Joseph wrote:
On Mon, 2006-10-09 at 01:45 -0300, Hermann Wecke wrote:
On Sun, Oct 08, 2006 at 10:39:26PM -0600, Joseph wrote:
I have bind-address = 127.0.0.1 in my.cnf
the cdr was working find with asterisk 1.0.1 just after upgrade
something is not connecting.
I don't know if
Greetings !
Its kind of OT, but if there are any Europeans going to Astricon in Dallas,
please send a message of-list.
It's possible we will be on the same flight,(i am flying from Frankfurt) ;)
so it will be a good way to know it's other and spend some of the 10 hours +
flight time .
Regards
I will be also on a flight from frankfurt (lufthansa), but a few days early.
Zoa.
Stelios Koroneos wrote:
Greetings !
Its kind of OT, but if there are any Europeans going to Astricon in Dallas,
please send a message of-list.
It's possible we will be on the same flight,(i am flying from
On 2006-10-09 22:05:06 -0700, Joseph [EMAIL PROTECTED] said:
On Mon, 2006-10-09 at 20:41 -0400, Forrest Beck wrote:
Anyone using the echo cancelation cards from digium? We are using the
single span T1 card with out echo cancel and I was curious if it was
worth the money.
I'm running
On 2006-10-09 15:53:36 -0700, Brandon Galbraith
[EMAIL PROTECTED] said:
Does anyone know of any ATA devices (Linksys, Dlink, Cisco, etc) that will
fail over to POTS for an emergency call? I'd like to route any call except a
911 call over SIP or IAX, but any 911 call should be routed out over
On 2006-10-09 05:31:30 -0700, Benny Amorsen [EMAIL PROTECTED] said:
PB == Peter Bowyer [EMAIL PROTECTED] writes:
PB Fair enough - that's a bit different to 'Asterisk 1.2 is not ready
PB for PRIME TIME' though, isn't it? There are plenty of stable 1.2
PB releases, all of which have many fewer
On 2006-10-08 21:28:08 -0700, Nic Bellamy [EMAIL PROTECTED] said:
Martin Joseph wrote:
I am seeing occasional stuck SIP channels that seem to occur when the
fricking Nokia E60 drifts out of WIFI range in the midst of a call.
This is particularly annoying when the stuck channels include my
Probably best change the login and password from the defaults now
you've posted this - your admin interface is wide open
On 09/10/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Alex...I do not have FreePBX. What I have is this:
http://70.89.124.237/
Ed
Hi All,
Would asterisk and zaptel compile on 64bit dual xeon hardware??
Rgds
From: Martin Joseph [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: Asterisk
On 9 Oct 2006, at 11:49, raviprakash sunkara wrote:
Hello Users.
I'm in Dilemma with the performance on SIP and IAX
Can any one help ...
1) Difference between the SIP and IAX...
which one is Best... in VOIP service
I'm using only SIP protocol for my
On 9 Oct 2006, at 21:19, Douglas Garstang wrote:
I'm just going to jump in here, and ask a stoopid question.
How could you possibly write a multi-user front end in AJAX without
using a database backend like MySQL?
Wandering off topic here, but I'll bite
There are a few options for
On 9 Oct 2006, at 17:36, Benko wrote:
Hello!
I'm having a problem which actually looks banal. I'm trying to
connect 3 servers via iax with each other. However, i've not been
successfull so far. Asterisk always tries to authenticate the calling
user with the credentials of the last entry in
Yes
Akpome Akpoguma wrote:
Hi All,
Would asterisk and zaptel compile on 64bit dual xeon hardware??
Rgds
From: Martin Joseph [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: asterisk-users@lists.digium.com
Subject:
Morning all,
We're looking for hand free solution to use with Asterisk beside BT
headsets. I was thinking on Sipura 841 but it seems that the headset
jack connector is not carrying voice (microphone), only audio.
Ideal would be a headset audio+microphone with RJ11 4p female that we
could
Hi All,
Would asterisk and zaptel compile on 64bit dual xeon hardware??
Rgds
From: Martin Joseph [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: Asterisk
On Mon, Oct 09, 2006 at 01:50:11PM -0400, J. Oquendo wrote:
Edit /etc/asterisk/logger.conf then asterisk -rx logger reload or in
cron the dirty way:
0 * * * * ls -ltha /var/log/asterisk/|awk '{print $5,$9}'|grep 0
|grep -v [1-9]|xargs rm -rf
Huh?
Is it supposed to pick files
When you want to ask a new message, rather than replying to an existing
one, please write a new message and don't reply to an existing one.
(and don't even reply to an existing one and delete its contets. This is
not the same).
Posting the same question twice is also not a good habit.
See reply
Joseph wrote:
On Mon, 2006-10-09 at 20:41 -0400, Forrest Beck wrote:
Anyone using the echo cancelation cards from digium? We are using the
single span T1 card with out echo cancel and I was curious if it was
worth the money.
I'm running Asterisk 1.0.11 with few Sipura 3000/2000 units and
Hi,
Is it within the realms of possibility to have a single conference with
multiple numbers?
I'm thinking of getting PSTN numbers in a number of different countries so
that people in those countries only pay for a local call.
At this stage doing it with VoIP is out of the question.
Thanks
Yes you can. If you are dealing striclty with non-voip you may have a bit of
a challenge getting the calls to the same server but if you had a way of
getting all the calls to the same box, or have boxes all over and have them
all conntect to one main box then it would be possible.
-
I have never used T1 cards but as far as POTS line cards I would say that I
like sangoma better. It is a little bit harder to set up but works wonders.
- Original Message -
From: Thomas Kenyon [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hello UsersHelp me ... the below error [codec_lpc10.so] = (LPC10 2.4kbps (signed linear) Voice Coder) == Parsing '/etc/asterisk/codecs.conf': Found --
Message count requested for mailbox [EMAIL PROTECTED] but voicemail not loaded. -- codec_lpc10: using generic PLC == Registered translator
On Tue, Oct 10, 2006 at 11:37:49AM +0100, Mike Williams wrote:
Hi,
Is it within the realms of possibility to have a single conference with
multiple numbers?
exten = 1234,1,Meetme()
exten = 5676,1,Meetme()
?
--
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755
Hi list,
out of pure frustration I have prepared another tutorial (must be the
season) about how to filter the various outputs of Asterisk in order to
keep track of what is going on in realtime in a call-center, to avoid
being swamped by too many logging and information on the * side.
I don't know if there is a better way to do this with meetme itself, but
you could use the manager interface (or even the file method described
in http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out).
You can pass a Data argument with the filename, to an extension that
simply plays
I know that this is a silly suggestion but you should check to make sure
that you actually have the cdr_mysql module, because at some point (I
believe at the 1.2 release or shortly thereafter), it was moved into
asterisk-addons.
--Brian
On Tue, Oct 10, 2006 at 08:31:43AM +0200, Garth van Sittert
Absolutely, the MeetMe command just takes a conference number. You
could have as many extensions invoke it as you would like.
--Brian
On Tue, Oct 10, 2006 at 11:37:49AM +0100, Mike Williams wrote:
Hi,
Is it within the realms of possibility to have a single conference with
multiple
Hi,
I have setup two asterisks with ucarp, to build a HA cluster. Everything works
fine, if one of the machines is going to die completely. But if the asterisk
software is running, but behaving not correctly, this cannot be detected by
the ucarp software.
I think I need a script that
Hi Sebastian, This url http://underlinux.com.br/content/view/6330/70/ have some thinks that you need.Leonardo Silva
2006/10/10, Sebastian Reitenbach [EMAIL PROTECTED]:
Hi,I have setup two asterisks with ucarp, to build a HA cluster. Everything worksfine, if one of the machines is going to die
On Tuesday 10 October 2006 12:24, Brian Rogan wrote:
Absolutely, the MeetMe command just takes a conference number. You
could have as many extensions invoke it as you would like.
Thanks all.
--
Mike Williams
___
--Bandwidth and Colocation provided
Does anyone have a quick howto and a sample to get
whisper paging to work? I'm running sterisk Asterisk 1.4.0-beta2
Thanks for your help!
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or
Assuming the outgoing T1 to the Norstar is a standard T1 that accepts ANI and
DNIS all have to do is
exten = _XXX,1,Dial(Zap/g2/*${CALLERIDNUM}*${EXTEN}*,,r)
This will redial the caller id (ANI) and the 3 digits Dialed (DNIS) to the
Norstar T1 in the formst *ANI*DNIS*
I did the same thing
Ideal would be a headset audio+microphone with RJ11 4p female that we
could plug into the handset cable of any IP phone, or a converter
2xjack2,5mm female RJ11 4p female -which seems not to exist-.
What are you recommanding/using/installing in such case?
I don't know if it would work on any
DG == Douglas Garstang [EMAIL PROTECTED] writes:
DG How can I check a number is within a specified range in the
DG dialplan? What's the greater than operator? How would I use a
DG combination of greater than and less than in conjection with
DG GotoIf()? The following seems to break the dialplan.
The Sangoma single port T1 card also works well, and , along with the 5
year warranty, works with just about any MB that it will fit, makes it a
no brainer choice over the Digium products.
Sangoma just doesn't say try another motherboard. If their product
doesn't work, they find out why and fix
Hi Leonardo,
I had the problem, asterisk was running, the port was open, but I
misconfigured Asterisk that way, that it was impossible, to register on the
asterisk. As it seems to me, hapm can only check whether a port is closed or
open. unfortunately I do not understand that brazilian
I´m about to acquire an E1 interface. I was reading about TE110P and
hardware incompatibilities issues with some boards, servers and
chipsets. I also read a lot of compliments about Sangoma Hardware
(specially for E1/T1 interfaces) and I was wondering if A101 from Sagoma
is a better choice
Try using testcall tool included with Unicall to debug, as shown in
this document I wrote a couple of months ago. It also shows how to use
zttool to detect problems in the E1 layer.
http://moy.ivsol.net/unicall/mfcr2-asterisk-unicall-0.2-english.pdf
The verbosity level in testcall.c must be at
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I'm quite happy with sangoma cards, no issues so far, plus their
installation/setup software makes it a breeze to get everything working.
R.R. Libera wrote:
I´m about to acquire an E1 interface. I was reading about TE110P and
hardware
On Mon, Oct 09, 2006 at 06:42:06PM -0400, Doug Lytle wrote:
Doug Lytle wrote:
Jay R. Ashworth wrote:
From the A102 spec sheet:
* DSU/CSU set up entirely in software.
I guess I need to learn to read a little more carefully. Looks like
it's 'set up' in software.
Well, I was working
[EMAIL PROTECTED] wrote:
Hello,
I'm having trouble getting this to work:
I have a ring group that dials an extension and if no answer dials a cell
phone. If the cell phone doesn't answer I want to go to voicemail or another
extension. I have set the timeout to 15 seconds but it never
AFAIK it's not possible.
On 10/10/06, Hall, Eric M. [EMAIL PROTECTED] wrote:
Does anyone have a quick howto and a sample to get whisper paging to work?
I'm running sterisk Asterisk 1.4.0-beta2
Thanks for your help!
___
--Bandwidth and Colocation
I am using the TE110P with the Intel 945P chipset, and I don't have
any issues with compatibility. The 945P chipset is a very common
chipset for the D and 4 processor.
Works quite well.
On 10/10/06, R.R. Libera [EMAIL PROTECTED] wrote:
I´m about to acquire an E1 interface. I was reading about
FYI.
TITLE:
Polycom SoundPoint IP 301 Denial of Service
SECUNIA ADVISORY ID:
SA22266
VERIFY ADVISORY:
http://secunia.com/advisories/22266/
CRITICAL:
Less critical
IMPACT:
DoS
WHERE:
From local network
OPERATING SYSTEM:
Polycom SoundPoint IP 301
http://secunia.com/product/12229/
Plantronics makes something like this...designed to go inline with handset
cable, with 2 2.5mm audio connectors for connection to PC.
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Time Bandit
Sent: Tuesday, October 10, 2006 9:19 AM
To: Asterisk
Mike Williams wrote:
Hi,
Is it within the realms of possibility to have a single conference with
multiple numbers?
I'm thinking of getting PSTN numbers in a number of different countries so
that people in those countries only pay for a local call.
At this stage doing it with VoIP is out of
R.R. Libera wrote:
I´m about to acquire an E1 interface. I was reading about TE110P and
hardware incompatibilities issues with some boards, servers and
chipsets. I also read a lot of compliments about Sangoma Hardware
(specially for E1/T1 interfaces) and I was wondering if A101 from
Sagoma is
I'm curious what asterisk solutions there are out there for inbound call
centers with multiple DIDs. I'm looking for solutions for a setup where
single system may have 1000 DIDs going to it, one for each account. Each
account may not get that many calls.
Solutions that will all reporting on
Hello. It would appear that the voicemail module is
not loaded. If this is a new install, did you install the sample config
files? Specifically voicemail.conf.
-Ejay
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of raviprakash
sunkaraSent: Tuesday, October 10, 2006 6:07
I thought whisper paging was implemented in 1.4?
-Original Message-
From: C F [mailto:[EMAIL PROTECTED]
Sent: Tuesday, October 10, 2006 7:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] whisper paging
AFAIK it's not possible.
On
Hi!
Does anyone have some experience with a Xorcom TS-1 and a 1 or 2 port Digium PRI card? I am looking for a SIP/IAX to ISDN gateway and this combination could by interesting.
But Xorcom writes that the TS-1 is compatible with Digium PRI cards but that it has not been tested much.-- Morten
Then correct me.
On 10/10/06, Douglas Garstang [EMAIL PROTECTED] wrote:
I thought whisper paging was implemented in 1.4?
-Original Message-
From: C F [mailto:[EMAIL PROTECTED]
Sent: Tuesday, October 10, 2006 7:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hello,
when trying to use a snom 360 (Firmware 6.2.3) with Asterisk 1.2.12, I receive
no audio. Asterisk 1.4.0 b2 works fine though. I´d upgrade to 1.4 if I hadn´t
just bought a Junghanns OctoBRI that apparently only works with bristuff,
which is stuck at the 1.2 series.
Is this a known
Seems that you guys are right, sorry.
http://www.digium.com/en/mediacenter/news/viewpress.php?id=Asterisk1.4
On 10/10/06, C F [EMAIL PROTECTED] wrote:
Then correct me.
On 10/10/06, Douglas Garstang [EMAIL PROTECTED] wrote:
I thought whisper paging was implemented in 1.4?
-Original
Crikey. I can't get this to work!
Allegedly, you can press 0 while in the voicemail greeting and be dropped to
the 'o' extension. For some reason, I can't get it to work. The 'docs' aren't
clear about what context the o extension should be in. The voip wiki says
the context for the voicemail
According to this:
http://bugs.digium.com/view.php?id=8019
it seems that it's part of Chan_spy, what does show application
chanspy on the cli give you?
On 10/10/06, C F [EMAIL PROTECTED] wrote:
Seems that you guys are right, sorry.
That's what we got told at the Asterisk bootcamp training in Kansas City a few
weeks ago... :)
-Original Message-
From: C F [mailto:[EMAIL PROTECTED]
Sent: Tuesday, October 10, 2006 9:26 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
C F wrote:
According to this:
http://bugs.digium.com/view.php?id=8019
it seems that it's part of Chan_spy, what does show application
chanspy on the cli give you?
I would wait until the next beta before giving whispering a try, it
underwent some major changes. Alternatively you can grab the
Tim Panton wrote:
On 9 Oct 2006, at 17:36, Benko wrote:
Hello!
I'm having a problem which actually looks banal. I'm trying to
connect 3 servers via iax with each other. However, i've not been
successfull so far. Asterisk always tries to authenticate the calling
user with the credentials of
Douglas Garstang wrote:
Crikey. I can't get this to work!
[foo]
exten = 556,1,Answer
exten = 556,n,Voicemail([EMAIL PROTECTED])
I believe it needs to be in the same context as your voicemail. Mine is:
[voice-mail]
exten = s,1,Set(CALLBACK=${DB(vmcallback/${CALLERIDNUM})})
exten =
If you get an answer for this please post it here on forum as I and at
least one other I've talked to have this same problem. I found it was
only a problem from external calls though not internally. Same for you?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Douglas Garstang wrote:
Crikey. I can't get this to work!
Allegedly, you can press 0 while in the voicemail greeting and be dropped to
the 'o' extension. For some reason, I can't get it to work. The 'docs' aren't
clear about what context the o extension should be in. The voip wiki says
the
Dang it. Thanks. Blindly trusting the voip-wiki is bad
When using the zero '0' and star '*' it's important to note that the context
you placed the application voicemail in is irrelvant...
Doug.
-Original Message-
From: Doug Lytle [mailto:[EMAIL PROTECTED]
Sent: Tuesday, October
Hi Kape,
With latest asterisk 1.2.12.1, zaptel 1.2.9.1 and bristuff 0.3.1s after
a while calls become stuck: either the caller or callee can't hear the
other party, or heavy static is heard. An asterisk restart fixes it for
a short while only.
This doesn't happen with our older installs
Jan, im sorry to get back to you so late, ive been busy. It seems i
sent you an incorrect patch I was testing, but I have found the
correct patch in mantis:
http://bugs.digium.com/view.php?id=6682
Please be aware that the patch I sent you initially used a funciton
that received 1 or more DTMF
That was it for me as well. Couldn't get that answer the last time I
asked. Thanks guys.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Tuesday, October 10, 2006 11:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Brian Candler wrote:
I'm looking for a way to set up a VOIP network in branch offices where one
or more phones have lifeline capability, i.e. can place calls if the IP
network or VOIP service dies, or even if power goes down. (I'm thinking of
business continuity here, not just emergency
Hi Michael,
If you want something very basic:
http://www.micpc.com/eventmonitor
will pop up a menu for an incoming call to an agent. It is a very basic system
but i wrote it as such to be both functional and a framework to build from.
You would need to enhance it (for your specific
Douglas Garstang wrote:
Dang it. Thanks. Blindly trusting the voip-wiki is bad
When using the zero '0' and star '*' it's important to note that the context you
placed the application voicemail in is irrelvant...
I've made note on the wiki. He did his testing via a macro under 1.07
We have a solution like this working just fine for almost a year. We are
using qurad card for that. It is a good idea to have both PRI on one card.
CLID shouldl remain the same.
Vlad
- Original Message -
From: Forrest Beck [EMAIL PROTECTED]
To: Asterisk Users List
Douglas Garstang wrote:
Crikey. I can't get this to work!
Allegedly, you can press 0 while in the voicemail greeting and be dropped to
the 'o' extension. For some reason, I can't get it to work. The 'docs' aren't
clear about what context the o extension should be in. The voip wiki says
the
I want to try something with my asterisk but I have something that I need to
know. The thing is that I am behind a NAT (I have to phones in a lan
connected to the internet with a router), my server is directly conected to
the internet on a different connection (in another place). I make a call
Does anybody know if this is supposed to work and if so, what, if
any, workaround is needed? I have other phones (Snom, Polycom) MWI
working with this system fine. 6.0.0.19 (latest) Mitel SIP firmware
is loaded.
Thanks for your time,
- Jesse
--
Jesse Peterson [EMAIL PROTECTED]
Hi,How do I cause the dial plan to dial a different extension if the first either never picks up or presses ignore or what have you?For example, something like this:exten = context,1,Dial(
SIP/[EMAIL PROTECTED])exten = context,2,Dial(SIP/[EMAIL PROTECTED])Currently, if the first number doesn't
Hi!
I'm trying to communicate a Cisco CCM 4.0 with Asterisk 1.2.11, I 've followed the info in
http://www.voip-info.org/tiki-index.php?page=Asterisk+Cisco+CallManager+Integration
but still not able to make Asterisk communicate with Cisco. I keep on receiving ---
SIP/2.0 400 Bad
If you wanted to everything manually it could be done. I would use asterisk
real time. Never worked with any specific programs that are designed for
this so I can't reccoemnd one.
- Original Message -
From: Michael Sampson [EMAIL PROTECTED]
To: Asterisk Users Mailing List -
I second that. I had card from Sangoma with echo can. When ever the echo
can. was enabled ZAP/2 would work with one way audio. Sangoma had a tech ssh
into my box for a few hours ar no charge. It was the first time they saw
such a problem with the card the supplier sent me a new one the next
Hi Michael,
do you want to do the reporting or to configure the dialplan? QueueMetrics
will do the reporting for no matter how many ACD queues, and will
automatically sync to the underlying * config files, so there should be no
problem with reporting. You can also configure it in
-Original Message-
From: Eric ManxPower Wieling [mailto:[EMAIL PROTECTED]
Sent: Tuesday, October 10, 2006 11:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Voicemail Press '0'
Douglas Garstang wrote:
Crikey. I can't get this to
Mark Price wrote:
Hi,
How do I cause the dial plan to dial a different extension if the first
either never picks up or presses ignore or what have you?
For example, something like this:
exten = context,1,Dial(SIP/[EMAIL PROTECTED])
exten = context,2,Dial(SIP/[EMAIL PROTECTED])
Currently, if
Looks like the CallManager is unable to find the endpoint in its database. Make
sure asterisk trunk on the Call manager is in the same calling Search Space
as the phones are in, or make sure there is access between the calling search
spaces
-Eric
-- Original message
Douglas Garstang wrote:
-Original Message-
From: Eric ManxPower Wieling [mailto:[EMAIL PROTECTED]
Sent: Tuesday, October 10, 2006 11:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Voicemail Press '0'
Douglas Garstang wrote:
Crikey. I
exten = context,1,Dial( SIP/[EMAIL PROTECTED])
exten = context,2,Dial(SIP/[EMAIL PROTECTED])
Currently, if the first number doesn't answer, the session is closed.
Specify a time out. Without it * will not continue to priority 2 if
[EMAIL PROTECTED] is reachable but does not answer.
exten =
I'm begining to think this is more of a Cisco config problem than Asterisk, has anyone had a similar problem??? I'm not a Cisco expert so dun't know if I need to enable SIP messageing/reception in the Cisco.
Yeah, pretty sure to enable it you're going to have to upgrade to CCM 5. I could be
Alternativly you can use the exitcontext parameter in the voicemail.conf
to define a separate context in your extensions.conf where the o or a
extensions are handled.
Douglas Garstang [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
-Original Message-
From: Eric ManxPower
I am testing 1.4 and am having trouble with the sound files. The gsm
files are much larger than they used to be. Sox (12.18.2) plays them
back really sllo. Apparently it thinks the
sampling rate is 8000. When I specify -r 48000 it play back properly.
I mention the sox
Hi Folks,
I'm not sure if this is possible, but I'd like to give users the
option of transfering to an employee's cell phone when they get to
their greeting. This is a feature that is common on Nortel KSUs.
Is there an easy way to do this on a per employee basis? I can see
how it can be done
I have seen this if you do not include -c1 for stereo audio files.
--Brian
On Tue, Oct 10, 2006 at 02:31:59PM -0400, Bill Merriam wrote:
I am testing 1.4 and am having trouble with the sound files. The gsm
files are much larger than they used to be. Sox (12.18.2) plays them
back really
Hi allI'm deploying a VoiceMailserver with Asterisk behind a legacy pbx, providing Voicemail to email services for Lecagy PBX extensions.On busy or unanswered calls, Legacy pbx will dial a specific DID (one per extension) to asterisk, and the call is handled by Voicemail application.
I've several
Is that only available in 1.4? 'exitcontext' does not exist anywhere in my
default 1.2.x voicemail.conf file.
-Original Message-
From: LJ [mailto:[EMAIL PROTECTED]
Sent: Tuesday, October 10, 2006 12:24 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: Voicemail
I was able to get pressing '0' while in voicemail to work in a simple test
case, but was unable in a more complicated scenario. Here's a stripped down,
sanitized version of that complex scenario...
[start] ; phones start here
include = some_contexts
include = some_more_contexts
include = route
Hi all..
I have a problem with my asterisk installation. I'm using a Wilcard
X100P clone in Spain. Incoming calls work fine, but when I make a
outgoing call, a hear the ringing, and the peer phone ring, when the
peer answer, asterisk hangup the call, or say busy.
This is my conf:
Polycom 601 with Sip 2.01
Anyone using Sip 2.01? I have upgraded my phones and now presence no longer
functions.
Buddy list shows all phones online but status does not change when someone
is on a call. Also blf does not function.
I am using trixbox, 1.67 was working fine on the same box.
Any
Hello!
In my relentless quest for knowledge, I pose this question: who's got
the biggest
dialplans, and how big are these monsters?
What's the biggest dialplan in use right now? If you feel you are a
competitor,
let me know how many contexts/extensions/priorities you are dealing
with. Maybe the
Simple
Exten = 1234,1,Dial(SIP/[EMAIL PROTECTED],90) ; This will ring thier phone
for 90 seconds
Exten =1234,2,Noop("If user dosent pick
up do something here")
Exten = 1234,102,Dial(SIP/[EMAIL PROTECTED],90) ; WIll ring user B if User is
Busy or hits the reject button
Exten =
You can create a macro that tells the caller that the user is unavailable.
It then asks them if they want to go to the usersVM or be transfred to thier
cell phone.
I also created a macro where users can dial an extension and set thier
mobile number. Let me know if you want it.
-
Steve Murphy wrote:
Hello!
In my relentless quest for knowledge, I pose this question: who's got
the biggest
dialplans, and how big are these monsters?
Sounds interesting. Small facility of 60 users:
-= 161 extensions (597 priorities) in 59 contexts. =-
--
Ben Franklin quote:
Those
Do you want single server stats, or cluster stats?
Single server:
-= 1004 extensions (1403 priorities) in 45 contexts. =-
Aaron
On Tue, 2006-10-10 at 14:16 -0600, Steve Murphy wrote:
Hello!
In my relentless quest for knowledge, I pose this question: who's got
the biggest
dialplans, and
Last machine that I set up is roughly 30 contexts 400 priorotys and 20
extensions. Did it on a dual core 3.0 with 2 gigs of ram and raid 1 sata.
System is a bit of an over kill but client wanted it. Works like a charm. I
know it's not a match for what you have but I figured I would throw it out
Steve is their a CLI command you can make from the console that will
tell you the answer? LOL or are we expected to count?
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Steve Murphy
Sent: Tuesday, 10 October 2006
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