On Wed, May 7, 2008 at 11:06 PM, Russell Bryant [EMAIL PROTECTED] wrote:
Sanjay Rajdev wrote:
I have Asterisk 1.4.15 installed on a Fedora Core 8 machine. Asterisk is
snip
In the dialplan we have used MixMonitor() to record the calls.
Can anyone help me on getting to the root of
Just FYI, I wrote an application that tracks the status of SIP or IAX2
extensions by listening to the AMI. It was for use by callshops but
would probably require minimal change to work for you. It's currently
part of the ASTPP source code.
Darren Wiebe
[EMAIL PROTECTED]
Atis Lezdins wrote:
The only things I set in relation to echo cancellation is in zapata.conf
where I put echocancel=yes
Ouch...any idea what echo cancellation your system is using?
PaulH
On Thu, 2008-05-08 at 14:55 +1000, Lee, John (Sydney) wrote:
the relaxdmtf (or similar) option in zaptel can make this
Thanks for the heads up again guys.
Still no go.
It's a ISDN30 PRI on NTL(Virgin) in the UK
I currently have a Mitel 3300 connected happily sending CallerID's so I
know it the teleco supports it.
The Mitel is set to send 01926xx so that's what I'm trying to get
Asterisk to send.
Running
Tim Guy wrote:
It's a ISDN30 PRI on NTL(Virgin) in the UK
I currently have a Mitel 3300 connected happily sending CallerID's so I
know it the teleco supports it.
The Mitel is set to send 01926xx so that's what I'm trying to get
Asterisk to send.
Running an Openvox D210E that runs
I have this simple queue for the reception set up such that the console
queue has only one agent.
I checked the number in the queue and if there is someone there, I play
back a busy please be patient message and then join the call to the
queue.
If there is no one in the queue, the caller will go
Ciao Matt,
Are you using IAX2 as your transport between the 2 servers or SIP?
If you are using IAX2, are you using Asterisk 1.4.18.1 or 1.4.19.1 on either
machine? If so, you may be encountering the IAX2 bug that some have been
discussing on the list recently you can read it here:
On Thu, May 8, 2008 at 5:40 AM, Steve Totaro
[EMAIL PROTECTED] wrote:
Can this thread be moved to the biz list? It really does not belong
here when words such as the best way to monetize an application or
The topic is still salient IMO, but again much posted here is opinion :)
The word
On Thu, May 8, 2008 at 11:25 AM, Lee, John (Sydney)
[EMAIL PROTECTED] wrote:
I have this simple queue for the reception set up such that the console
queue has only one agent.
I checked the number in the queue and if there is someone there, I play
back a busy please be patient message and
On 8 May 2008, at 09:36, randulo wrote:
On Thu, May 8, 2008 at 5:40 AM, Steve Totaro
[EMAIL PROTECTED] wrote:
Can this thread be moved to the biz list? It really does not belong
here when words such as the best way to monetize an application or
The topic is still salient IMO, but again
On Thu, 8 May 2008, Lee, John (Sydney) wrote:
I have this simple queue for the reception set up such that the console
queue has only one agent.
I checked the number in the queue and if there is someone there, I play
back a busy please be patient message and then join the call to the
queue.
Hi All,
Whats the SLN file format (for import/export to Audacity)?
Need to avoid Sox if I can
Adrian Marsh
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On 8 May 2008, at 11:00, Adrian Marsh wrote:
Hi All,
Whats the SLN file format (for import/export to Audacity)?
Need to avoid Sox if I can
16 bit signed audio at 8khz.
2 bytes per sample, no compression, 8000 samples per second, network
byte order, no header.
Tim.
On Thu, May 08, 2008 at 10:54:36AM +0100, Tim Panton wrote:
On 8 May 2008, at 09:36, randulo wrote:
On Thu, May 8, 2008 at 5:40 AM, Steve Totaro
[EMAIL PROTECTED] wrote:
Can this thread be moved to the biz list? It really does not belong
here when words such as the best way to
On Thu, May 08, 2008 at 11:00:18AM +0100, Adrian Marsh wrote:
Hi All,
Whats the SLN file format (for import/export to Audacity)?
Need to avoid Sox if I can
export it as wav. It's basically the same as SLN, but with an extra
header that tells everyone what the exact format is. Thus
Tzafir,
It's not working it, here's what the utilities told me:
[EMAIL PROTECTED]:~# invoke-rc.d asterisk stop
* Stopping Asterisk PBX: asterisk
...done.
[EMAIL PROTECTED]:~# invoke-rc.d zaptel stop
Unloading zaptel hardware drivers:.
[EMAIL PROTECTED]:~# /usr/share/zaptel/xpp_fxloader usb
Tzafrir Cohen wrote:
The only downside is that you can simply concatenate two files using
'cat file1 file2 file1file2' with wav as you can with raw formats
(provided that both originals are of the same format), because the
header is not part of the stream.
Correction for the archives ... you
JR Richardson wrote:
I have 1.4.9.1 setup, with the compiler flags enabled for T38, and
have a Mediatrix 2102 and a Linksys SPA 8000-G1. I can pass faxes
between devices but can't seem to invoke T38 pt UDPTL. It's enabled
in sip.conf [general] and well as the [peer].
I get an error
On Thursday 08 May 2008 00:52:35 Steve Totaro wrote:
I can certainly post more dirt from Mark's previous right hand man
if you wish to continue this argument.
I'd enjoy the chance to debunk the myths that you've heard. So keep it
coming.
Why would Mark build a PBX from scratch for his
Julian,
Thanks for the information. We'll wait for a new version, then.
Kevin
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Yap
Sent: Wednesday, May 07, 2008 5:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Tzafrir Cohen wrote:
The only downside is that you can simply concatenate two files using
'cat file1 file2 file1file2' with wav as you can with raw formats
(provided that both originals are of the same format), because the
header is not part of the stream.
Correction for the
Thank Russell, I will try to manage it through the modules.conf file.
Regards,
Sanjay Rajdev
- Original Message -
From: Russell Bryant [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, May 8, 2008 4:11:00
I had a problem in the dictate app, which I have fixed. Thanks for the help.
By the way here is a description of what was happening.
app_dictate does not close the file descriptor after the call hangs or a new
dictation starts, as and when the dictation increased the count of open file
On Thu, May 08, 2008 at 09:23:29AM -0400, David Backeberg wrote:
Tzafrir Cohen wrote:
The only downside is that you can simply concatenate two files using
'cat file1 file2 file1file2' with wav as you can with raw formats
(provided that both originals are of the same format), because
On Thu, May 08, 2008 at 08:33:08AM -0300, Guilherme Loch Waltrick Góes wrote:
Tzafir,
It's not working it, here's what the utilities told me:
[EMAIL PROTECTED]:~# invoke-rc.d asterisk stop
* Stopping Asterisk PBX: asterisk
...done.
[EMAIL PROTECTED]:~# invoke-rc.d zaptel stop
David Backeberg schrieb:
Tzafrir Cohen wrote:
The only downside is that you can simply concatenate two files using
'cat file1 file2 file1file2' with wav as you can with raw formats
(provided that both originals are of the same format), because the
header is not part of the stream.
Sanjay Rajdev wrote:
I am not a developer for Asterisk and even cannot make changes in the SVN as
I do not know lot about the branches in it, but if someone from your side can
take the effort to change this It would be great help for others.
Please open a report on http://bugs.digium.com
Philipp Kempgen wrote:
Just out of curiosity:
I can't remember when I last had to concatenate 2 sound files.
So why does this always come up? IMHO it's one of those things
you hardly ever need.(?)
I can't remember the last time I have done that. :)
Anytime I need to do something like that,
Hi all,
I am using a simple perl script to connect with ast manager api. the script
tries to set a channel variable. It extracts the channel name from the
events it recieves after dial command. When i try to set the channel
variable, asterisk responses with an error saying that the channel does
Just out of curiosity:
I can't remember when I last had to concatenate 2 sound files.
So why does this always come up? IMHO it's one of those things
you hardly ever need.(?)
It's all about how you define need. Obviously anybody can make
multiple script entries to play multiple files. My
To this end, I have taken a first pass at a Perl AGI script to look up
and return a list of URIs for a given phone number. I will not pretend
that I have read the relevant RFCs but have implemented based on the
knowledge I have gathered about ENUM lookups from various sources.
Given my dialplan
Well database really is a bottleneck for me. I am currently trying to do
rating stuff in agi using perl. What im doing is i lookup the rate of every
dialed code for every call from the mysql database using the longest match
technique. The longest match technique costs atleast 2-3 mysql queries for
7 maj 2008 kl. 21.11 skrev Anthony Francis:
[EMAIL PROTECTED] wrote:
On my SIP carrier, I register to a proxy
sipconnect.dal0.cbeyond.net
which ends up being 192.168.22.212 (They supply a T1 bundle)
#sip show peers
Name/username HostDyn Nat ACL Port
Which phones are you using and what software revision. I've had a crash
course in Snom phone lately and can probably help with at least the park
orbits.
-Brent
Thermal Wetland wrote:
I would like to hire someone to help us tweak our asterisk system for
Snom phones.
We would like to
My exact requirement.. to edit out some recorded hiss and then put the
file back...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Russell
Bryant
Sent: 08 May 2008 15:37
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Russell Bryant schrieb:
Philipp Kempgen wrote:
Just out of curiosity:
I can't remember when I last had to concatenate 2 sound files.
So why does this always come up? IMHO it's one of those things
you hardly ever need.(?)
I can't remember the last time I have done that. :)
Anytime I
On Thu, May 8, 2008 at 7:07 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Thu, May 08, 2008 at 11:00:18AM +0100, Adrian Marsh wrote:
Hi All,
Whats the SLN file format (for import/export to Audacity)?
Need to avoid Sox if I can
export it as wav. It's basically the same as
On Thu, May 08, 2008 at 09:36:50AM -0500, Russell Bryant wrote:
Philipp Kempgen wrote:
Just out of curiosity:
I can't remember when I last had to concatenate 2 sound files.
So why does this always come up? IMHO it's one of those things
you hardly ever need.(?)
I can't remember the last
On Thu, May 08, 2008 at 07:39:39PM +0500, Rizwan Hisham wrote:
Hi all,
I am using a simple perl script to connect with ast manager api. the script
tries to set a channel variable. It extracts the channel name from the
events it recieves after dial command. When i try to set the channel
G'day all,
I'm trying to come up with a quick, easy solution to have a static
inbound number in my dialplan, rotate calling 2 numbers. Example:
1st call into asterisk
exten = 1234,1,Dial(sip/,10)
exten = 1234,n,Dial(sip/,10)
2nd call into asterisk
exten = 1234,1,Dial(sip/,10)
Brian J. Murrell wrote:
Does anyone have a better ENUM lookup handler than the built-in
ENUMLOOKUP() function? The built-in function does not properly handle
multiple return values such as:
8.9.9.3.2.8.8.6.6.8.1.e164.org has NAPTR record 200 10 u E2U+SIP
!^\\+1866(.*)$!sip:[EMAIL
Tilghman Lesher wrote:
On Thursday 08 May 2008 00:52:35 Steve Totaro wrote:
I can certainly post more dirt from Mark's previous right hand man
if you wish to continue this argument.
I'd enjoy the chance to debunk the myths that you've heard. So keep it
coming.
Why would
On Thu, 2008-05-08 at 11:46 -0400, Paul Belanger wrote:
G'day all,
I'm trying to come up with a quick, easy solution to have a static
inbound number in my dialplan, rotate calling 2 numbers. Example:
1st call into asterisk
exten = 1234,1,Dial(sip/,10)
exten =
On Thu, 2008-05-08 at 10:51 -0500, Russell Bryant wrote:
Have you taken a look at the ENUMQUERY() and ENUMRESULT() functions that are a
part of Asterisk 1.6?
I have not even entertained thinking of 1.6 yet. :-/
The ENUMQUERY() function lets you do a single enum query for a number. Then,
Paul Belanger schrieb:
I'm trying to come up with a quick, easy solution to have a static
inbound number in my dialplan, rotate calling 2 numbers. Example:
1st call into asterisk
exten = 1234,1,Dial(sip/,10)
exten = 1234,n,Dial(sip/,10)
2nd call into asterisk
exten =
Hello...
We're attempting to track down an intermittent echo issue. Our setup is
phonesipasterisksiptntpri to carriers. We have less than 2 ms latency on
the networks (FTTx), totally SIP w/ G711u. The party hearing the echo is the
subscriber using sip. The PSTN users does not hear the echo.
I've found an interesting link. It might help you out.
http://www.cisco.com/en/US/docs/ios/solutions_docs/voip_solutions/EA_ISD.html
Peter
Joe Carroll wrote:
Hello...
We're attempting to track down an intermittent echo issue. Our setup is
phonesipasterisksiptntpri to carriers. We have
On Thursday 08 May 2008 11:03:34 John Novack wrote:
Tilghman Lesher wrote:
On Thursday 08 May 2008 00:52:35 Steve Totaro wrote:
I can certainly post more dirt from Mark's previous right hand man
if you wish to continue this argument.
I'd enjoy the chance to debunk the myths that you've
Brian J. Murrell wrote:
I have not even entertained thinking of 1.6 yet. :-/
Fair enough. That's why I pointed out the feature.
Dude! Where were you yesterday, before I spent a few hours last night
writing my AGI? :-)
Sorry. I have trouble keeping up with this list. :)
Now that's
Tilghman Lesher wrote:
On Thursday 08 May 2008 11:03:34 John Novack wrote:
Tilghman Lesher wrote:
On Thursday 08 May 2008 00:52:35 Steve Totaro wrote:
I can certainly post more dirt from Mark's previous right hand man
if you wish to continue this argument.
I'd enjoy the chance to debunk the
An option to rotate between numbers is to add a queue to the system
and add and as agents and pick the proper strategy (rrmemory
or leastrecent). This has some advantages:
- the calls are devided as you have in mind
- when there are more calls coming in they are queued instead of
I have a situation here where a user has an AAstra 480i phone, which
function corectly. The phone is behing a nat-router (a linksys wrv200 for
it's VPN point to point facility). The phone is plugued in a port wich has
qos enabled.
And when the user places a call, sometimes (not always), we get
I am still having a very frustrating problem win an Avaya-Asterisk
system. I have written about this before but I am expanding the
description of the problem just in case someone can give me some
insight.
This installation is an Asterisk 1.4.19.1 server connected to an Avaya
PBX
I do link the idea of have a queue answer the calls and route to the
extensions, but will have to figure out a way to do this with have the
SIP extensions logging into the queues.
On Thu, May 8, 2008 at 1:53 PM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
An option to rotate between numbers is to
On Tue, May 06, 2008 at 02:13:02PM -0400, Matt Watson wrote:
That's fine... honestly I hate the message myself, however corporate
policy is corporate policy so there isn't much of a point in
discussing it.
That being said, the message does clearly say that the message is for
the named
Two things you could consider trying:
1) In sip.conf, set the externip and localnet parameters correctly.
2) Also in sip.conf, try the following on the PAP2's sections:
disallow=all
allow=alaw:10
In case that fails, try also
disallow=all
allow=alaw:20
Att
Vinícius Fontes
Desenvolvimento
I have narrowed the problem to a parameter called Symmetric RTP on
the SPA3102. If I disable that I will get the same one way audio
problem as the PAP2T. Unfortunately it seems that the Symmetric RTP
parameter is only available on the SPA3102 and not on the PAP2T. I got
this definition
Which Cepstral voice is best for Asterisk?
We need to license one.
Regards,
Sanjay Rajdev
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david-8khz and the regular david aren't bad in my experience.
On Thu, May 8, 2008 at 4:54 PM, Sanjay Rajdev
[EMAIL PROTECTED] wrote:
Which Cepstral voice is best for Asterisk?
We need to license one.
Regards,
Sanjay Rajdev
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I second the queues idea. You can make static queues including the
sip channels. The previous mentioned ideas, while they may work, are
a little more intricate than then the queue idea.
I think that if you do not need 1.4, 1.2 has much less bugs than 1.4.
This conclusion is not from experience
I have a client that is using the Sangoma A200DE with two phone lines
attached.
The problem is:
They use their phone (Grandstream GXP2020) to dial out of the system.
Instead of getting ringing, there is someone on the other end of the
line that happened to dial in at the exact same moment.
Well.. Now I'm confused.
Recap.. Couldn't appear to get out going callerid to work on a UK NTL
PRI connection.
Id been testing it with my Orange Mobile phone.. Dial the 07973xx
and it displays private.
Called my girlfriend tonight on our land line (all be it NTL again but
this time
At 5:22 PM on 08 May 2008, Forrest Beck wrote:
I have a client that is using the Sangoma A200DE with two phone
lines attached.
The problem is:
They use their phone (Grandstream GXP2020) to dial out of the system.
Instead of getting ringing, there is someone on the other end of the
Hello All,
Recently, I build three Asterisk 1.4 box and installed licensed copy of
G729 codec. Before installing the G729 codec I tested the MOH on all
three Asterisks box and it was working fine. So I install G729 codec and
retested MOH and it was all wavy... Meaning the music was going up
I know that everyone has gaps in their knowledge, but I am just
staggered that
systems are being sold/deployed with such fundamental TELCO workings not
being
understood. Frightening.
C. Chad Wallace wrote:
At 5:22 PM on 08 May 2008, Forrest Beck wrote:
I have a client that is using the
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim Guy
Sent: 08 May 2008 22:28
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Out-Going Callerid
Well.. Now I'm confused.
Hmm.. Just phoned the sprogs mobile of
After many days of testing I finally found the problem. It turns out
that Asterisk was ignoring the externip setting in sip.conf. Today I
decided to enable externhost with the FQDN of the server and magically
the PAP2T started working!
On Thu, 2008-05-08 at 16:38 -0300, Vinícius Fontes
Hello...
We're attempting to track down an intermittent echo issue. Our setup is
phonesipasterisksiptntpri to carriers. We have less than 2 ms latency
on the networks (FTTx), totally SIP w/ G711u. The party hearing the echo is
the subscriber using sip. The PSTN users does not hear the
We are looking for a female voice.
Regards,
Sanjay Rajdev
- Original Message -
From: Matthew Gibson [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, May 9, 2008 2:35:24 AM GMT +05:30 Chennai, Kolkata,
On Thu, 8 May 2008, Paul Belanger wrote:
I do link the idea of have a queue answer the calls and route to the
extensions, but will have to figure out a way to do this with have the
SIP extensions logging into the queues.
You can define a device to be a member of a queue in queue.conf. For
They have demos of all the voices on their site..
On Thu, May 8, 2008 at 6:25 PM, Sanjay Rajdev
[EMAIL PROTECTED] wrote:
We are looking for a female voice.
Regards,
Sanjay Rajdev
- Original Message -
From: Matthew Gibson [EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Sanjay Rajdev wrote:
We are looking for a female voice.
I use Callie-8KHz.
Never much cared for Alison so I tried most of them from the demo site
and found Callie to be the smoothest/calmest sounding.
You can download the demo version of the software and try them on the
system(s) they will
Ok it's all fixed! A silly mistake in my zaptel.conf meant that
packets destined for the foneBridge device were leaving the wrong
interface... So although I could confirm that I was recieving packets
from the fonebridge and everything appeared green, any packets
destined for the device were never
Runnig the xpp_fxloader before the Zaptel and Asterisk scripts solves the
problem.
thank you Tzafir.
On Thu, May 8, 2008 at 11:07 AM, Tzafrir Cohen [EMAIL PROTECTED]
wrote:
On Thu, May 08, 2008 at 08:33:08AM -0300, Guilherme Loch Waltrick Góes
wrote:
Tzafir,
It's not working it, here's
Ok, I''ll bite. The question is:
Do we want asterisk to contain a licensing engine ?
That depends on the implementation. Your questions, I'm sure, will be
discussed on the call tomorrow.
Such an engine would need to :
Hand out license tokens to proprietary modules linked to
asterisk
Hi
I have a little probelm with my ip phone and asterisk, i dont know where
can i look?
When i place a call o receive a call, after talk or the other side
finish talk, we both side hear ss (noise).
i have installed the last zaptel branches and the last asterisk
branches, 6 digium card
Queue(console,r)
would do what you want, but so you would need to have two entry points
to
queue.
Thanks Atis. Your suggestion did magic!
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dmesg | grep -i zap
Should give you a version, and an echo cancellation technology.
PaulH
On Thu, 2008-05-08 at 17:05 +1000, Lee, John (Sydney) wrote:
The only things I set in relation to echo cancellation is in zapata.conf
where I put echocancel=yes
Ouch...any idea what echo
The Allison voice is nice and matches with the built in recordings fairly well.
Steve
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I never kick myself on issues like this. I enjoy the challenge and
the eventual success by jumping around and yelling YES, YES, YES!
and the sad part is I am all alone. ;-)
Thanks,
Steve Totaro
On Thu, May 8, 2008 at 7:20 PM, Mike Hardman
[EMAIL PROTECTED] wrote:
Ok it's all fixed! A silly
On Thu, May 8, 2008 at 5:20 AM, Brent Davidson [EMAIL PROTECTED]
wrote:
Which phones are you using and what software revision. I've had a crash
course in Snom phone lately and can probably help with at least the park
orbits.
-Brent
Brent,
We have the phones in the lab, we have 1 - 370,
Steve Prior's mention of using Allison's voice with Cepstral reminded me
to ask: for a listing of the text for the built-in recordings.
I found a web page but I'd prefer not having to scrape the info out of
it. I didn't notice anything while wandering through the source code/files.
I want to
Roderick A. Anderson wrote:
Steve Prior's mention of using Allison's voice with Cepstral reminded me
to ask: for a listing of the text for the built-in recordings.
I found a web page but I'd prefer not having to scrape the info out of
it. I didn't notice anything while wandering through
We've inherited a pair of mostly identical PBX systems, each with a
TDM400P Rev I boards and 4 FXO modules. The production system is
running Asterisk-Now with 1.4.9, and despite some other issues, it
is able to answer inbound calls just fine. The replacement system is
currently running
Russell Bryant wrote:
Roderick A. Anderson wrote:
Steve Prior's mention of using Allison's voice with Cepstral reminded me
to ask: for a listing of the text for the built-in recordings.
I found a web page but I'd prefer not having to scrape the info out of
it. I didn't notice anything
Hi,
I am not sure if this is the right forum to ask for this. If not, let's me
say sorry for my mistake in advance.
Does anyone know where I can find a copy of t38modem that can work with
Opal? And which Opal version should I use?
Any help or hint will be greatly appreciated.
Thanks,
Mark
I think his connect/disconnect is going to take far longer than his 3
queries.
The fact that Asterisk doesn't support sustained MySQL connection from
the DialPlan
is in fact quite a big deal that Digium seems to have its head in the
sand about.
And one of those things that SHOULD come up in
Perhaps this should be tagged under Is * Ready For Prime Time ? Thread
Isn't an 'appliance' supposed to be a 'plug-it-in-and-runs' sort of thing ?
Julian Yap wrote:
On Tue, May 6, 2008 at 1:38 AM, Benoit Plessis [EMAIL PROTECTED] wrote:
We are actually running an AsteriskNow appliance
On Thursday 08 May 2008 23:19:19 Al Baker wrote:
The fact that Asterisk doesn't support sustained MySQL connection from
the DialPlan
is in fact quite a big deal that Digium seems to have its head in the
sand about.
And one of those things that SHOULD come up in those Is * Ready For
Prime
Take a big shot of Valium before dealing with the bug tracker folks.
There idea of help is to post You have an extra space in your line
then CLOSE the ticket.
That kind of clear, specific help is just what my doctor ordered to keep
my BP nice and low
Benoit Plessis wrote:
Tilghman Lesher a
Hi,
Could anyone explain to me what is this Warning mean and how can I overcome
this
*[May 9 12:53:39] WARNING[3626]: codec_zap.c:155 zap_framein: G.729B CNG
frame received but is not supported; dropping.
[May 9 12:53:39] WARNING[3626]: translate.c:211 framein: zapg729toulaw did
not update
On Thursday 08 May 2008 23:38:14 Al Baker wrote:
Take a big shot of Valium before dealing with the bug tracker folks.
There idea of help is to post You have an extra space in your line
then CLOSE the ticket.
That kind of clear, specific help is just what my doctor ordered to keep
my BP nice
dmesg | grep -i zap
Should give you a version, and an echo cancellation technology.
Thanks Paul.
# dmesg | grep -i zap
Zapata Telephony Interface Registered on major 196
Zaptel Version: 1.4.6
Zaptel Echo Canceller: MG2
Zaptel Transcoder support loaded
Which is reasonably new, but an upgrade to the latest version (1.4.10.1)
will only take 5 minutes and is worth a shot.
PaulH
On Fri, 2008-05-09 at 15:24 +1000, Lee, John (Sydney) wrote:
dmesg | grep -i zap
Should give you a version, and an echo cancellation technology.
Thanks Paul.
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