologises if this has been disussed and I missed it.
My SetUp
I have a sip phone registered to an asterisk box (a1) in one location 1.
This phone dials an extension which is in another location, so a1 passes
the call via IAX to the other
Andrew Kohlsmith wrote:
My suspect is the SDSL modem; what is it? We use ADC Megabit modems
here and they work fairly well. We've had some issue with the old
Flowpoint 5250s.
It is a Speedtouch 610s. Seems like a pretty robust small biz class modem
but it could be the issue. We are just
In article [EMAIL PROTECTED],
[EMAIL PROTECTED] says...
On http://www.voip-info.org/wiki-Asterisk+config+features.conf:
;courtesytone = beep; Sound file to play to the parked caller
; when someone dials a parked call
;xfersound = beep
Hi Fran, you could do it using Adobe/Macromedia Flash Media Server 2,
but I guess that's not the answer you are looking for.
If you manage to do this and release it under GPL I'll kick in $50 for a
bounty.
Regards,
Dean Collins
[EMAIL PROTECTED]
+1-212-203-4357
+61-2-9016-5642 (Sydney
Good to know its not just me then.
Thanks
Scott
-Original message-
From: Francesco Peeters (Asterisk) [EMAIL PROTECTED]
Date: Wed, 11 Jan 2006 07:18:30 -0600
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users]
On Wed, 2006-01-11 at 08:19 -0500, Geoff Manning wrote:
Andrew Kohlsmith wrote:
My suspect is the SDSL modem; what is it? We use ADC Megabit modems
here and they work fairly well. We've had some issue with the old
Flowpoint 5250s.
It is a Speedtouch 610s. Seems like a pretty robust
Hi,
can anybody tell me what the errors mean and why my asterisk server falls from
time to time. From time to time means several hours, not regularly.
I also can provide a core if someone can debug?
Thanks and regards
Jan 11 14:34:59 NOTICE[13573] chan_zap.c: Hangup, did not find cref 83,
First,
Something seems to be wrong with the list. I'm not the only person
who has expressed seeing their messages either arrive late, or not at
all.
With that out of the way..
Is anyone aware of any type of failover device for PRI on asterisk?
I've found the ISDNGuard, however it is currently
Pete Barnwell wrote:
Are you sure about that? Most ADSL in the UK is on PPPoA (BT supplied
- it may be different for LLU providers), not PPPoE so I wouldn't
think this has actually changed.
Correction, you are right. The old ADSL we were running was indeed PPPoA.
That has not changed.
Hi there,
I am trying desperatly to register my WiFi Phone UTStarcomm F1000 with
my asterisk server. I already changed the name of the user to
anonymous since it looks like the phone sends that name. The WiFi
Phone's IP is 192.168.1.217, the asterisk server's IP is 192.168.1.200
What is it
This is a great idea!
You could have an IVR presented by a computer generated figure. You
could play viewzak to folks on hold. Or how about the company promo
reel when waiting for you turn in the call center queue?
I'm loving this idea!!
In a previous life I used to be a video editor for
Hello,
I am have trouble figuring out how to connect my [EMAIL PROTECTED] system (2.2)
to a legacy PBX extension. I have FXO ports available to use, and I am able
to dial in to Asterisk from any extension via port 1, and I want to use port
2 for dial from an Asterisk extension (SIP, IAX, etc)
I've tried the Sipura and it doesn't work. It says it's sending a
notify but the SPA-2002 doesn't reboot.
On 1/5/06, Jian Hong GUAN [EMAIL PROTECTED] wrote:
Hi,
Can you give me some councils of remotely rebooting sip phones in asterisk
server? How to configure sip_notify.conf and sip.conf?
Hi
Someone knows a free web based SIP client for use with any provider ?
Thanks
roberto-- Ing. Roberto PereyraContenidosOnlineServidores BSD, Solaris y LinuxSoporte técnico ISPsJabber ID: [EMAIL PROTECTED]
For reliable and professional DNS, use DNS Made Easy!http://www.dnsmadeeasy.com/u/14989
Are there a SIP standard to transmit flash? For instance I would like to
send a SIP message indicating to a FXO gateway to apply a flash for
transfer.
In RFC 2833 page 11, in DTMF Events, the table show that DTMF 16
(decimal) is used for flash. Can I use this?
Jorge Mendoza
As far as I know, you define the interface to TDMoE when you choose the
zaptel driver to work with. One of the options is Zaptel over Ethernet.
After that everything belongs to a PtP Ethernet connection between the box
with the TDMoE the Interface to T1, FXO or what ever you has and your
asterisk
Do you need failover on wich side? PRI or Asterisk? Both?
Straight to the last option:
PRI: the best if you have more than one PRI is to do hunt on the provider
side, so when one is full or down, all calls are going to be directed to the
second one.
Asterisk: Do redundancy, so you need to have
I would like to develop a video file player tool inside Asterisk. When
calling to an extension answer and Play a video file (H264). With the
applications PlayBack is not possible to give a video extension (only
sound
file extension). is it posible?
How do u start in this development? With AGI
In article [EMAIL PROTECTED],
[EMAIL PROTECTED] says...
I assume I need a TDM400P (TDM20B flavor for 2
analog stations), but I am not sure.
You can buy ATA (analog terminal adapter) or the card you mention. Bouth
of them shuld work just fine.
--
Tomislav Parcina
[EMAIL PROTECTED]
Over the last couple of weeks I have seen a thread about remotely rebooting SIP
phones from Asterisk.
Is there something inherent in Asterisk that *requires* that SIP phones to be
rebooted in a particular scenario, or is it just so that phones can pickup new
firmware and/or configuration from
On Wed, 11 Jan 2006 15:38:04 +0100
Matt Riddell (IT) [EMAIL PROTECTED] wrote:
I would like to develop a video file player tool inside
Asterisk. When
calling to an extension answer and Play a video file
(H264). With the
applications PlayBack is not possible to give a video
extension (only
We have to reboot our phones sometimes when we do something server side,
mainly because the cisco firmware doesn't seem to handle everything very
well. Usually it's just to pull new configs though, as we test more
features and roll them out.
Aaron
Steve Langstaff wrote:
Over the last
vmail*CLI realtime mysql status
Jan 11 09:53:04 ERROR[3597]: res_config_mysql.c:623 mysql_reconnect:
MySQL RealTime: Failed to reconnect. Check debug for more info.
vmail*CLI realtime mysql status
Connected to [EMAIL PROTECTED], port 3306 with username voicemail for 1 days, 5 hours, 32 minutes,
As a rule of thumb, I always explicitly set CallerID in my dialplan before
making a call through IAX, SIP or PSTN. If you make it part of a generic
dialout routine then it isn't a hassle. It always works.
-Original Message-
From: scott [mailto:[EMAIL PROTECTED]
Sent: Wednesday, January
On Tue, Jan 10, 2006 at 06:52:43PM +0100, Louis-David Mitterrand wrote:
On Tue, Jan 10, 2006 at 05:43:12PM +0100, Armin Schindler wrote:
On Tue, 10 Jan 2006, Louis-David Mitterrand wrote:
[C:4] 22:0188:202 - D-X(003) 02 01 7F
[C:4] 22:0189:202 - D-X(003) 02 01 7F
[C:4]
Also, the old grandstreams would lose their registrations periodically.
I have not played with a grandtream in quite a while so I would assume
they fixed this in firmware but that was another reason for regular
reboots.
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Thank u Matt!!
I will try it!!!
and what about the extensions supported? file.gsm and file.h264 is
possible?
how do u create both files? would it be possible to create both files from
an AVI or a MPEG? may i use MPEG4IP??
Thank u in advance!!!
Fran
-Mensaje original-
De: [EMAIL
Polycom phones need a reboot after making configuration changes.
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Wednesday, January 11, 2006 7:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Why remotely reboot SIP
Sig Lange wrote:
vmail*CLI realtime mysql status
Jan 11 09:53:04 ERROR[3597]: res_config_mysql.c:623 mysql_reconnect: MySQL
RealTime: Failed to reconnect. Check debug for more info.
vmail*CLI realtime mysql status
Connected to [EMAIL PROTECTED], port 3306 with username voicemail
for 1 days, 5
Robert Webb wrote:
As a noob that might be interested in this also, how well does this work
with the seperate audio and video files and keeping them in sync? I just
keep flashing back to the old days of trying to do stereo with music
using two C64's.. :-)
Heh, my nick is ZX81! :)
The thing
Hello everybody,
Do you know if it's possible to push the status of an extension (a
phone) to a phone like blinking a light on the phone ? And do you know
wich brand of phone can do this ?
I'd like to make the same as the secretary phones that can see the
status of lines before putting a call
Roberto Pereyra wrote:
Hi
Someone knows a free web based SIP client for use with any provider ?
Thanks
roberto
--
Ing. Roberto Pereyra
ContenidosOnline
Servidores BSD, Solaris y Linux
Soporte técnico ISPs
Jabber ID: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
Hi Roberto, im looking for a
Call parking...
I can park a call that was received on a particular phone.
But I can not park a call from the phone that initiated a call. The DTMF
are just sent out to audio channel.
Any hints anyone?
Thanks,
Andre
___
--Bandwidth and Colocation
We use Snom phones for the BLF function as you are suggesting and it
works great. The Grandstream GXP-2000 with the beta firmware supports
this as well but I hear its a bit buggy. The snom phones are nice
because depending on the size of the office you can add an additional
side cart with many
[EMAIL PROTECTED] wrote:
The second edition of my book VoIP Telephony with Asterisk is now in
print and available. You can find out more about it at our web site
http://www.signate.com/products.php
You've posted this every week for the past three or four weeks now;
please stop.
Eric Lyons wrote:
I got zttool running and selected loop on the interface, but it didn't
seem to do what they wanted (nor could I tell that it did anything at
all). Many googles for zaptel and loop didn't turn up anything useful.
This is a bug that needs to be fixed; currently the
Tomislav Parcina wrote:
In article [EMAIL PROTECTED],
[EMAIL PROTECTED] says...
I assume I need a TDM400P (TDM20B flavor for 2
analog stations), but I am not sure.
You can buy ATA (analog terminal adapter) or the card you mention. Bouth
of them shuld work just fine.
Wonderful
Do you mean changes to the phone's configuration, or changes to Asterisk's
configuration?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Douglas
Garstang
Sent: 11 January 2006 15:07
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE:
Miguel wrote:
Roberto Pereyra wrote:
Hi
Someone knows a free web based SIP client for use with any provider ?
Thanks
roberto
--
Ing. Roberto Pereyra
ContenidosOnline
Servidores BSD, Solaris y Linux
Soporte técnico ISPs
Jabber ID: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
Is there any documentation around for running Asterisk in a Cluster (I
assume you mean a n+1 cluster as you list a failover cluster as a
different option). I was under the impression that it can't be done..
Thanks.
On 1/11/06, Carlos Alperin [EMAIL PROTECTED] wrote:
Do you need failover on wich
Hi,
I tried to patch asterisk 1.2.1 on a Debian Sarge distro with the patch
made by tzafrir but I still cannot set writing permission to directories.
I tried to put umask 007 inside .bash_profile but it doesn't work.
Is there anyone who can help me?
TIA
Giorgio Incantalupo
Am trying to monitor and record an in-process phone call using a
remote computer and the Asterisk Manager API. Have something that is
working, but the call recording volume is to low to be usable.
dialplan
exten = 8159,1,ZapBarge(Zap/1)
remote application with Asterisk Manager API
Hi,
we are running in difficulties with some (rare) numbers: Asterisk doesn't
detect answer and rings indefinitely or drops call with NOANSWER.
It seems that these numbers are automatic responders.
I tried to debug with 'pri intense debug span 1' but no useful info.
I'm using a Sangoma A102 card
Jean-Michel Hiver wrote:
Is there a way to have Asterisk remember which agents are registered
to it using a MySQL database rather than in memory? It would help with
high availability / clustering scenarios. It also means you could
restart the server without loosing this information...
Check
Fran wrote:
Thank u Matt!!
I will try it!!!
and what about the extensions supported? file.gsm and file.h264 is
possible?
how do u create both files? would it be possible to create both files from
an AVI or a MPEG? may i use MPEG4IP??
I use VCDCutter to create a fake webcam which can be fed
Well, another try
[general]
Disallow=all
Allow=ulaw
Allow=g729
For the Uas, they are sets to have g729 first
Calls to/from pstn needs g729
Calls to/from zap needs Ulaw
ALL incoming calls works OK even if the caller is G729(I have made a caller
using g729 only)...
Calling zap = no problem, Ulaw
Jorge Mendoza wrote:
Are there a SIP standard to transmit flash? For instance I would like to
send a SIP message indicating to a FXO gateway to apply a flash for
transfer.
In RFC 2833 page 11, in DTMF Events, the table show that DTMF 16
(decimal) is used for flash. Can I use this?
Yes. This
Hi all, I've 3 providers (A, B, and C) the A is giving me freecalls to
USA, the B is giving my freecalls to Europe, and C is to call the otre
destinations. My question is, how can I configure the outboud routing to
select the right trunk for every destination?
All the providers uses the dialing
extensions.conf
[context]
exten = s,n,Set(DYNAMIC_FEATURES=zapflash)
exten = s,n,Dial(SIP/,15,tw)
features.conf
[applicationmap]
zapflash = *3,caller,flash,() needed a comma between flash an the
()
I Wonder (aloud) if there'd be a way to send the incoming call to another phone?
On Wed, Jan 11, 2006 at 04:59:00PM +0100, gincantalupo wrote:
Hi,
I tried to patch asterisk 1.2.1 on a Debian Sarge distro with the patch
made by tzafrir
from a deb or self built? If from a deb:
dpkg -l asterisk
but I still cannot set writing permission to directories.
I tried to put
Hi,
I have an issue, I can hear music on hold with
MusicOnHold() but I cant hear anything with Dial(,m).
(I did: make mpg123, cd mpg.., make, make
install). Mi extensions.conf is:
[incoming]
exten
=s,1,Answer()
exten =s,n,Background(welcome)
exten =s,n,WaitExten(20,m)
;at this
Give me your providers and I give you the agi script to do that :)
Olivier
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Guillermo
Salas M
Envoyé : mercredi 11 janvier 2006 17:17
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet :
You could use a 2 span T1 card from Digium and plug one span into a
channel bank, and have FXS ports on the CB for the fax machines. With
the latest firmwares from Digium these streams are bridged internaly
on the card, and don't even come on to the PCI bus.
On 1/11/06, John Crew [EMAIL
I am having a bit of a problem with several phones (Polycom 601 and Aastra 9133i). I have a new installation in a brand new office. The office is bare and there is a lot of echo. This causes all the phones on the office to have a very audible echo. I know it is not really a hardware problem,
Thanks so much for the suggestions. I'm having trouble getting the list
emails so I just looked at the archives for yesterday. Funny, yesterday I
did run into a broken pipe error while restarting using asterisk -v. It was
wilcalu.so (or something like that). I've stopped and started asterisk
Looking for bulk DID's for the following location's in China (+86):
Shanghai (021)
Guangzhou (020)
Shenzen (755)
Also looking for bulk DID's in Hong Kong (+852).
Thanks
Steven Ducat.
___
--Bandwidth and Colocation provided by Easynews.com --
I am happy to announce the release of ruby-agi-1.0.2
This is a stable release of ruby-agi.
ruby-agi is available at
http://rubyforge.org/projects/ruby-agi/
You can also install ruby-agi via gem.
To install ruby-agi gem package, try
% gem install ruby-agi
Feel free to send me your
On Wed, January 11, 2006 16:00, Colin Anderson said:
As a rule of thumb, I always explicitly set CallerID in my dialplan before
making a call through IAX, SIP or PSTN. If you make it part of a generic
dialout routine then it isn't a hassle. It always works.
It sometimes doesn't for my
As far as I know, the Polycom's don't have any kind of echo cancellation for this type of thing, however there is a technology called a FICUS PLANT, which inhabits many offices and can solve your bare office problem :)
--TomOn 1/11/06, Carlos Chavez [EMAIL PROTECTED] wrote:
I am having
Carlos Chavez wrote:
I am having a bit of a problem with several phones (Polycom 601
and Aastra 9133i). I have a new installation in a brand new office.
The office is bare and there is a lot of echo. This causes all the
phones on the office to have a very audible echo. I know it is not
Is it
possible to have nested MySQL queries in extensions.conf?
Ie,
perform a query, grab a value, and then jump to another location in the dialplan
and do another query based on that original value. I'm having problems with the
result and fetchid's and I'm not sure if it's even
I am having an issue using a Polycom 501 and VoIP for outgoing calls where if I call say my credit card company and try to follow their PBX menu, the key presses never register with their PBX. It's as if every key press I make absolutely nothing is being sent to them. Is there some setting in the
On 11/01/06, Douglas Garstang [EMAIL PROTECTED] wrote:
Is it possible to have nested MySQL queries in extensions.conf?
Ie, perform a query, grab a value, and then jump to another location in the
dialplan and do another query based on that original value. I'm having
problems with the result
Usually just phone changes, but if you reboot the server, or reload
something, sometimes the phones need to re-register and it's just easier
to send a remote reboot.
Aaron
Steve Langstaff wrote:
Do you mean changes to the phone's configuration, or changes to Asterisk's
configuration?
I'm running Asterisk on a Gentoo box with the Zaptel 1.2.1 drivers.
If I boot the machine without having the wcfxs module autoload, then
install the module with modprobe, asterisk works just fine.
If I boot the machine and autoload the wcfxs module, the module loads fine:
Jan 11 11:06:55
Peter,
Too slow! We're going to potentially be doing several MySQL lookups for routing
even the most basic of calls, and if every one of those queries has to make a
call out to an AGI script, it would become a performance problem.
Douglas.
-Original Message-
From: Peter Bowyer
Scratch the last mail this is the right
one
Hi,
I have an issue, I can
hear music on hold with MusicOnHold() but I
cant hear anything with WaitExten (,m).
(I did: make mpg123, cd
mpg.., make, make install). My extensions.conf is:
[incoming]
exten =s,1,Answer()
exten
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