Hi all, I am configuring asterisk in a cable modem network, using a
motorola TM401A.
I can make calls from the MTA but I can receive, display the following
error:
-- Executing [1...@alberti:1] Dial("OSS/dsp",
"MGCP/aaln/1...@0-13-11-82-bd-a.ssw.intercal.net|30") in new stack
[Sep 2 00:10
Am 17.01.2010 11:28, schrieb Tzafrir Cohen:
What is the output of:
ls -l /usr/lib/libspandsp.so*
lrwxrwxrwx 1 root root 19 18. Sep 19:40 /usr/lib/libspandsp.so ->
libspandsp.so.0.0.2
lrwxrwxrwx 1 root root 19 18. Sep 19:40 /usr/lib/libspandsp.so.0 ->
libspandsp.so.0.0.2
-rwx
-= Info about application 'RxFAX' =-
[Synopsis]
Receive a FAX to a file
[Description]
RxFAX(filename[|caller][|debug]): Receives a FAX from the channel
into the
given filename. If the file exists it will be overwritten. The file
should be in TIFF/F format.
The "caller" opti
30.h and /usr/lib/include/spandsp/t30.h and find what
the compile says it can't find, for example t30_terminate).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen
Sent: Tuesday, July 21, 2
Hi Nicholas!
Perhaps, there are other ways as I describe here, but I use this way
successfully about 4 years
- install latest spandsp version
- went to root directory of your svn asterisk
- type "make distclean" (because there are preconfigured things in
downloaded version)
- change to follow
_
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Fa. IT-Connect
Karl Tischler
Tel.+49 941 705
Hans Witvliet schrieb:
On Thu, 2008-01-10 at 08:35 +0100, IT-Connect wrote:
stoffell schrieb:
Has anyone been able to get ISDN-BRI support to work reliably on
Asterisk 1.4? If so, I'd love to know how you did it (hardware,
distro, kernel, modules, versions, config
stoffell schrieb:
Has anyone been able to get ISDN-BRI support to work reliably on
Asterisk 1.4? If so, I'd love to know how you did it (hardware,
distro, kernel, modules, versions, config files).
Maybe your best bet is using bristuff, the bristuff-0.4.0 series are
tests for asteris
I have now within 18 months had a second TDM400P die, the first time was
random call drops, and now it will not go off hook when making a call.
To summarise, the card stopped making calls, I replaced the computer
hardware, installed new OS and new Asterisk (from 1.2 to 1.4) without
making a
a small/medium
business with multiple line support, BLF, etc.
Are the Cisco 7960 the best of the bunch, or Aastra etc?
I have just had a GXP start its daily lockup where you come in in the
morning and it requires a hard reset, and this is getting a little
boring having to keep flashing and
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[EMAIL PROTECTED] wrote:
> Best prices World-Wide
AGAIN?!
I thought they were banned?!
- --
Cheers,
Matt Riddell
___
http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://wap.sineapps.com
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[EMAIL PROTECTED] wrote:
> Hello Matt,
>
> I have not seen how to add a site.
> Could you help me (us) ?
> Tks
When you are in the site list:
Click the link:
http://asterisk.group.stumbleupon.com/sites/
It's titled, View/Add Sites
- --
Cheers,
M
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Hi,
Just thought I'd let people know that I've created a new StumbleUpon
group for Asterisk sites.
If you have a site that is related to Asterisk and is not listed, feel
free to add it.
Alternatively, if you're new to Asterisk and w
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Jason Parker wrote:
> The x86 and x86_64 Digium G.729 codec binaries have been updated for use with
> the Asterisk 1.4 beta (which should also work on current svn trunk).
>
> Anybody that is using the older modules with the 1.4 beta (or svn trunk new
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Jason Parker wrote:
> The x86 and x86_64 Digium G.729 codec binaries have been updated for use with
> the Asterisk 1.4 beta (which should also work on current svn trunk).
>
> Anybody that is using the older modules with the 1.4 beta (or svn trunk new
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The beta of the 1.4 version of Asterisk, LibPRI, Zaptel and
Asterisk-Addons has been released:
http://www.sineapps.com/news.php?rssid=1537
- --
Cheers,
Matt Riddell
___
http://www.sineapps.com/news.php (D
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yusuf wrote:
> That is usually the case, however, there is a feature freeze some time
> before stable releases, and since RTP packetization was only committed
> to trunk on 09-18-06, does that mean that it wont be in 1.4, maybe only
> 1.
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yusuf wrote:
> Hi all,
>
> I'm so excited about 1.4 coming out soon :) , I was wondering if anyone
> can comment on the following:
>
> 1. Will RTP packetization (5162) committed to trunk (43243) be in 1.4?
> I have i
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Roy Sigurd Karlsbakk wrote:
>>> Linked from /. today,
>>> http://www.networkworld.com/news/2006/091206-von-sam-houston.html
>>> talks about the Sam Houston State University (SHSU) migrating a
>>> rather large amount of users to asterisk. The article d
gt; I have RTFM, such as it is, and really don't see how it can be properly
> configured
>
> Would I be better off with the 2100 and tough through the NAT issues?
> Any suggestions??
register them both to freevoip and call each other via the freevoip numbers.
http://freevoip.ged
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The Asterisk Development Team wrote:
> Earlier this week 'refresh' releases of these two projects were put on
> our FTP servers, but due to some miscommunication on our end no
> announcements were sent out... so here they are :-)
:)
All good, Daily A
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Douglas Garstang wrote:
> Sphinx must have been written by the same people as Asterisk then...
Yeah, because the people over at AsteriskDocs wrote Asterisk?
Seriously Doug, if you're looking for a supported commercial
implementation that works with A
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Douglas Garstang wrote:
> What sphinx documentation? All I could find was docs on the code, not on how
> to USE the software.
:) One and the same!
Sphinx is not a commercial application.
I think you might have been mistaken, and are actually lookin
? :))
:)
I kinda knew that you meant that, but was saying that pretty much all
issues can be resolved, you just need to search for the solutions, give
them a tracking number, and let them know you're working on a fix.
Update the customer on the progress, and if possible let them know how
l
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stoffell wrote:
> Hi list,
>
> Any suggestions on how to deal correctly (socially and technically)
> with users "complaining" about features/issues? For instance, users
> complaining about echo; personally I ask the user(s) to give me all
> the detail
rity/limits.conf file and
> then rebooting.
Type
ulimit -n 8192
before starting Asterisk.
ulimit -n without a number will tell you what it is currently set to.
- --
Cheers,
Matt Riddell
___
http://www.sineapps.com/news.php (Daily Asteri
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Boneym wrote:
> Hi,
>
> I would like to test asterisk 1.4 development version , can anyone send me
> a link to it . Thanks in advance.
This would be SVN trunk (http://www.asterisk.org/download):
Commands to check out code from our SV
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Douglas Garstang wrote:
> The docs at that URL say that the dictionary has 'yes' in it... although I
> don't understand how I can get replies like 'YOU HALF' if it doesn't exist in
> the dictionary.
Did you
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Douglas Garstang wrote:
> Anyone played with Sphinx2 and Asterisk much?
>
> I followed the docs at:
> http://turnkey-solution.com/asterisk-sphinx.html
>
> and after getting the server up and running, streamed it some wav files
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Daniel Cyt wrote:
> Hi,
>
> I was having a conversation with some friends and one of them brought up
> the question about find out information about one sip number. Is there
> any way to find out that a number, for instance 1(646)- is actually
>
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Xue Liangliang wrote:
> Hi, all currently we have a requirement from our customer. They want to
> capture the wrap up time for agent, they want the agents status become
> wrap up state automatically after a successful call, and only change
> back to av
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James Jones wrote:
> but in the register program it is staticly linked.
Maybe because you are top posting you didn't see my reply:
>
> Best place to ask this though is:
>
> http://licensing.digium.com/main_page.php
Some help if
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Zeeshan Zakaria wrote:
> Everything was working perfect until I updated CentOS using yum update.
> errors are like these. Please help, what is the solution for this.
> Obviously
> the zaptel hardware is not loading, byt why? How to l
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James Jones wrote:
> Does anyone know why the g729 codec module sold by diguim does not
> display the OpenSSL copyright information. Do they have an agreement
> with OpenSSL to not display the Copyright Information that is required
> ny their license
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William M Conlon wrote:
> I recorded an internet radio program using iTunes, and somehow got an echo.
>
> Anyone have any suggestions on how to remove echo from an existing file?
>
> My wife was on the program, and I promised to re
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Mark Hulber wrote:
> Yes, it worked. I didn't get the announcement of 1.2.9.1.
[Shameless self plug]
Read the Daily Asterisk News, it was announced there with changes today.
:)
[/Shameless self plug]
- --
Cheers,
Matt
llows:
>
> exten => 6274,1,ChanSpy(SIP/403274)
>
> Intra-building we dial 3 digit extensions, monitor is obviously 6+3 digit
> extension
>
> I don't think it's a utilization issue, it seems to happened when there were
> 80 phones on the network, and still
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Arun Kumar wrote:
> hi
>
> thanks for reply.
>
> I'm using vicidial to make calls at 2.0 dial level it is able to make calls
> but when I see the asterisk -r most of the time it shows Outgoing Spool
> Failed. Which Spoo
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Mark Phillips wrote:
> What tools are you using for this?
>
> I'm sure you are aware of SIPp but wondered if you had anything else?
>
> Mark
For IAX2 internally we use a modified version of testcall from
http://iaxclient.sf.net.
Otherwise we just
the
> advice still to use branches 1.2 ???
Use svn branch 1.2 for now.
A beta of 1.4 (svn trunk) will be released later this week.
If you're happy to be using a beta, use it when released later this week.
After 1.4 is released, 1.4 will be the "STABLE" branch (i.e. only
bugfixes)
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Arun Kumar wrote:
> hi
>
> my asterisk -r shows me Most of the times Outgoing Spool Failed. Can some
> one tell me why is it happening and how to solve this issue. Is it a
> problem
> ?
You'd need to provide more information.
the CPU itself and the chipset on the motherboard? If I
> boot into an SMP kernel with Asterisk compiled with the SMP kernel
> source, would it just make use of multi-threading as the load
> increases on cpu-intensive operations?
The best use I have seen is the newly converted IAX2 which
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Kokfoo Soo wrote:
> Does anyone know how many active channels can support for transcoding ulaw to
> G729 by using 4x 3.6GHz Xeon Processors?
In one machine?
I'd guess at around 200-300 absolute max if the calls are spread evenly
across CPUs.
Normal
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Joe Shmoe wrote:
> You say its not your code. But yet, why would you
> actually admit to one of your own leaking it. Well
> some research has been done one the code.. here's what
> we found..
When and where did KPF admit to it be
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Steven Ringwald wrote:
> Mike wrote:
>> Hi all,
>>
>> I just found out how to set the column userfield, in the CDR DB to
>> whatever I desired. Can I add multiple custom columns to the DB and
>> fill them from the
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Kannaiyan Natesan wrote:
>> Digium really did make an effort in this case and it's not worth
>I appreciate Digium and Mark. They have released the
> G729 and G731 source code and you need to pay the license directly to
> voiceage, not d
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Joe shmoe wrote:
> Well you can call me a newb all you want.. The
> software was released to me by a birdie from digium.
> This is just the source code. Nothing more. You
> still need the license for the g729 or g723 but this
> code from digium wil
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We've just completed a couple more interviews with Digium staff:
File (Josh Colp): http://www.sineapps.com/news.php?rssid=1475
Mog (Matthew O'Gorman): http://www.sineapps.com/news.php?rssid=1465
We've got a couple more in the pipeline and I'll post t
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Steve Edwards wrote:
> I'm having trouble getting app_conference to work and I'm feeling
> pretty clueless right know.
Probably the iaxclient list would be the better forum to discuss this as
its not in the Asterisk codebase.
To sign up for the iaxcl
Does a normal phone attached to the line work?
Does it work connected to an FXS socket (i.e. FXO -> FXS)?
Do you see anything when you ring the line?
- --
Cheers,
Matt Riddell
___
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http://fr
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Nir Simionovich wrote:
> I've now also enabled RTP debugging, and noticed that Asterisk doesn't
> send out RTP at all.
> All the lines appear as the following:
>
> Got RTP packet from 62.219.61.73:59436 (type 0, seq 1243, ts -1997588432,
> len 80)
Wh
look right (ie uk like) and look like they are working when I try
> diagnostics with lspci, dsmsg, asterisk -c etc - but when I connect
> the FXO unit to the BT line it just makes it (the BT line) go
> permanently engaged. I'm tearing my hair out and about to chuck it
> all in th
I've tested this scenario via several termination gateways with SIP, and
> always
> there was no RTP in either directions. More then that, when running tcpdump,
> it
> appears as if asterisk isn't even sending any RTP to the outbound SIP gateway.
>
> This was seen o
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ajmcello wrote:
> I'm looking for a way to dial my contacts using a SIP or VOIP gateway in
> Thunderbirds Addressbook. I can do this using Outlook with SIPTAPI, ASTAPI,
> and a couple of others, however, I have not found a way to do so in
> Thunderbird
ng and for this customer are
> charged without any calls.
>
> Any can suggest me how i can stop this issue i checked with my sipprovider
> (MCI) and they monitor the call but the told our GW is not sending call
> connect message while remote is ringing so it means there is some mis
>
e testing out
>> patches waiting for completion?
>>
>> Really, once all the new features have been completed, it will be released.
>>
>> If you would prefer it to be released now (I.E. before everything has
>> been tested and possibly fixed), just download SVN t
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Steve Edwards wrote:
> It's not clear if the OP wanted 1) information on how to analyse the
> core file or 2) provide information to the bug tracker for others to
> analyse.
>
> Matt's answer addresses #2. How about #1?
>
> Anybody care to share thei
hat I sent a text message to
> gets a call with my message (text to speach). I was wondering if there was
> any way for me to detect if a number is a mobile phone or a landline. Is this
> something that only cellular providers can do or can I have asterisk do it (I
> asume I woul
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equis software wrote:
> Hi, I have a Call Center running with safe_asterisk script.
> When Asaterisk crash produce a core file but I don´t know how analyze it!
>
> Any ideas??
http://www.asterisk.org/doxygen/AstDebug.html
- --
Cheers,
Really, once all the new features have been completed, it will be released.
If you would prefer it to be released now (I.E. before everything has
been tested and possibly fixed), just download SVN trunk.
- --
Cheers,
Matt Riddell
___
http://www.sine
now happening on outbound calls
> via ISDN, i.e. calls that don't use IAX2 or the inter-office network. It
> also happens on inbound calls.
Are you using realtime?
- --
Cheers,
Matt Riddell
___
http://www.sineapps.com/news.
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[EMAIL PROTECTED] wrote:
> AGI
> ===
> $res = $AGI->exec("Hangup");
>
> $foo = ${CDR(billsec)};
> myVerbose($foo); #print on CLI
> $foo = ${CDR(duration)};
> myVerbose($foo);
> $foo = ${CDR(ans
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John Millican wrote:
> Hello all,
> I am trying to test if the length of a dialed number is greater than 7. When
> i use:
> exten => 1,n,GoToIf($["${LEN(${numdial})}">"7"]?dialout:nodial);
> and I dial an 11 digit number i.e. 1 800 xxx
> i get t
trary file-overwrite via directory traversal
techniques. The impact of this vulnerability is minimal, however, as it
requires an administrator to use a client-controlled variable as part of
the filename.
Solution:
Mu Security would like to thank the Asterisk security team for their
timely response to
gets the call
redirections after hours, then reload the extensions/pbx_config.
Basically, a MySQL database will contain the phone numbers of
individuals. Another table will have the "Time Of Day" configuration in
it. The extensions.conf will have all the configurations for the
extensions
If price is an issue, then Grandstream is the go.
If quality is the issue, then Snom or Cisco.
(poet laureate)
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Forrest
Beck
Sent: Thursday, 29 June 2006 5:06 AM
To: 'Asterisk Users Mailing List - Non-Commercial Di
I have a network of GXP 2000 phones and would like to know if there is a
way to configure the phones so that if there is one person talking, and
another call comes in then they can hold/hangup that call and take the
incoming call.
At the moment, when a call comes in and the phone is offhook, then
Wouldn't it be easier to do time based context?
i.e. - http://www.voip-info.org/wiki/view/Asterisk+tips+openhours
You can drop the Gotoiftime and create contexts for the hours instead.
For example ...
[default]
include => holiday|*|*|25|dec
include => day|09:00-17:30|mon-fri|*|*
inc
()
I can't get it to work.
Here is my zapata.conf
[channels]context=pstn-linesignalling=fxs_ksechocancel=yesechotraining=800echocancelwhenbridged=norxgain=0.0txgain=0.0immediate=nocallprogress=no;channel => 4
If I uncomment channel => 4, asterisk will not start.
Looking forwar
returned an interesting link. Are there any redirectors
> available?
>
>
Either Asterisk reinviting to the back end or SER if you want software.
Otherwise maybe SIP load balancer (haven't tried it):
http://www.vovida.org/applications/downloads/loadbalancer/
--
Cheers,
Matt Rid
TA firmware images support MWI, one reason
> I've never bought one.
>> Apparently
>> some IP phones based on the PA168V chip has this support already
>> (Atcom AT-320 for example)
> Uses a PA168S.
>> by configuring Asterisk with
>> 'mailboxdetails=yes&
Tristan wrote:
> Hi,
>
> I have troubles setting the userfield in mysql ( using asterisk 1.2.8 /
> addons 1.2.3 )
> I use this in my dialplan:
> exten => s,n,SetCDRUserField(SOMEVALUE)
>
> I tried also:
> exten => s,n,Set(CDR(userfield)=SOMEVALUE)
>
> But everytime i look at the cdr database the
Michael Collins wrote:
>> If you can't afford to purchase both cards, then a safe bet is to
> simply
>> purchase the Sangoma card since it can address more echo issues then
> the
>> Digium card.
>
> Also, don't forget that the high-end A104d has more
> So honestly now, Sahil, what did you guys do that was so different? It
> *really* pisses me off when people like you give a half-assed, half-baked
> "digium sucks" post. If you've got an honest beef with Digium, then sure,
> lay it all out, but don't present
[EMAIL PROTECTED] wrote:
>> Yes you are correct... by default asterisk will >send the call to priority
>
>> N+101... what is your point?
>>
>> You asked about turning off "call waiting".
> In the example that I provided,
>> if the amount of
Andrew D Kirch wrote:
> I am currently running Asterisk 1.2.8, on an AMD Sempron 3100+ (ASUS K8N
> Nforce based board). I am using a Digium wctdm24xxp to terminate 8 (2x
> quad) FXO lines. Using ztmonitor (From zapata) I cannot see that there is
> any registerable incoming volume from these lines
onds, which for me is about 3-4 rings on the phone before it
>> goes looking for the next agent. The other option you can manually
>> remove the interface from the queue via the CLI by the following:
>>
>> remove queue member from
>>
>> However, I'm not sur
Steven wrote:
> I there any good reason that is doesn't get posted to the ftp site?
> People that only use stable may find it easier.
You mean like this?
http://ftp.digium.com/pub/telephony/
--
Cheers,
Matt Riddell
___
http://www.s
Joseph wrote:
> On Thu, 2006-06-01 at 03:45 +0200, Matt Riddell (IT) wrote:
>> Joseph wrote:
>>> Thanks for suggestion, but I'm not looking for software (spyware) type
>>> service.
>>> I to be able to register my asterisk and sipura with the provider
your Asterisk
server and it will even dynamically create extensions.conf and iax.conf
entries for you.
--
Cheers,
Matt Riddell
___
http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://freevoip.gedameurope.com (Free Asterisk Voip C
Joseph wrote:
> What are the alternatives to FWD with IAX2 registration capability.
FreeVoIP:
http://freevoip.gedameurope.com
--
Cheers,
Matt Riddell
___
http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://freevoip.gedameurope.com (
Jean-Michel Hiver wrote:
> Hi List,
>
> Yesterday night after a power off due to a faulty UPS my asterisk
> doesn't want to start anymore. Here is what I get on the CLI:
>
> Asterisk Ready.
> *CLI>
> Disconnected from Asterisk server: Bad file descriptor.
> Executing last minute cleanups
> == De
; you are expecting heavy load, the native format is the way to go. You
> might decide not to use mp3 format at all, recompressing your MoH files
> using sox to the formats you gonna use, such as .al, .ul, .gsm, or leave
> it at .sln to cut the decoding leg only.
Heh, damn this GPRS
Erick Perez wrote:
> should I use mpg123 with asterisk 1.2.7 or should i use the native
> player asterisk has?
> the target machine will receive heavy load.
If the machine will have heavy load then using mp3 files would not be
advised, no matter how you play them. I would recommend using native
f
Curt Shaffer wrote:
> I have been tasked with setting up video conferencing utilizing asterisk.
> One of the requirements is a softset that has video capabilities. Eyebeam
> looks promising but I was just wondering if anyone out there knows of any
> freeware with comparable features of Eyebeam that
http://www.voip-info.org/wiki/view/GXP-2000+Extension+Unit
No release date as yet.
Regards,
Michael Dunne
Corporate IT Solutions Cairns
P.O. Box 2092 Cairns 4870
Ph. +61-7-4051-4377
Fax. +61-7-4051-4390
Mob. +61-400
santosh y wrote:
> I'm very new to Asterisk, I'm tracing the Asterisk code,
> i'm feeling difficulty in understanding the code, so please tell me
> where i can get the documentation of the code and,
> design and architecture of the code.
www.asteriskdocs.org is your best bet. Also sign up to svn-
nd" or "need" faxin as a day to day service on
> their PBX's.
>
> Sad to see that faxing is nearly impossible on digium cards. To me is like
> saying "here you have a great car but.. It cannot handle a car stereo" :(
Is this not possibly also related t
Hi Matt,
Any decent quad, quad-core Opteron system should be able to handle it
with ease! :)
--
Regards,
Hilton Travis Phone: +61 (0)7 3344 3889
(Brisbane, Australia) Phone: +61 (0)419 792 394
Manager, Quark IT http
this point in time, but I'm
keeping a close eye on OpenPBX development.)
--
Regards,
Hilton Travis Phone: +61 (0)7 3344 3889
(Brisbane, Australia) Phone: +61 (0)419 792 394
Manager, Quark IT http://www.quarkit.com.au
Qu
Hi Sam,
There's an Asterisk module for OpenWRT running on the WRT54G and similar
Linksys WiFi routers. Is that embedded enough? :)
--
Regards,
Hilton Travis Phone: +61 (0)7 3344 3889
(Brisbane, Australia) Phone: +61 (0)419 792 394
Manager, Qua
wiki, It definitely works for me in
Australia.
James
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Tim Robinson
> Sent: Saturday, 18 March 2006 02:56
> To: Asterisk Users Mailing List - Non-Commercial Discu
: SIP/201 State:Idle
Watchers 4
200 : SIP/200 State:Busy
Watchers 4
- 5 hints registered
However extension 200 has just hung up and is not in fact busy as it
claims.
I am not sure where to even begin troubleshooting this one. I don't
Ahha ...
That would explain it then, that makes more sense too I guess
considering STRFTIME is quite configurable.
Thanks very much for the help.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
Hulber
Sent: Friday, 3 March 2006 3:35 PM
To
I am using the latest SVN version 1.2 of Asterisk
When I attempt to test the output of certain variables, for use in file
naming etc, certain key ones appear to fail.
exten => ,1,NoOp(${EPOCH})
Returns
-- Executing NoOp("SIP/200-638c", "1141352935") in new stack
exten => 5556,1,NoOp(${TI
gned to those variables it seems.
Is there another location these variables need to be initialised or
should they work out of the box.
I am using the latest svc from Digium.
Regards,
Michael Dunne
----
Corporate IT Solutions Cairns
P.O. Bo
615 "asterisk", fin=0x995740,
> fout=0x407c, ferr=0x407c) at el.c:67
> #1 0x080bc315 in ast_el_initialize () at asterisk.c:1675
> #2 0x080be87b in main (argc=2, argv=0xbff959f4) at asterisk.c:2225
> (gdb)
>
> I just trapped this
> it does not happen every
Dan Journo wrote:
> Ok, i feel like im getting somewhere but i need a little help.
>
> Asterisk displays this when its loading:-
> [res_musiconhold.so] => (Music On Hold Resource)
> == Registered application 'MusicOnHold'
> == Registered application 'WaitMusicOnHold'
> == Registered applica
Christian Benke wrote:
> hello!
>
> has this made it into 1.2.3 already:
> http://bugs.digium.com/view.php?id=6128 ?
>
> i'm trying to set a variable that should be used as a dialstring in the
> dial-command, including parameters seperated with the respective
> del
e target answer, Who can help me?, Which
> event notify me that the phone call was answered?
Which type of channel?
If analogue (tdm400 etc) then the call is answered as soon as it
connects to the pstn, not when the other end answers.
--
Cheers,
Matt Riddell
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