Hi,
Has anyone seen this before?
Sep 12 22:31:38 WARNING[17472]: codec_ilbc.c:175 ilbctolin_framein:
Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP
(4)?
I found a bug tracker http://bugs.digium.com/view.php?id=6333 talking
about this, but didn't really understand why it
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
I tried a lot of SIP and IAX softphones looking for ones I liked, noticing
some have certain features and others did not. For things like call
transfer, call park, group pick-up, line presence, and all those kinds of
extras I have a
Hi Arik,
I have Asterisk running as guest on my Debain Xen system and it works
fine. I used to work with an AVM Fritz!PCI ISDN card as well by compiling
the ISDN driver (Hisax) in the XEN-kernel. You have to be aware that the
host system does not use the ISDN PCI card by putting
On Tue, 2006-09-12 at 16:39 +0200, Olivier wrote:
Hi,
What would you suggest to implement directed call pickup on bristuffed
Asterisk 1.2 ?
I'm after tle ability to pick a specific ringing call (without caring
about which call arrived first, for example).
Something like : *8 + local
On Tue, Sep 12, 2006 at 09:55:17PM -0700, Christopher Corn wrote:
Christopher Corn [EMAIL PROTECTED] wrote:
i've got trixbox installed and grandstream 101 phones.
out of my 4 phones, one of them has call waiting working.
they all the same version of firmware and settings. i tried
Hello - Hope someone can help on this. We are experiencing incredibly high latency with IAX calls; however not so under SIP calls with identical setups. We have tried multiple IAX clients. We have tested with two Asterisk setups - one on the local network running on a OS X and another on a remote
On 09/13/06 07:22 Ronald Wiplinger said the following:
I need h.264 and tried therefore svn checkout
http://svn.digium.com/svn/asterisk/trunk asterisk
asterisk supports h.264 in passthru mode. we've tested this with eyebeam
video SIP clients without problems.
--
Regards,
well this is my guess :) i had call features (call waiting) enabled on my budgetone 100. i guess when i would dial *70 to enable call waiting, it wasn't reaching my asterisk server. I then turned off call features on my phone then when I would dial *70, I could hear a voice response telling me
Hi,
is there anyone working on VXML or CCXML integration for asterisk?
If not, anyone interested in developing it?
-- Arnd
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Hi Steve,
I will try to make how you suggest...to use G instead of g but the
problem remains inside Asterisk. If there is a dynamic channel
allocation made by Asterisk, that message is non-sense.if telco
sends a call on channel X and Asterisk has a free channel Y, Asterisk
should use
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Giorgio Incantalupo wrote:
Hi Steve,
I will try to make how you suggest...to use G instead of g but the
problem remains inside Asterisk. If there is a dynamic channel
allocation made by Asterisk, that message is non-sense.if telco
sends a
Kevin Smith wrote:
Here is what the configuration looks like for one of the phones, the
other is 284:
[283](Empire-Defaults)
[EMAIL PROTECTED]
[283a](Empire-Defaults) [EMAIL PROTECTED]
[283b](Empire-Defaults)
[EMAIL PROTECTED]
So actually you are trying to use one phone to monitor
On 9/12/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote:
I'm interested to understand why I many messages like:
WARNING[21314] chan_zap.c: Ring requested on channel 0/1 already in use
on span 1. Hanging up owner
How can a channel be already in use??? That means the channel is
busy...if it is
Hi list!
I have few questions about queue and persistent queue members.
If there is queue with only one persistent member, what happens if it doesn't
answer the phone for timeout = 10 seconds? Calling person still waits in
queue and what happens with agent? Will his phone ring after retry = 20
Hi Steve,
I agree with you..telco knows better!
If telco sends a ring on channel X and asterisk has already used it,
couldn't asterisk shift that call on another channel Y or it is
obliged to answer on channel X?
In other words, if asterisk get a ring on channel 3 and channel 3 is in
use,
On 9/13/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote:
Hi Steve,
I agree with you..telco knows better!
If telco sends a ring on channel X and asterisk has already used it,
couldn't asterisk shift that call on another channel Y or it is
obliged to answer on channel X?
The telco is in
Hello,
Has anyone noticed that the Polycom IP430 has a low incoming/outgoing
sound level?
Is it a firmware issue or should I adjust my zap's tx/rxgain?
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To
Hello everyone,
What would be the best way to solve this error on ARI?
We are using ARI version 00.08.04 on an [EMAIL PROTECTED] server.
--
Stand before it and there is no beginning.
Follow it and there is no end.
Stay with the ancient Tao,
Move with the present.
I am trying to connect a Definity G3 to an asterisk system. I had it
working OK with the exception of the caller ID on the Definity handsets
just showing the trunk-group name, this was using Switchvox. Prior to
that, I used Trixbox and caller ID was fine on the phones. Just to make
sure
Hi list!
Does anyone have experiences with the updated model of the Kirk IP600?
The V3 model is supposed to support SIP instead of only SCCP or H323 which
would make the use with Asterisk a lot easier.
I have only tested the Kirk IP600 V2 with SCCP / Skinny protocol which is
still giving me
Actually the problem was somewhere in the Cisco equipment, as the service provider has confirmed. Some option in their device to conserve bandwidth by compressing voice data was causing this choppyness. As they've turned this option off now, MoH works perfect.
Matt Arnilo S. Baluyos (Mailing Lists) wrote:
Hello everyone,
What would be the best way to solve this error on ARI?
We are using ARI version 00.08.04 on an [EMAIL PROTECTED] server.
Check the asterisk readme.
___
--Bandwidth and Colocation
Hello,
I had received a lot of unreadable pages with rxfax. I've been doing
some search on net and found this:
http://threebit.net/mail-archive/asterisk-users/msg15708.html
It looks like rxfax/spandsp doesn't support ecm error correction. Bad
news for me. Is it still the case? app_rxfax.c dated
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
To start the ball rolling:
Software: Zaptel 1.0.9, Asterisk 1.0.9, BRIstuff-0.2.0, wanpipe-2.3.2
PRI interface: Sangoma A101U (UK E1)
Phones on sites with NO problems: snom, elmeg, Aastra, Linksys/Sipura
Phones on problem site:
Hi
as i have posted before iam trying to integrate IVR with Balance
exten = 888,1,Read(${CALLERIDNUM})exten = 888,2,MYSQL(Connect connid 127.0.0.1 rootpassword database)exten = 888,3,MYSQL(Query resultid ${connid} select\ saldo\ from\ balance\ where\ username=${CALLERIDNUM})
exten =
VAD maybe was caussing this.
Regards
Zeeshan Zakaria escribió:
Actually the problem was somewhere in the Cisco equipment, as the service
provider has confirmed. Some option in their device to conserve
bandwidth by
compressing voice data was causing this choppyness. As they've turned this
OK, I thought that it started with 1.2, but that is also when I started using
freePBX. (to make it easier for support staff)
So, I take from your comments that you have heard of this before and that it
may be hardcoded into freePBX.
I will check there.
--
--
Steven
Artifex Maximus wrote:
Hello,
I had received a lot of unreadable pages with rxfax. I've been doing
some search on net and found this:
http://threebit.net/mail-archive/asterisk-users/msg15708.html
It looks like rxfax/spandsp doesn't support ecm error correction. Bad
This is correct. You'll
spandsp supports 9600 rx and does not support ecm. If you want ecm, use
iaxmodem with hylafax - http://iaxmodem.sourceforge.net , currently hylafax
in conjunction with iaxmodem seems to be more reliable than rxfax and
spandsp by themselves.
Craig
- Original Message -
From: Artifex
ram wrote:
Hi
as i have posted before iam trying to integrate IVR with Balance
exten = 888,1,Read(${CALLERIDNUM})
exten = 888,2,MYSQL(Connect connid 127.0.0.1 http://127.0.0.1
root password database)
exten = 888,3,MYSQL(Query resultid ${connid} select\ saldo\ from\
balance\ where\
Hi,Is there somewhere a sample configuration for asterisk as gateway (pri - isdn)Thanks
Stay in the know. Pulse on the new Yahoo.com. Check it out.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To
Hi,
I'm wondering how to configure asterisk the right way to handle 305 Use
Proxy correctly.
I've asterisk sitting as registrar and rtp-proxy and so on in front of
two opensers. These both opensers redirect INVITEs as soon as their load
is beyond a configure limit to each other. And they
It is hardcoded into extensions.conf of FreePBX. You can change it to some other sound which you like. You can check the sounds located in /var/lib/asterisk/sounds. Or remove this line altogether from the code.
___
--Bandwidth and Colocation provided by
Hello ppl,
I am getting the following errors when accessing voicemails
Sep 13 16:43:59 ERROR[19020]: app.c:1161 ast_lock_path: Unable to create
lock file '/var/spool/asterisk/voicemail/pbx1VmBoxes/555123/Old': No
such file or directory
Sep 13 16:43:59 ERROR[19020]: app.c:1196 ast_unlock_path:
On 9/13/06, Forum Expansive [EMAIL PROTECTED] wrote:
What is the latest polycom firmware and where can I get it?
sip2.0.1, ask your reseller, they must give it to you.
cheers
___
--Bandwidth and Colocation provided by Easynews.com --
Benjamin Jacob wrote:
Hello ppl,
I am getting the following errors when accessing voicemails
Sep 13 16:43:59 ERROR[19020]: app.c:1161 ast_lock_path: Unable to
create lock file
'/var/spool/asterisk/voicemail/pbx1VmBoxes/555123/Old': No such file
or directory
Just as the error states, the
I have queue with member defined as:
member = Agent/SIP/148,1
member = Agent/SIP/143,2
And when I do show queues this is what I see.
pbx*CLI show queues
prodaja has 0 calls (max 5) in 'rrmemory' strategy (0s holdtime), W:10, C:0
, A:1, SL:0.0% within 60s
Members:
Agent/SIP/148
Steve Totaro wrote:
I am trying to connect a Definity G3 to an asterisk system. I had it
working OK with the exception of the caller ID on the Definity
handsets just
I'll give this to our Definity guy and see what he suggests. I'll
report back
Doug
--
Ben Franklin quote:
Those who
Steve Totaro wrote:
I am trying to connect a Definity G3 to an asterisk system. I had it
working OK with the exception of the caller ID on the Definity
handsets just
He wants to know if your Definity is an S, SI or an R?
Doug
--
Ben Franklin quote:
Those who would give up Essential
is there anyone working on VXML or CCXML integration for asterisk?
I've integrated OpenVXI 3.4 (the latest one) with Asterisk for a
client. It is now in production, interpreting their VXML pages
using Asterisk for SIP/IAX telephony (but could use anything).
They don't require ASR for now, so I
Doug Lytle wrote:
Steve Totaro wrote:
I am trying to connect a Definity G3 to an asterisk system. I had it
working OK with the exception of the caller ID on the Definity
handsets just
He wants to know if your Definity is an S, SI or an R?
Doug
I am not sure, it is a refrigerator sized
Steve Totaro wrote:
Doug Lytle wrote:
Steve Totaro wrote:
I am trying to connect a Definity G3 to an asterisk system. I had
it working OK with the exception of the caller ID on the Definity
handsets just
He wants to know if your Definity is an S, SI or an R?
Doug
I am not sure, it is a
Do you have queues/agents configured?
Tomislav Parčina wrote:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
To start the ball rolling:
Software: Zaptel 1.0.9, Asterisk 1.0.9, BRIstuff-0.2.0, wanpipe-2.3.2
PRI interface: Sangoma A101U (UK E1)
Phones on sites with NO problems:
I am using asterisk branch 1.2 revision 42600 and zaptel 1.9 and I am
having a strange problem -- if I hit * as the first dtmf once a call
is answered, everything dies, no audio from the other end and nothing
works again till the other endhangs up. I get the following log
entries:
Sep 13
One of my clients is saying that this happens after a queue agent
performs an attended transfer. Has anyone else seen this?
Tomislav Parčina wrote:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
To start the ball rolling:
Software: Zaptel 1.0.9, Asterisk 1.0.9, BRIstuff-0.2.0,
Craig Doug,
Thanks for your info. I'll do that way.
Is there any chance for implementing ecm in rcfax/spandsp? I think
using rxfax is more friendly than using a modem emulator connected
through a virtual device to a fax software. It's sound as a very
bizarre way to me. :-)
bye,
Zsolt
On
Thank you for the suggestion; I ran strace (which is an awesome util to
discover in its own right, thanks Jeff!), and here's the relevant
section of output:
ioctl(4, 0x40044a03, 0xbfd4840c) = 0
write(4, \0\0\20)r2\230\34)\3\315\373#\2\231\5\277\1\206\0\211\7..., 4000) = 4000
read(4,
Steve Totaro wrote:
Steve Totaro wrote:
Doug Lytle wrote:
Steve Totaro wrote:
I am trying to connect a Definity G3 to an asterisk system. I had
it working OK with the exception of the caller ID on the Definity
handsets just
He wants to know if your Definity is an S, SI or an R?
Doug
I
Hello,
You don't need Agent. Use this instead:
member = SIP/148,1
member = SIP/143,2
Agent is for members defined in agents.conf.
bye,
Zsolt
On 9/13/06, Tomislav Parčina [EMAIL PROTECTED] wrote:
I have queue with member defined as:
member = Agent/SIP/148,1
member = Agent/SIP/143,2
And when
Hi,
I use IAX2 trunk between two asterisk server.
At a few calls (less than 30) enviorment, both caller and callee hear each
other clearly. But when calls reach 45 or above, the quality of sounds is bad.
I wonder if a IAX2 Trunk should limit concurrent calls?
I use ILBC codec in the trunk.
Artifex Maximus wrote:
using a modem emulator connected
through a virtual device to a fax software. It's sound as a very
bizarre way to me. :-)
Funny how it seemed to be a very straight-forward approach to me.
Lee.
___
--Bandwidth and Colocation
Hi
guys,
i have asterisk
1.0.9 with bristuff 0.2.0 rc8n running on a VIA motherboard with processor C3
and i have this kind of problem: during the office time the system work
perfectly, but on the next moring, if i try to make an outgoing call i get this
message
== Primary
D-Channel on
Hi,Is there somewhere a sample configuration for asterisk as gateway (pri - isdn)Thanks
Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+ countries) for 2¢/min or less.___
--Bandwidth and Colocation provided by
Hi list,
I've got to be honnest and admit that I don't know the SIP-related RFC's
by heart. So my question may seem completely heretic to some of you.
What I want to know is whether it's possible to set, somewhere in
Asterisk's configuration files, the host part of the Contact header for
INVITE
Rene wrote:
Hi Arik,
I have Asterisk running as guest on my Debain Xen system and it works
fine. I used to work with an AVM Fritz!PCI ISDN card as well by compiling
the ISDN driver (Hisax) in the XEN-kernel. You have to be aware that the
host system does not use the ISDN PCI card by putting
ram wrote:
exten = 888,8,Playback(${AMOUNT-DUE})
Change this from Playback to SayNumber
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SayNumber
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither
exten = 888,8,SayDigits(${AMOUNT-DUE})
B
ram wrote:
Hi
thanks for the quick reply
yes as suggested i did the Following modification
exten = 888,1,Read(${CALLERIDNUM})
exten = 888,2,MYSQL(Connect connid 127.0.0.1 http://127.0.0.1 root
password database)
exten = 888,3,MYSQL(Query
On 9/13/06, Steve Totaro [EMAIL PROTECTED] wrote:
Doug Lytle wrote:
Steve Totaro wrote:
I am trying to connect a Definity G3 to an asterisk system. I had it
working OK with the exception of the caller ID on the Definity
handsets just
He wants to know if your Definity is an S, SI or an R?
On 9/13/06, Michael Welter [EMAIL PROTECTED] wrote:
Tomislav Parčina wrote:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
To start the ball rolling:
Software: Zaptel 1.0.9, Asterisk 1.0.9, BRIstuff-0.2.0, wanpipe-2.3.2
PRI interface: Sangoma A101U (UK E1)
Phones on sites
I've followd the instructions as in tutorials, created folder stream, created file stream.mp3, in musiconhold.conf added 'stream = mp3:/var/lib/asterisk/mohmp3/stream,http://216.126.84.50:8000
', and in extensions.conf added
exten = 466,1,Answerexten = 466,n,MusicOnHold,streamBut when I try to
Hi bails
thanks for the reply
just now i have changed to the same and replying to list
your mail came with the same answer
thanks for the help
after anouncing mysql connection will be closed right ?
Ram
___
--Bandwidth and Colocation provided by
Hi
thanks i have rectified the problem
ram
On 9/13/06, Doug Lytle [EMAIL PROTECTED] wrote:
ram wrote: exten = 888,8,Playback(${AMOUNT-DUE})Change this from Playback to SayNumber
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SayNumberDoug--Ben Franklin quote:Those who would give up
BJ Weschke wrote:
On 9/13/06, Steve Totaro [EMAIL PROTECTED] wrote:
Sounds like a G3R. How are you signaling between the two? PRI?
Actually, we've discovered it's a DefinityG3 V4
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
ram wrote:
Hi
thanks for the quick reply
yes as suggested i did the Following modification
exten = 888,1,Read(${CALLERIDNUM})
exten = 888,2,MYSQL(Connect connid 127.0.0.1 http://127.0.0.1 root
password database)
exten = 888,3,MYSQL(Query resultid ${connid} select\ saldo\ from\
balance\
BJ Weschke wrote:
On 9/13/06, Steve Totaro [EMAIL PROTECTED] wrote:
Doug Lytle wrote:
Steve Totaro wrote:
I am trying to connect a Definity G3 to an asterisk system. I had it
working OK with the exception of the caller ID on the Definity
handsets just
He wants to know if your Definity is
Hi,
I have a 2 E1 system with 32 zap FXS extensions (all Zaptel, with
TDM2400), on a PIV, 3GHz, 1GB,
Well my question is wether I'll be able to use it for peak demand moment,
that is having all 60 channels busy 30 talking to agents on the FXS,
while recording their conversation at the
Hello
Is there a possibility for sending an event on the managerinterface
(AMI) when a call is put on/off hold?
Or is there any other way to detect when a call is placed on hold?
Michael
___
--Bandwidth and Colocation provided by Easynews.com --
Hello (again)
When doing a STATUS request on the AMI, I get informations like these:
Event: Status
Privilege: Call
Channel: SIP/311-08b97790
CallerID: 310
CallerIDName: Aastra
Account:
State: Ringing
Uniqueid: 1157967815.19
My problem is that I want to tie my own information to a call,
Hi all
Same like reading Numbers
how can read words
since i dont see India Currency anouncing ( i see Dollars)
i want to anounce after 17 then Rupees
how can i achive this
Ram
On 9/13/06, bails [EMAIL PROTECTED] wrote:
exten = 888,8,SayDigits(${AMOUNT-DUE})Bram wrote: Hi thanks for the quick
It would be nice if someone could do that but I doubt it will happen.
Hylafax / iaxmodem is more complicated and more effort to set up than rxfax
but the end result is worth the effort. My only criticism is that I set up
2 x E1's on a server (60 channels) and I didn't enjoy having to configure
Hi, is there a
variable hold the ip address of the incoming sip call ?
Thanks
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
On 9/13/06, Steve Totaro [EMAIL PROTECTED] wrote:
BJ Weschke wrote:
On 9/13/06, Steve Totaro [EMAIL PROTECTED] wrote:
Doug Lytle wrote:
Steve Totaro wrote:
I am trying to connect a Definity G3 to an asterisk system. I had it
working OK with the exception of the caller ID on the Definity
MF wrote:
Hi,
I have a 2 E1 system with 32 zap FXS extensions (all Zaptel, with
TDM2400), on a PIV, 3GHz, 1GB,
Well my question is wether I'll be able to use it for peak demand moment,
that is having all 60 channels busy 30 talking to agents on the FXS,
while recording their
You have to leave a message in the voicemail, then listen it and the error
will not apear again.
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Doug Lytle
Enviado el: Miércoles, 13 de Septiembre de 2006 08:45 a.m.
Para: Asterisk Users Mailing List -
I am having half channel audio after about 6 minutes. asterisk 1.2.11
and 1.2.8 zaptel.
My call is originating on a a SIP extension going out to nufone to my cell.
After about 6 minutes the SIP extension can still hear me on my cell
but I cannot hear the SIP extension.
If the SIP extension hits
Artifex Maximus wrote:
Hello,
I had received a lot of unreadable pages with rxfax. I've been doing
some search on net and found this:
http://threebit.net/mail-archive/asterisk-users/msg15708.html
It looks like rxfax/spandsp doesn't support ecm error correction. Bad
news for me. Is it still
Hi all,
is there any possibility of setting the global variables from outside of
asterisk?
Like manager api or something like that.
Thanks a lot
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or
Hello Steve,
On 9/13/06, Steve Underwood [EMAIL PROTECTED] wrote:
Artifex Maximus wrote:
Hello,
I had received a lot of unreadable pages with rxfax. I've been doing
some search on net and found this:
http://threebit.net/mail-archive/asterisk-users/msg15708.html
It looks like
Thanks Raphael
I'll plan to upgrade to amd64 right away, but in the mean time do you
think working with PIV could be a major problem? that is is it
probable that this setting will miserably fail with a PIV ?? Or
just a tight situation where it might work with some limitations when
Raphaël Jacquot wrote:
MF wrote:
Hi,
I have a 2 E1 system with 32 zap FXS extensions (all Zaptel, with
TDM2400), on a PIV, 3GHz, 1GB,
Well my question is wether I'll be able to use it for peak demand moment,
that is having all 60 channels busy 30 talking to agents on the FXS,
while
Any pointers about on how to get around this build problem in Zaptel 1.2.9?
/usr/src/zaptel-1.2.9/wct4xxp/fw2h
/usr/src/zaptel-1.2.9/wct4xxp/OCT6114-128D.ima
/usr/src/zaptel-1.2.9/wct4xxp/vpm450m_fw.h
make[3]: *** No rule to make target
BJ Weschke wrote:
On 9/13/06, Steve Totaro [EMAIL PROTECTED] wrote:
BJ Weschke wrote:
On 9/13/06, Steve Totaro [EMAIL PROTECTED] wrote:
Doug Lytle wrote:
Steve Totaro wrote:
I am trying to connect a Definity G3 to an asterisk system. I
had it
working OK with the exception of the
Try, http://www.soft-switch.org/downloads/snapshots/snapdsp,
On 9/13/06, Artifex Maximus [EMAIL PROTECTED] wrote:
Hello Steve,On 9/13/06, Steve Underwood [EMAIL PROTECTED] wrote: Artifex Maximus wrote: Hello, I had received a lot of unreadable pages with rxfax. I've been doing
some search on
Try this one:
http://www.soft-switch.org/downloads/snapshots/spandsp/
- Original Message -
From: Artifex Maximus [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, September 13, 2006 11:33 PM
Subject: Re:
Hi Bruce,
Looks like your typing is as bad as mine :-)
Try http://www.soft-switch.org/downloads/snapshots/spandsp
Steve
Bruce Reeves wrote:
Try, http://www.soft-switch.org/downloads/snapshots/snapdsp
http://www.soft-switch.org/download/snapshots/snapdsp,
On 9/13/06, *Artifex Maximus*
Ok I see, thanks for your info Steve,
Is the Mux problem because of processor time or Disk access time?
because if it is a processor thing, I should think on a solution
where, instead of sending the files over , I just send a command and
from the other machine Mux them on a mapped
On Wed, Sep 13, 2006 at 12:00:27PM -0400, Mark Hulber wrote:
Any pointers about on how to get around this build problem in Zaptel 1.2.9?
Get 1.2.9.1, that has fixed exactly that.
(and improvd Astribank drivers, thanks Kevin)
--
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755
Use SVN and not the tarball.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Mark Hulber
Sent: Wednesday, September 13, 2006 12:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Building
Anyone using that product with success?
Bill
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Hi, it's possible to implement a callback without agi?, i'm trying this
but * exits without dialing (if I hungup during s,3 wait) but if it
hungs in s,4 it dials, so is there an explanation to this behavior?
there is an alternative to do it? just for learning
thanks for your answers
On 9/13/06, Steve Underwood [EMAIL PROTECTED] wrote:
Artifex Maximus wrote:
If you look in http://www.soft-switch.org/download/snapshots/snapdsp,
the latest snapshot of spandsp and the app_rxfax and app_txfax
applications there provide ECM. It is less well tested than the
spandsp-0.0.2 code,
Interesting article I found linked from Groklaw:
Sam Houston State University replaces Cisco CallManagers, Nortel PBXs
with Linux-based VoIP and messaging servers
http://www.networkworld.com/news/2006/091206-von-sam-houston.html?page=1
Doug
--
Ben Franklin quote:
Those who would give up
I tried the had a problem with an NFS dropping while writing audio
directly from the monitor app and asterisk froze for about 10 seconds
and then became responsive again. I write all the recording un-MUXed
locally and run a cron that ftps them. We currently have 12 T1s turned
up, not sure of
These are just PSTN calls, and I have set busydetect=no and
callprogress=no in zapata.conf as per voip-info guidance, but problem
persists.
CPU load never breaks 20, so that doesn't seem to be the problem, but it's a 1.2Ghz Athlon with 768MB RAM.
Power supply to system is clean, there's no heavy
uh.. maybe recording a gsm file with rupees and playing it just after SayDigits.. ?
2006/9/13, ram [EMAIL PROTECTED]:
Hi all
Same like reading Numbers
how can read words
since i dont see India Currency anouncing ( i see Dollars)
i want to anounce after 17 then Rupees
how can i achive this
On Wed, Sep 13, 2006 at 12:33:01PM -0400, Steven Totaro wrote:
Use SVN and not the tarball.
Digium updated to 1.2.9.1 earlier this week.
Steve
--
NetTek Ltd UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM/Gizmo/Mac
Hi
is this possible to read words like digits in asterisk
Ram
On 9/13/06, Héctor Maldonado [EMAIL PROTECTED] wrote:
uh.. maybe recording a gsm file with rupees and playing it just after SayDigits.. ?
2006/9/13, ram [EMAIL PROTECTED]:
Hi all
Same like reading Numbers
how can read words
Yes, it worked. I didn't get the announcement of 1.2.9.1.
MARK.
Tzafrir Cohen wrote:
On Wed, Sep 13, 2006 at 12:00:27PM -0400, Mark Hulber wrote:
Any pointers about on how to get around this build problem in Zaptel 1.2.9?
Get 1.2.9.1, that has fixed exactly that.
(and improvd
They need to document the exact day and
time so you can look in the logs. Is this a T1 or POTS?
Customers always complain and threaten to
go back to their old PBX. First, calm down, then calm them down and make sure
they know you are working on it. Every new install is going to have
I think what matters is what directory the file resides in. They are
all wav or gsm.
ram wrote:
Hi
is this possible to read words like digits in asterisk
Ram
On 9/13/06, *Héctor Maldonado* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
uh.. maybe recording a gsm file with
1 - 100 of 160 matches
Mail list logo
