Re: [asterisk-users] app_swift 3 and asterisk 1.8.13.0 fails with undefined symbol: swift_port_close
both would be appreciated. if you can send me a backtrace, that'd be great On Jun 22, 2012, at 8:06 PM, Jeremy Kister wrote: On 6/20/2012 8:24 AM, Darren Sessions wrote: I just finished replying to your direct email (which you can disregard now as this seems to be a different problem). I'm pretty sure I know what the issue is, but I'll have to get back to you later this evening (my time). I have a different problem- i just compiled app_swift 3 from the new git repo for asterisk 1.8.13.0 asterisk loads the module fine, but as soon as i try to swift anything, asterisk core dumps. i'll be glad to post the corefile or sample extensions.conf if desired. -- Jeremy Kister http://jeremy.kister.net./ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_swift 3 and asterisk 1.8.13.0 fails with undefined symbol: swift_port_close
Hi Jakob, I just finished replying to your direct email (which you can disregard now as this seems to be a different problem). I'm pretty sure I know what the issue is, but I'll have to get back to you later this evening (my time). - D On Jun 20, 2012, at 4:41 AM, Jakob-Matthias Böttger wrote: Hi, i am trying to install the just from git cloned app_swift version. Compiling works fine. Install as well. But if i try to load the module at Asterisk it fails with. Command 'module load app_swift.so ' failed. [Jun 20 11:29:51] WARNING[24217]: loader.c:460 load_dynamic_module: Error loading module 'app_swift.so': /usr/lib/asterisk/modules/app_swift.so: undefined symbol: swift_port_close [Jun 20 11:29:51] WARNING[24217]: loader.c:850 load_resource: Module 'app_swift.so' could not be loaded. My System Informations: server*CLI core show version Asterisk 1.8.13.0 built by root @ server on a x86_64 running Linux on 2012-06-20 08:55:14 UTC root@server:~# uname -r 3.2.0-25-generic root@server:~# ldd /usr/lib/asterisk/modules/app_swift.so linux-vdso.so.1 = (0x7fff6d3ff000) libc.so.6 = /lib/x86_64-linux-gnu/libc.so.6 (0x7f2010a65000) /lib64/ld-linux-x86-64.so.2 (0x7f2011041000) root@server:~# cat /etc/ld.so.conf.d/swift.conf /opt/swift/lib root@server:~#ldconfig -v | grep swift /opt/swift/lib: libswift.so.6 - libswift.so.6.0 libceplex_de.so.6 - libceplex_de.so.6.0 libceplang_de.so.6 - libceplang_de.so.6.0 root@server:~# swift -V Cepstral Swift v6.0.1, March 2012 Default Voice: Matthias-8kHzv6.0.0 Language: German v5.1.0 Lexicon:unknown v0.0.0 Concurrency:1 Port(s) Registered 0 Port(s) In Use Distribution: No audio distribution license was found. Saving audio to a file is disabled. Copyright (C) 2000-20012, Cepstral LLC. Do You have any Ideas why that won't work? Best Regards Jakob Böttger -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] app_swift beta release
Hi folks, Just a note to let everyone know I've finally finished up the new BETA release of app_swift (now v3.0.1 b1). This release introduces some pretty major changes to app_swift such as: - The entire code-base has now been unified and the build system auto detects which Asterisk version you're using (yay! one branch!) - Auto-detection and support for both the Cepstral 5.0 and 6.0 engines - Support for Asterisk 1.4, 1.6, 1.8, 1.8 Certified, and 10 - Asterisk 1.2 support has been dropped. I have only been able to do some basic testing with all these permutations of Asterisk and the Cepstral engines on a few of my machines here at the house and need some volunteers to help out and be guinea-pigs. Please email me directly with any feedback you might have. I've updated my github repo with the new app_swift code which can be downloaded using git. git clone git://github.com/dmsessions/app_swift.git Thanks, - D -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] app_swift tts module - new home.
Hi Folks, After receiving a surprising amount of emails from Asterisk community members, I thought I'd fire something off to the users list to clear any confusion regarding the Asterisk Forge (forge.asterisk.org) website and the future of the app_swift text-to-speech module. With regards to the Asterisk Forge website redirecting to GitHub, this has been a long time coming. Emails were sent out to the various lists warning folks that the hosted GForge site was going away - so no one should be too surprised - 'nuf said there. As far as the app_swift project is concerned, with the exception of moving things around as far as location, it is business as usual. The app_swift code for *all* the different versions of Asterisk is now being hosted on GitHub at https://github.com/dmsessions/app_swift . This is a good thing and will make life easier. btw, I love git. If you don't yet, you will . . someday soon . . Individual tar files for each of the different versions of app_swift, which is what 99% of people are going to want, are all available for download on my website at http://www.darrensessions.com by clicking the 'Downloads' button at the very top of the page. That is all my friends. Seasons Greetings! - Darren -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] app_swift for Asterisk 10
Hey there folks, I'd sent this to the list last night and got reject email this morning. Apparently it is always a good idea to have an active subscription to the list you are trying to post to - just one of those things. :) In any case, a new beta version of app_swift is available for Asterisk 10. I put it up in the Asterisk Forge on the 25th of last month, but wanted to wait to post something on the users list until I'd had a chance to really test it a bit (so far so good). http://forge.asterisk.org/gf/project/app_swift/frs/ I have to say, the combination of Asterisk 10 and this latest version of app_swift is absolutely the best sounding of any release to-date! I've been *very* impressed so far. Also, just fyi . . there are some extremely minor tweaks I'll be back-porting to the other app_swift versions shortly. I hope to get that done this weekend or next depending on my free time. Enjoy, - Darren -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Load Balance and Failover
You could use a sip proxy front end like Kamailio. Sent from my iPhone On Nov 18, 2010, at 7:39 AM, Antônio Theóphilo anto...@freeddom.com wrote: Hi All Does anyone know about any tool that does to Asterisk what mod_jk does for JBoss/Tomcat: a load-balance/failover server that is constantly connected to Asterisk backend servers and is capable of identify loaded or down servers? Regards Antônio Theóphilo -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cepstral voice quality
Well, the downside to wav files is the disk i/o. Asterisk will and does translate the audio frames from ulaw to whatever other codec. Sent from my iPhone On Oct 24, 2010, at 9:42 AM, Zeeshan Zakaria zisha...@gmail.com wrote: Do you recommend using wav files instead? Will there be any downside of using wav? Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) -- _ -- Bandwidth and Colocatio... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cepstral voice quality not good
Are you using app_swift or wav files? On Oct 23, 2010, at 5:26 PM, Zeeshan Zakaria zisha...@gmail.com wrote: Hello list, I have been using Cepstral's 8KHz voices for my text-to-speech service for some time now, and have been noticing that the voice quality is really poor, doesn't matter what phrase I give it to convert. None of the other 8KHz voices I have ever used were this bad. It doesn't seem good enough system to be used in a commercial system. Is there any better quality text-to-voice engine? Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] app_swift for Asterisk 1.8
Just thought I'd let everyone know I've got a new beta version of app_swift up for Asterisk 1.8 on http://forge.asterisk.org. - Darren -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] app_swift v2.0 released
Hi all, Thought I'd mention that the new version of the app_swift text-to-speech module for Asterisk 1.2, 1.4, and 1.6 has been released at it's new home on the Asterisk Forge. http://forge.asterisk.org/gf/project/app_swift/ For those that are unaware, app_swift provides a direct interface with the Cepstral text-to-speech engine so instead of having to call the Cepstral engine and write then read an audio file (i.e. disk I/O), you can call the library directly and stream the audio straight to the Asterisk channel. Additionally, the app_swift module supports DTMF detection with a max digits and timeout value as well (similar to the AGI get data functionality). The new version of app_swift has been built and tested on the latest releases of Asterisk for each of their respective code-bases (1.2.40, 1.4.32, and 1.6.2.8) using the Cepstral 5.x libraries. Any questions or feedback, please let me know. Thanks, - Darren -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe Conferencing - Announce your own join/leave to yourself and other conference members
Well, never mind on this (didn't get any responses anyways). I basically
removed the meetme announcement options and wrote the functionality from
scratch into my AGI framework along with an announcement queuing daemon that
runs continuously every second in the background that generates a call file and
plays back the user name recording. Hasn't added any CPU overhead with the call
processing and along with working as intended I think there maybe some other
unique capabilities for it down the road.
In any case, thought I'd update the thread.
Cheers,
- Darren
On Jan 11, 2010, at 10:05 AM, Darren Sessions wrote:
Hi all,
I'm trying to get the MeetMe system to take a caller and announce to them
they've joined the conference in addition to the other members of the
conference assuming previous members of the conference = 1.
I can see where the meetme.c app actually processes it using the
ast_pthread_create_background(conf-announcethread, NULL, announce_thread,
conf); function. The problem is that it's passing the conf data and not the
chan data so it filters out the new caller to the conference and announces
the caller's name to the rest of the conference with the announce_thread
function. Without the chan data available, it makes quick and dirty hacks
even impossible without more insight into the structure of the app ( i was
thinking of just adding a seperate ast_streamfile / ast_waitstream with the
chan variable using an if current-announcetype == CONF_HASJOIN or something
like that).
Unless I'm missing a way to pass the Asterisk API function
ast_pthread_create_background more than one argument and then modify the
announce_thread to accommodate it, I'm at a bit of a loss on a good way of
doing this without making Asterisk seg fault.
The second idea I had was to use a simple conf-background.agi (below) and do
it that way while altering how meetme is called from the actual separate
conferencing agi app. This method does work for announcing the user but the
separate channels refuse to mix audio afterwards (and I have tried every
trick in the book I can think of with this one down to EAGI stuff). If I take
the 'b' option off of the MeetMe call in the AGI script, the audio passes
perfectly. Additionally, attempts at using the manager interface to unlock,
unmute, etc. the conference have no effect. Aside from the audio (obviously a
big deal), the script runs as designed (DTMF detection, etc.).
Any ideas or help would be appreciated.
Many thanks,
- Darren
POI:
Asterisk 1.6.1.6
app_meetme.c - line 1601 (the announce_thread function)
app_meetme.c - line 1817 (the conf_run function)
-- snip --
#!/usr/bin/perl -w
use strict;
use warnings;
use lib '/var/lib/asterisk/agi-bin';
use DBI;
use Asterisk::AGI;
our ($AGI,%v,%ast);
$AGI = new Asterisk::AGI;
%ast = $AGI-ReadParse();
$v{chan} = $ast{channel};
$v{lang} = $AGI-get_variable('CHANNEL(language)');
$v{conf} = $AGI-get_variable('conference_call');
$v{dbh} = sanitized
($v{q},$v{r}) = undef;
$v{q} = SELECT members FROM sanitized WHERE confno = '.$v{conf}.';
$AGI-verbose($v{q});
$v{q} = $v{dbh}-prepare($v{q});
if (!$v{q}-execute) {
exit;
}
$v{r} = $v{q}-fetchrow_hashref();
$v{q}-finish();
$v{dbh}-disconnect;
if ($v{r}{members} 1) {
$AGI-stream_file(/var/spool/asterisk/meetme/meetme-username-.$v{conf}.-.$v{r}{members});
}
while (!$v{loop}) {
exit if (!$AGI-channel_status($v{chan}));
$v{rc} = $AGI-wait_for_digit('6');
}
exit;
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[asterisk-users] MeetMe Conferencing - Announce your own join/leave to yourself and other conference members
Hi all,
I'm trying to get the MeetMe system to take a caller and announce to them
they've joined the conference in addition to the other members of the
conference assuming previous members of the conference = 1.
I can see where the meetme.c app actually processes it using the
ast_pthread_create_background(conf-announcethread, NULL, announce_thread,
conf); function. The problem is that it's passing the conf data and not the
chan data so it filters out the new caller to the conference and announces the
caller's name to the rest of the conference with the announce_thread function.
Without the chan data available, it makes quick and dirty hacks even impossible
without more insight into the structure of the app ( i was thinking of just
adding a seperate ast_streamfile / ast_waitstream with the chan variable using
an if current-announcetype == CONF_HASJOIN or something like that).
Unless I'm missing a way to pass the Asterisk API function
ast_pthread_create_background more than one argument and then modify the
announce_thread to accommodate it, I'm at a bit of a loss on a good way of
doing this without making Asterisk seg fault.
The second idea I had was to use a simple conf-background.agi (below) and do it
that way while altering how meetme is called from the actual separate
conferencing agi app. This method does work for announcing the user but the
separate channels refuse to mix audio afterwards (and I have tried every trick
in the book I can think of with this one down to EAGI stuff). If I take the 'b'
option off of the MeetMe call in the AGI script, the audio passes perfectly.
Additionally, attempts at using the manager interface to unlock, unmute, etc.
the conference have no effect. Aside from the audio (obviously a big deal), the
script runs as designed (DTMF detection, etc.).
Any ideas or help would be appreciated.
Many thanks,
- Darren
POI:
Asterisk 1.6.1.6
app_meetme.c - line 1601 (the announce_thread function)
app_meetme.c - line 1817 (the conf_run function)
-- snip --
#!/usr/bin/perl -w
use strict;
use warnings;
use lib '/var/lib/asterisk/agi-bin';
use DBI;
use Asterisk::AGI;
our ($AGI,%v,%ast);
$AGI = new Asterisk::AGI;
%ast = $AGI-ReadParse();
$v{chan} = $ast{channel};
$v{lang} = $AGI-get_variable('CHANNEL(language)');
$v{conf} = $AGI-get_variable('conference_call');
$v{dbh} = sanitized
($v{q},$v{r}) = undef;
$v{q} = SELECT members FROM sanitized WHERE confno = '.$v{conf}.';
$AGI-verbose($v{q});
$v{q} = $v{dbh}-prepare($v{q});
if (!$v{q}-execute) {
exit;
}
$v{r} = $v{q}-fetchrow_hashref();
$v{q}-finish();
$v{dbh}-disconnect;
if ($v{r}{members} 1) {
$AGI-stream_file(/var/spool/asterisk/meetme/meetme-username-.$v{conf}.-.$v{r}{members});
}
while (!$v{loop}) {
exit if (!$AGI-channel_status($v{chan}));
$v{rc} = $AGI-wait_for_digit('6');
}
exit;
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Re: [asterisk-users] app_swift installation problems
What version of Asterisk and what version of app_swift?
On 29 Oct 2008, at 15:10, [EMAIL PROTECTED] wrote:
Hi, I have tried installing app_swift on both mac os x and ubuntu now
and am getting the same error. I must be missing something, as I have
tried multiple versions and everytime do sudo make install i get:
if ! [ -f /etc/asterisk/swift.conf ]; then \
install -m 644 swift.conf.sample /etc/asterisk/swift.conf ; \
fi
if [ -f app_swift.so ]; then \
install -m 755 app_swift.so /usr/lib/asterisk/modules ; \
fi
and when i do just sudo make, it spits out a ton of junk, this is at
the end:
/usr/lib/gcc/i486-linux-gnu/4.2.4/include/stddef.h:214: error:
declaration for parameter ‘size_t’ but no such parameter
app_swift.c:451: error: expected ‘{’ at end of input
make: *** [app_swift.o] Error 1
Im not sure whats going on here, i have setup asterisk and gotten it
configured with the x-lite soft phone, so i know that is working. I
am ultimately trying to use adhearsion to integrate with my rails
app. I have also installed cepstral voices and these work in the
terminal so i am confident that is also installed correctly.
Thanks.___
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Re: [asterisk-users] Asterisk and voice recognition
Not sure about the Swedish, but Lumenvox has a great speech recognition app for Asterisk. - D On 26 Oct 2008, at 19:53, Christian wrote: Hi all, Yes, this might not be the proper list for this, but i have a question about Asterisk and voice recognition. If I want to create a menu system where the user can say different things in the Swedish language what should I look at? For example, i want the user to be able to say something simular in Swedish: connect disconnect help and so on. Best regards and thanks, Christian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and voice recognition
Sphinx http://cmusphinx.sourceforge.net/html/cmusphinx.php Not sure how the implementation works with Asterisk but I know it's been done (I'd google it). - D On 26 Oct 2008, at 20:55, Christian wrote: Hi, Many thanks for that info. Is there any free solution available as well? Many thanks, Christian On 2008-10-26 at 20:32 Darren Sessions wrote: Not sure about the Swedish, but Lumenvox has a great speech recognition app for Asterisk. - D On 26 Oct 2008, at 19:53, Christian wrote: Hi all, Yes, this might not be the proper list for this, but i have a question about Asterisk and voice recognition. If I want to create a menu system where the user can say different things in the Swedish language what should I look at? For example, i want the user to be able to say something simular in Swedish: connect disconnect help and so on. Best regards and thanks, Christian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Load Balancing
One other thing you could try would be to use OpenSIPS and use a standard config that routes to a hostname (with a creative failure route setup). You'd then setup the hostname in DNS as multiple SRV records reflecting your pool of Asterisk servers (set your TTL very low for these records). You could have something like sipsak send test messages every 30 seconds or so to each of the Asterisk servers. If one quits responding, then the monitoring app updates your DNS servers removing the effected Asterisk server from the DNS pool and effectively from the usable gateway pool. I actually wrote one of these ages ago that worked fairly well with a10 calls per second SER server. How many calls per second are you looking to process? - D _ Darren Sessions [EMAIL PROTECTED] http://www.darrensessions.com _ On Oct 4, 2008, at 9:59 PM, John D wrote: Hi all, I've googled around for concrete solutions on load balancing Asterisk, and it appears there are several ways to skin this cat -- but not one solution which is all appealing. I have the following requirements, which aren't anything extraordinary: * I need to handle roughly 300 simultaneous phone calls to start * Eventually scale to 1000 simultaneous phone calls * I want to be able to pull out an entire server from the cluster without affecting my application * I'm doing all my trunking over SIP So far I've seen folks mention the use of DUNDi and OpenSER(Now OpenSIPS), but unfortunately the documentation out there is rather sparse and lacks detail for someone who isn't extremely keen with the intricate details of Asterisk or OpenSIPS. Would anyone be able to suggest a good starting point in as far as reading documentation and testing out some solutions? I'd also be up for hiring a consultant to help me get started -- but I believe the proper forum for that is asterisk-biz. (Which I've already posted to). Thank you for your insight on load balancing Asterisk. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Load Balancing
I know. :) I've already mentioned some of the OpenSIPS options to him on the OpenSIPS users list (LCR module specifically). Just brain dumping everything that came to mind. - D _ Darren Sessions [EMAIL PROTECTED] http://www.darrensessions.com _ On Oct 4, 2008, at 10:31 PM, Alex Balashov wrote: OpenSIPS/Kamailio have modules designed specifically for that kind of functionality now without a need for an outside monitoring process or SRV reliance. Darren Sessions wrote: One other thing you could try would be to use OpenSIPS and use a standard config that routes to a hostname (with a creative failure route setup). You'd then setup the hostname in DNS as multiple SRV records reflecting your pool of Asterisk servers (set your TTL very low for these records). You could have something like sipsak send test messages every 30 seconds or so to each of the Asterisk servers. If one quits responding, then the monitoring app updates your DNS servers removing the effected Asterisk server from the DNS pool and effectively from the usable gateway pool. I actually wrote one of these ages ago that worked fairly well with a10 calls per second SER server. How many calls per second are you looking to process? - D _ Darren Sessions [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] http://www.darrensessions.com _ On Oct 4, 2008, at 9:59 PM, John D wrote: Hi all, I've googled around for concrete solutions on load balancing Asterisk, and it appears there are several ways to skin this cat -- but not one solution which is all appealing. I have the following requirements, which aren't anything extraordinary: * I need to handle roughly 300 simultaneous phone calls to start * Eventually scale to 1000 simultaneous phone calls * I want to be able to pull out an entire server from the cluster without affecting my application * I'm doing all my trunking over SIP So far I've seen folks mention the use of DUNDi and OpenSER(Now OpenSIPS), but unfortunately the documentation out there is rather sparse and lacks detail for someone who isn't extremely keen with the intricate details of Asterisk or OpenSIPS. Would anyone be able to suggest a good starting point in as far as reading documentation and testing out some solutions? I'd also be up for hiring a consultant to help me get started -- but I believe the proper forum for that is asterisk-biz. (Which I've already posted to). Thank you for your insight on load balancing Asterisk. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - Failover System
I agree that an OpenSER solution on top of Asterisk for a 120 users is massive overkill to say the least. High calls-per-second? Multiple Asterisk servers? Multiple vendors? Advanced LCR? or plans for any of that in the near future? Then I think it makes sense to look at fronting Asterisk with OpenSER for such a small amount of users. Asterisk can do everything you'll need it to do otherwise. - D _ Darren Sessions [EMAIL PROTECTED] http://www.darrensessions.com _ On Oct 1, 2008, at 7:44 PM, Alex Balashov wrote: Jai Rangi wrote: Openser? for 120 user? I would not do that. This would be an extra layer to configure, support, maintain and one more layer to debug if things go wrong. Its like spending a Dollar when you can be done with a quarter. (my 2 cents) All depends on how important those 120 users are. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco + Asterisk
Any particular reason you're using H323 instead of SIP ? _ Darren Sessions [EMAIL PROTECTED] http://www.darrensessions.com _ On Sep 16, 2008, at 12:04 PM, Guilherme Loch Waltrick Góes wrote: I have a Cisco 3845 with a ISDN PRI port connected to my legacy PBX, this router is running IOS 12.4(5) T5. I'm trying to integrate Asterisk with this router through H.323, I tried ooh323 (comes with asterisk-addons) and it works partially, I can make calls from Cisco to Asterisk, but the other way around dosn't work. Does anybody have any hints of what could be wrong ? -- Guilherme Loch Góes Notícias e Fórum sobre VoIP com software livre: http://www.voipexperts.com.br ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP to IAX?
I would suggest using OpenSIPS with Asterisk and bypass IAX all together for this particular application. An OpenSIPS solution will take care of your traveler's NAT issues (and could handle the registrations) while you used Asterisk for voicemail and whatever else. I've personally used this type of general setup in the past with a great deal of success for remote offices and soft-phones on laptops. _ Darren Sessions [EMAIL PROTECTED] http://www.darrensessions.com _ On Sep 9, 2008, at 1:19 PM, Mattias Andersson wrote: Hi all! I am looking for some software that would work as a proxy between a SIP device (SIP phones and ATA boxes) and the Asterisk system running IAX. The reason is that I can only get IAX trow the firewalls, and can't change the settings. One solution I am using are getting several Asterisk system communicate trow IAX bout in this case would I rater have every persons computer act as a proxy for their own phones (Running Widows XP). The reason is that the are using laptops and travel, some are already using softphons and IAX bout some don't like softphons for some reason. If it is not any proxy out their, the will I write o of my own. (Of cause giving it out for free), I think Asterisk for Windows would be overkill. Sorry for my poor English. Regards Mattias Andersson Sweden ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk phone conferencing performance
You shouldn't have any delays at all. Are you using ztdummy for timing? and what kind of load does the box have on it? _ Darren Sessions [EMAIL PROTECTED] http://www.darrensessions.com _ On Sep 9, 2008, at 4:23 PM, George Williams wrote: Hi, I just set up my first Asterisk with MeetMe conference support on my local LAN. It works great, but I think it may need a little tuning - I am getting audio delays of up to 1 second. Should I expect better performance in this area, or is this to be expected? Thanx! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] extensions.conf programming?
A cheaper alternative would be the voip wiki. http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf _ Darren Sessions [EMAIL PROTECTED] http://www.darrensessions.com _ On Sep 4, 2008, at 12:13 PM, Mark Michelson wrote: Ken D'Ambrosio wrote: Hey, all. I haven't really gotten deep into Asterisk since 1.0.x, and I'm afraid I've forgotten a fair bit. One big thing that I've forgotten is the syntax, etc., for extensions.conf. Where do I find that? I'm looking for stuff about commands, syntax for commands, variables, etc. Is there a handy-dandy manpage, webpage, or what-have-you that I'm missing? Thanks! -Ken Your best bet is to read chapters 5 and 6 of Asterisk: The Future of Telephony. Here's a link for the book itself: http://www.oreilly.com/catalog/9780596510480/ Here's a link for the downloadable pdf: http://downloads.oreilly.com/books/9780596510480.pdf Here's a link for the book in html format http://tfot.leifmadsen.com Best of luck to you! Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Tips and Tricks: Dynamic Subroutines inAGI
Impressive work Bradley! I tested it and it worked great, even with my
mandatory 'use strict'.
Thanks,
- Darren
_
Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
_
On Aug 29, 2008, at 5:47 AM, Watkins, Bradley wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Darren Sessions
Sent: Thursday, August 28, 2008 10:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk Tips and Tricks: Dynamic
Subroutines inAGI
...
The hurdle in doing something like this was how to
dynamically execute
a subroutine from the results of the database query which
were dumped
into a variable. The method I used with the subroutine reference
doesn’t allow for arguments to be passed (if anyone finds /
knows a
way to do this, let me know), so I use global variables.
This is a simple example of dynamic subroutine execution
(without the
database query):
use strict;
use warnings;
our $called_number;
our $calling_number;
sub run_me {
$AGI-verbose(”Called Number = “.$called_number, 1);
$AGI-verbose(”Calling Number = “.$calling_number,
1);
}
sub set_variables {
$called_number = “8005551212″;
$calling_number = “300222″;
}
sub dynamic_execute {
my ($sub) = @_;
if (!$sub) {
$AGI-verbose(”No subroutine name passed!!”, 1);
return(-1);
}
my $exec = \{$sub};
return($exec-());
}
set_variables();
dynamic_execute(”run_me”);
If you don't mind disabling strict refs (no strict 'refs';), you
could easily do this.
This would allow you to use something like: $sub($argument1,
$argument2);
The only other way I can think of (though I have not tried it) would
be to populate a hash with subroutine refs and use the string as the
index into it.
Something like this:
#!/usr/bin/perl
use strict;
use warnings;
sub print_ref { print @_; };
my %sub_hash = (print_ref, \print_ref);
sub print_stuff {
my $sub = shift;
my $string = shift;
$sub($string);
}
print_stuff($sub_hash{print_ref}, This is printed.\n);
The first idea uses the symbol table directly, and the second one
essentially is building your own symbol table.
Hope that helps,
- Brad
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[asterisk-users] Asterisk Tips and Tricks: Dynamic Subroutines in AGI
When I set out to develop a basic service provider Perl AGI framework
for Asterisk three or four years ago, I wanted to design something
that would make developing additional Perl AGI apps under this
framework scalable and easy to do. One of the features I wanted to
have in this framework was the ability to do a database dip on a
particular incoming called number to see which service I needed to
execute and then to dynamically execute that subroutine from the
database servers results. I could switch services or point the number
to a canceled operator message by simply doing an update to that
telephone number’s record in the database - instantly re-provisioning
the telephone number.
The hurdle in doing something like this was how to dynamically execute
a subroutine from the results of the database query which were dumped
into a variable. The method I used with the subroutine reference
doesn’t allow for arguments to be passed (if anyone finds / knows a
way to do this, let me know), so I use global variables.
This is a simple example of dynamic subroutine execution (without the
database query):
use strict;
use warnings;
our $called_number;
our $calling_number;
sub run_me {
$AGI-verbose(”Called Number = “.$called_number, 1);
$AGI-verbose(”Calling Number = “.$calling_number, 1);
}
sub set_variables {
$called_number = “8005551212″;
$calling_number = “300222″;
}
sub dynamic_execute {
my ($sub) = @_;
if (!$sub) {
$AGI-verbose(”No subroutine name passed!!”, 1);
return(-1);
}
my $exec = \{$sub};
return($exec-());
}
set_variables();
dynamic_execute(”run_me”);
_
Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
_
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Re: [asterisk-users] Pri to sip interfaces
Are you using an Asterisk PBX? _ Darren Sessions [EMAIL PROTECTED] http://www.darrensessions.com _ On Aug 27, 2008, at 7:06 PM, Tom Moore wrote: Hi guys, What are your suggestions to people who have pbx systems that interface with the world over pri and want to convert them to sip interfaces so that they can use sip trunking? Tom ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pri to sip interfaces
You can use an extremely simple Asterisk config to do the SIP-PRI call conversion that'd be very solid. _ Darren Sessions [EMAIL PROTECTED] http://www.darrensessions.com _ On Aug 27, 2008, at 7:37 PM, Tom Moore wrote: No, these are mainly Samsung pbx systems. I know I can use Asterisk to do this but what be a solid platform to go with that can go in the phone closet? tom From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Darren Sessions Sent: Wednesday, August 27, 2008 9:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Pri to sip interfaces Are you using an Asterisk PBX? _ Darren Sessions [EMAIL PROTECTED] http://www.darrensessions.com _ On Aug 27, 2008, at 7:06 PM, Tom Moore wrote: Hi guys, What are your suggestions to people who have pbx systems that interface with the world over pri and want to convert them to sip interfaces so that they can use sip trunking? Tom ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail has issues with DTMF
If the Linksys unit is forced to a single specific DTMF type, and Asterisk is set specifically to something other than the Linksys, then when the Asterisk server answers the line your DTMF will not be recognized. If your outbound termination vendor supports the Linksys' DTMF settings, then that would also explain why outbound PSTN DTMF is functional. Hope this helps. _ Darren Sessions [EMAIL PROTECTED] http://www.darrensessions.com _ On Aug 23, 2008, at 12:39 AM, Max Alex wrote: Hi everybody, I have linksys phone at my location, i am using asterisk version 1.4.19, I have a issue regarding dtmf mode, i have set the Asterisk DTMF mode to Auto in order to eliminate Asterisk effect on the DTMF transmission. Both Inband and AVT from Linksys worked with PSTN IVR. But, We have the issue why Asterisk Voicemail doesn't work with Linksys set to Inband and Asterisk set to Auto. And what is the reply of asterisk while the dtmf configuration like this? Anyone please help me for this issue, i have searched many pages but i haven't found the exact solution or reason for this? -- Thanks, Max Alex Voip Developer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Semi-OT Satellite?
I've used C-Band, Ku-Band, and DVB satellite internationally with VoIP for years at a previous employer and rarely had any problems was the sat link was up and running. If you do plan on having 'remote offices', you'll want to make sure they all come back to a central earth station (hub and spoke topology) or you'll have virtually insurmountable latency issues (as Femi mentioned). Whatever you do though, don't stick the remote offices with their own internet bandwidth using VPN to connect to the home office for voice, data services as VPNs are extremely problematic over satellite. _ Darren Sessions [EMAIL PROTECTED] http://www.darrensessions.com _ On Aug 23, 2008, at 4:45 PM, Femi wrote: I’ve used VOIP over satellite for years and while it’s not perfect it is sometimes actually better than cellular voice Unless you have a double hop scenario where the traffic makes two satellite hops from one remote to a central hub and then to another satellite remote the latency is actually not noticeable Satellite usually has a latency of 250 – 300 ms and in most cases this does not have a noticeable effect on the conversation Femi From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Tom Moore Sent: 23 August 2008 15:50 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Semi-OT Satellite? Hi, using Asterisk over satellite can be done. Not all satellite providers are created equal and some are better than others. If you are going to do communications between offices that are connected over satellite office to office you may have a problem. My personal choice for satellite connections is the Idirect platform. Tom From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Ken Williams Sent: Saturday, August 23, 2008 9:44 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Semi-OT Satellite? We're entertaining moving our intranet to Hughes satelite for our remote locations. I'm curious if anyone with Asterisk servers has used satellite, and if so, is the latency an issue. My understanding is that you immediately introduce 250ms latency for travel time up and back down, however it is a much more direct connection then offered by traditional land lines. Perhaps someone has some other suggestions? We've started looking into Global Crossing as an alternative to have more control and reliability between all of our remote facilities, maybe this is a better alternative. Our biggest problem is most of our sites are in smaller cities where your bigger connections are more limited. Looking for any suggestions. Thanks, Ken ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] set callerid with plus sign
Just change your dial command and add the plus sign there. _ Darren Sessions [EMAIL PROTECTED] http://www.darrensessions.com _ On Aug 22, 2008, at 1:28 AM, ronald wrote: Hi, Is it possible to assign a plus sign on the callerid(num) ? currently this is what i do CALLERID(num)=+6523450017 but telco is denying calls, coz they said they are seeing bs523450017 instead of +6523450017. i tried putting it inside double quotes CALLERID(num)=+6523450017 telco says the same thing. is this possible? thank you Regards, nhadie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Suddenly the voice become like robot (cutting), like sick man
It's tough to say why a voice would start sounding like a robot. There are so many variables that could effect your Asterisk server. I always go for process of elimination when I have a problem similar to this with call quality. What I would do is install an end point on the same local network / subnet as your asterisk server (either a hard phone or a soft phone like X-Lite by Counterpath). Register the phone locally with your Asterisk server and make some calls or put an echo tester up. If things sound good, you know your Asterisk server is working just fine, and the problems lies somewhere on your network between the Asterisk server and whatever gateway / device. If it sounds awful, and the codecs match, then it's time to start troubleshooting the server. _ Darren Sessions [EMAIL PROTECTED] http://www.darrensessions.com _ On Aug 22, 2008, at 3:14 AM, bilal ghayyad wrote: Dear Darren; You might be right because one day it happened with me and the situation was same like this as following: The status that the ping result is very good for all partied (Asterisk machine, IP Phones on the Internet), and no problem in the processor utilization or RAM or hard disk space. Previously, we changed the DSL router and it worked fine !! But what can I do on the Asterisk level to overcome the problem? I already enabled the jitter on the IAX and SIP, but did not resolved. And I am using the G729 codec and sometimes I use GSM. Any advise for the robot voice with weak battery :) ?! Regards Bilal --- On Thu, 8/21/08, Darren Sessions [EMAIL PROTECTED] wrote: From: Darren Sessions [EMAIL PROTECTED] Subject: Re: [asterisk-users] Suddenly the voice become like robot (cutting), like sick man To: [EMAIL PROTECTED] Date: Thursday, August 21, 2008, 9:47 PM I doubt recompiling is going to help you unless you've got a very unstable system (hard drive going out or something), and then you've got bigger things to worry about then anyways. You should install (if you haven't already) the 'top' program. Top gives you a nice set of system statistics and a list of processes. If you're only having issues on the IP origination side of things, I would start checking your bandwidth and latency on your network. Is the originating end point on the Internet? or local? _ Darren Sessions [EMAIL PROTECTED] http://www.darrensessions.com _ On Aug 21, 2008, at 4:55 PM, bilal ghayyad wrote: Dear Darren; I discovered that calling from the Asterisk to the IP Phone Extension (like calling from mobile to digium and then enter the IP Phone extension, or calling from fxs to the IP Phone extension), it goes very good without any problem. But calling from the same IP Phone to another IP Phone or to any mobile (via fxo port) or to the fxs, it cause the problem (voice become very very bad, like robot with weak battery or sick man). Another way for the problem, if I called from another Asterisk PBX to our Asterisk PBX (that has the problem) and the call was via IAX, and I was need to reach to the IP Phone, then I hear the voice like robot with weak battery. So, the problem appear if the call originator was IP and not TDM. What could be the reason for the problem? No one did any change, I am sure, it suddenly become like this. Any help? Regards Bilal --- On Thu, 8/21/08, Darren Sessions [EMAIL PROTECTED] wrote: From: Darren Sessions [EMAIL PROTECTED] Subject: Re: [asterisk-users] Suddenly the voice become like robot (cutting), like sick man To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non- Commercial Discussion asterisk-users@lists.digium.com Date: Thursday, August 21, 2008, 6:13 PM I'd run top on the server to see if the CPU utilization is going through the roof. If you use AGI, make sure there aren't any orphaned processes consuming resources. If all else fails on the software side of things, I'd restart the server. _ Darren Sessions [EMAIL PROTECTED] http://www.darrensessions.com _ On Aug 21, 2008, at 4:03 PM, bilal ghayyad wrote: Hi All; My asterisk version is 1.4.19.2 and it contains one digium card of 2 fxs and 2 fxo ports, it was working great for more than one month without any problem. Suddenly, any call will be done, then voice becoming like robot (or sick man), it slow and cutting. I restarted the machine, but it is the same !!! I checked the RAM which is 1 GB and I found a lot of space. Any advise what could be the problem? Regards Bilal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http
Re: [asterisk-users] Problem with modem data calls and xorcom astribanks
Not sure what you've heard before, but I have successfully used a modem at 9600 baud (forced via AT commands) through a zaptel card on several occasions. _ Darren Sessions [EMAIL PROTECTED] http://www.darrensessions.com _ On Aug 22, 2008, at 8:14 AM, Greg Woods wrote: I have been told before on this list that a modem through a zaptel card will not work. And mine doesn't, at least not for data calls (it works fine for fax). Apparently the modem requires the full bandwidth of the POTS line, which you do not get through the zaptel card. You might at least check to make sure that echo cancellation is turned off. That can interfere with a data call. --Greg ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] After Dial execution, using DIALEDTIME, ANSWEREDTIME
We recently discussed DeadAGI on the list - I'd check the archives
first.
I just finished doing a write up on DeadAGI and Perl on my website if
you're interested.
DeadAGI *can* be very reliable if done properly.
- Darren
_
[EMAIL PROTECTED]
http://www.darrensessions.com
_
On Aug 21, 2008, at 9:35 AM, selmak se wrote:
Hi,
I noticed that when dial terminates it does not return to the
dialplan, and therefore can not execute any entry after Dial().
Is there any trick to overcome this limitation ?
How I am supposed to handle the returned vales DIALEDTIME,
ANSWEREDTIME if I can not execute anything after Dial()?
I made a workaround with DeadAGI (below) but it is unreliable: if 2
calls end nearly at the same time I do not know to whom belongs the
ANSWEREDTIME value :
exten = h,1,DeadAGI(myagi.agi,0,${DIALEDTIME},${ANSWEREDTIME},00)
Any comments?
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Re: [asterisk-users] Suddenly the voice become like robot (cutting), like sick man
I'd run top on the server to see if the CPU utilization is going through the roof. If you use AGI, make sure there aren't any orphaned processes consuming resources. If all else fails on the software side of things, I'd restart the server. _ Darren Sessions [EMAIL PROTECTED] http://www.darrensessions.com _ On Aug 21, 2008, at 4:03 PM, bilal ghayyad wrote: Hi All; My asterisk version is 1.4.19.2 and it contains one digium card of 2 fxs and 2 fxo ports, it was working great for more than one month without any problem. Suddenly, any call will be done, then voice becoming like robot (or sick man), it slow and cutting. I restarted the machine, but it is the same !!! I checked the RAM which is 1 GB and I found a lot of space. Any advise what could be the problem? Regards Bilal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Perl AGI defunct process
Ruddy,
I've used deadagi for years with perfect success.
If it's a perl agi module, you need to make absolutely sure that
you're using 'use strict' and 'use warnings' in the main agi file -as
well- as any includes. You'll need to test your agi while in console
mode, so any of the perl warning messages that get sent to the console
are visible. You'll want to get rid of any errors and warnings.
In addition, I've setup my agi scripts to execute cleanup functions
when they detect any kind of sig message just for good measure.
$SIG{INT} = 'cleanup';
$SIG{TERM} = 'cleanup';
$SIG{QUIT} = 'cleanup';
$SIG{HUP} = IGNORE;
With this approach, as I said before, I've ran perl agi apps in very
high call volumes at various companies for years without any issues.
Hope this helps.
- Darren
_
[EMAIL PROTECTED]
http://www.darrensessions.com
_
On Aug 19, 2008, at 10:20 PM, Igor A. Goncharovsky wrote:
Hi!
Ruddy Gbaguidi wrote:
I'm using DeadAgi and has set AGISIGHUP to no because I don't want my
script to stop if the user hangs up.
But when it reach the end of the script, the child process should
die.
And I don't see why I only have this trouble with perl agis.
Can you check if your script realy don't get SIGHUP?
Some time ago I have problem with that setting AGISIGHUP to 'no'
have no
effect.
--
Best regards,
Igor A. Goncharovsky
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Re: [asterisk-users] US-based echo test servers?
Another thing you may want to do is try a simple ping test to the far end host. While this may not always be a reliable way to test lag given that the far end maybe just a proxy and your RTP may be terminating to another device, it still should give you a good idea what your lag times are at least on the signaling end of things. You could also do a traceroute to see how many hops your having to jump through as well. You could use a tool like ngrep to actually see the sip signaling and copy out the media gateway from the SDP if you really wanted to, and do a ping on that. I've done extensive work with international voip origination and termination, and typically I haven't had any problems unless it's going over satellite (lag) or there is a problem at the far end (usually pdd or quality issues). If things keep up, I'd even consider running top during a call to see what kind of utilization your local server is at just to make sure something isn't wrong there either. Hope this helps, - Darren _ [EMAIL PROTECTED] http://www.darrensessions.com _ On Aug 18, 2008, at 10:41 AM, Nikhil Nair wrote: Hi, I'm running a small Asterisk server in the UK, just for personal use. I've been experimenting with various VoIP providers for international calls to PSTN numbers, particularly to the US (often California). My results, to date, have been very variable indeed, so much so that I'm considering getting a suitable card and using the PSTN. I have found a VoIP provider with an excellent reputation, and it gives very good quality. However, I seem to get quite a bit of delay at times, enough to make conversation awkward. As the setup at the far end was not completely trivial, I'm not 100% sure the problem was in my connection, but I'd like to test that. Are there any US numbers I can call to get an Asterisk-style echo test? Ideally, a California-based numnber, so I can try to call it from an ordinary PSTN phone here, and compare calling via VoIP, and see if there's an appreciable difference in the delay/quality. I don't anticipate using this for very long, so it doesn't necessarily need to be a free service. Failing that, does anyone have access to a US-based Asterisk server which would allow me to make connections to its echo test? Presumably, if I had this, I could rent a PSTN number from a US-based provider, and point it to the appropriate SIP/IAX address. I expect my total usage would be just a few minutes, though having the facility available for a few weeks would be helpful, to allow me to play around with various options. Again, I'd be willing to pay a modest amount for this. Thanks in advance for any suggestions! Best wishes, Nikhil. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open door automatically...
Set it so when they dial the number, it calls an AGI script that instantly answers and generates a call file and hangs up. That way, you could dial and then hangup, and the system generates a call file that calls the door phone and does whatever it needs to do separate of the initial call. I just posted a Perl based call file generator to the list not to long ago that would easily work for this application. Hope that helps, - Darren _ [EMAIL PROTECTED] http://www.darrensessions.com _ On Aug 13, 2008, at 4:20 PM, Carlos Chavez wrote: I have a new setup that uses a 2N Entrycom door phone that has a switch to open an electric lock. The way this works is that when you are speaking with someone at the door you dial a code and it releases the lock on the door. This part works great. My customer wants to be able to dial a certain number and have the door open automatically without having to wait on the phone. I can simulate this option by using the D option of the Dial command to send DTMF to the door phone once it answers. The only problem is that they do not want to wait until the door phone answers. They just want to dial a number and hangup immediately. How can I do this? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Auto Dialer proof of concept
Here is a simple Perl implementation to generate call files . .
You'll still need something for it to execute after the call files are
generated; either a simple AGI app that streams a file, a Macro, or a
nice dialplan layout.
In any case, you could call something like this very rapidly with
whatever parameters to create as many call files as you felt like, and
Asterisk would start acting on them immediately (if the call files
were generated without wait time).
- Darren
_
[EMAIL PROTECTED]
http://www.darrensessions.com
http://www.linkedin.com/in/dsessions
_
use strict;
use warnings;
sub call_file_name_generator {
my ($len, $str, @chars);
$len = shift;
@chars = ('a'..'z','A'..'Z','0'..'9','_');
foreach (1..$len) {
$str.= $chars[rand @chars];
}
return($str);
}
sub call_file_generator {
use Asterisk::AGI;
my ($channel, $retries, $retry_interval, $wait_time, $application,
$data, $ob_clid) = @_;
if (!$channel || !$retries || !$retry_interval || !$wait_time || !
$application || !$data || !$ob_clid)
$AGI-verbose(Missing data to create call file!!, 1);
return(1);
}
my $ob_file = /var/spool/
asterisk/.call_file_name_generator()..call;
unless(open(CFILE, . $ob_file)) {
$AGI-verbose(Can't open call file for writing!!, 1);
return(1);
}
$file = \#\nChannel: .$channel.\n\nMaxRetries: .$retries.\n;
$file.= RetryTime: .$retry_interval.\nWaitTime: .$wait_time.\n
\n;
$file.= Application: .$application.\nData: .$data.\nCallerid:
.$ob_clid.\n;
printf CFILE $file;
close(CFILE);
system(mv $file /var/spool/asterisk/outgoing);
return(0);
}
On Aug 8, 2008, at 1:48 PM, Bradley Sumrall wrote:
I am a returning Asterisk user.
It has been a few years since I played with it and trying to get a
server up for proof of concept
What is the easiest method of having asterisk dial 5 numbers
simultainiously and deliver a pre recorded message?
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Re: [asterisk-users] Transcoding
I would make absolutely sure you've got your linux distro's version of libgsm installed. I can't really speak to the difference between those two versions of Asterisk without looking at a change-log, but I highly doubt a serious modification to the gsm code took place between sub- versions. Hope this helps, - Darren _ [EMAIL PROTECTED] http://www.darrensessions.com http://www.linkedin.com/in/dsessions _ On Aug 6, 2008, at 7:02 AM, Guilherme Loch Waltrick Góes wrote: I have a server with Asterisk 1.4.21.1 and some prompts recorded in GSM format. I have these same prompts in another server with Asterisk 1.4.18, on this server the prompts sound pretty nice, but on the first one they sound pretty choppy. Was there any changes on the transcoding code between this 2 versions ? Any hints ? Best Regards, -- Guilherme Loch Góes Visite nossa loja virtual: http://www.shopvoip.com.br Notícias e Fórum sobre VoIP com software livre: http://www.asteriskexperts.com.br ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transcoding
I am a **BIG, BIG** fan of OpenSUSE. :) Use yast under 'Software Management' and do a search for 'gsm'. Make sure gsmlib and gsmlib-devel are *both* installed. Then scroll down and make sure that libgsm and libgsm-devel are *both* installed. After that, you'll have to recompile Asterisk. See if that does anything for you. - Darren _ [EMAIL PROTECTED] http://www.darrensessions.com http://www.linkedin.com/in/dsessions _ On Aug 6, 2008, at 8:48 AM, Guilherme Loch Waltrick Góes wrote: I'm using OpenSUSE 10.3, the funny thing is: if the softphone is using GSM the sounds is perfect, if I use Alaw as the softphone CODEC the sounds is pretty bad. The softphone is in the same LAN as the Asterisk server, so I don't think it's a bandwidth issue. Best Regards, On Wed, Aug 6, 2008 at 10:13 AM, Darren Sessions [EMAIL PROTECTED] wrote: I would make absolutely sure you've got your linux distro's version of libgsm installed. I can't really speak to the difference between those two versions of Asterisk without looking at a change-log, but I highly doubt a serious modification to the gsm code took place between sub-versions. Hope this helps, - Darren _ [EMAIL PROTECTED] http://www.darrensessions.com http://www.linkedin.com/in/dsessions _ On Aug 6, 2008, at 7:02 AM, Guilherme Loch Waltrick Góes wrote: I have a server with Asterisk 1.4.21.1 and some prompts recorded in GSM format. I have these same prompts in another server with Asterisk 1.4.18, on this server the prompts sound pretty nice, but on the first one they sound pretty choppy. Was there any changes on the transcoding code between this 2 versions ? Any hints ? Best Regards, -- Guilherme Loch Góes Visite nossa loja virtual: http://www.shopvoip.com.br Notícias e Fórum sobre VoIP com software livre: http://www.asteriskexperts.com.br ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Guilherme Loch Góes Visite nossa loja virtual: http://www.shopvoip.com.br Notícias e Fórum sobre VoIP com software livre: http://www.asteriskexperts.com.br ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transcoding
I have used virtually all versions of Asterisk 1.0+ (literally, either in production or testing) with OpenSUSE 10+ and 11 on AMD and Intel and haven't had any issues with gcc optimizations with regards to audio sounding choppy. This scenario for me has always been the gsm libs. _ [EMAIL PROTECTED] http://www.darrensessions.com http://www.linkedin.com/in/dsessions _ On Aug 6, 2008, at 9:16 AM, Mark Michelson wrote: Guilherme Loch Waltrick Góes wrote: I have a server with Asterisk 1.4.21.1 http://1.4.21.1 and some prompts recorded in GSM format. I have these same prompts in another server with Asterisk 1.4.18, on this server the prompts sound pretty nice, but on the first one they sound pretty choppy. Was there any changes on the transcoding code between this 2 versions ? Any hints ? Best Regards, -- Guilherme Loch Góes Visite nossa loja virtual: http://www.shopvoip.com.br Notícias e Fórum sobre VoIP com software livre: http://www.asteriskexperts.com.br One important difference between the servers may be the compiler used. We have heard reports that using GCC 4.2 or later with optimizations on causes choppy audio when using GSM. Solutions to this include either downgrading your compiler to GCC 4.1 or earlier, or selecting DONT_OPTIMIZE in menuselect under compiler options and then recompiling Asterisk. I also believe that you can set the optimization level for compilation to -O2 in Makefile.rules and have no choppy audio, but I cannot confirm this. Of course, if this server isn't running GCC 4.2, then you can ignore everything I've said so far :) Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 2000+ user Asterisk PBX
I can speak first hand to this having gone through it just a few months ago . . After being spoiled with all the features and standard compliance in Postgres, I was put in a position with a new project to setup a redundant (Master-Slave) database cluster. I immediately jumped to Postgres to do the job (using 8.3). My biggest gripe at the time was that there was really nothing built IN postgres to do the replication as I soon found out. Everything was third party and there were several replication modules suggested to me that seemed stagnant or un-maintained or required an older version of Postgres (bypassing the massive performance increase of the 8.3 release). Of those that I did try that were opensource, all of them seemed fairly complex to get up and running - to say the least. Also having used MySQL extensively, I decided to give it a test run on a separate set of boxes. I'm not exaggerating when I say the replication was up and running in about 10 minutes. While I do appreciate (a lot) how standards compliant Postgres is, MySQL was an absolute clear winner in my book with regards to the replication. Just my two cents . . - Darren _ [EMAIL PROTECTED] http://www.darrensessions.com http://www.linkedin.com/in/dsessions _ On Aug 3, 2008, at 12:26 PM, Tzafrir Cohen wrote: On Sun, Aug 03, 2008 at 08:13:30AM +0100, Grey Man wrote: We use Postgresql which does a good job but the big problem with it is redundancy. Postgresql does not really have an industrial strength replication solution Hmmm... is that really the case? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] app_flite 0.6 released
I've updated the app_flite module to work with the Asterisk 1.6.x code- base in addition to it already working with the 1.4.x, and 1.2.x. (1.0.x support is untested and unsupported). It can be downloaded on my website at: http://www.darrensessions.com/downloads/app_flite-0.6.tar.gz Additional details are in the ChangeLog and README files in the tar ball. As always, if there are any questions or comments, please forward them to me at [EMAIL PROTECTED] Thanks, - Darren _ [EMAIL PROTECTED] http://www.darrensessions.com http://www.linkedin.com/in/dsessions _ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how many quad T1 cards
If you had a dax in front of all your circuits, you could move them from one server to another without physically touching anything. I've done about 300 calls on a dual processor box doing just SIP with an entirely AGI based setup and it held up just fine, but doing TDM, I'd worry about your PCI bus at those call levels. - D _ [EMAIL PROTECTED] http://www.darrensessions.com http://www.linkedin.com/in/dsessions _ On Aug 1, 2008, at 1:07 PM, Al Baker wrote: You mean running , 400 Calls on 1 BOX ? Even if you COULD do it, the gods of TELCO would have you burn in hell for stacking that much critical traffic on ONE Intel, non - high availability box Jerry Geis wrote: Assuming you have a Quad core machine, at least 4 GIG ram, will a machine like this handle 4 Quad T1 cards? is that advisable? What about running AGI's on such a machine. Will the machine handle starting/stopping all those AGI's? Thanks, Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ** app_swift v1.6.2 released for Asterisk 1.6.x code-base **
2008-07-08 - app_swift v1.6.2 released for Asterisk 1.6.x code-base --- Added support for handling multiple dtmf input Added support for input timeout and max input digits (similar to AGI's get_data) Ignores DTMF if no timeout and max digits args are specified Can now wait for DTMF after text-to-speech processing is done if the timeout and max digits args are specified Entire DTMF input placed into channel variable Can be downloaded from http://www.darrensessions.com Thanks, - Darren ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ** app_swift v1.2.2 released for Asterisk 1.2.x code-base **
2008-07-09 - app_swift v1.2.2 released for Asterisk 1.2.x code-base --- Added support for handling multiple dtmf input Added support for input timeout and max input digits (similar to AGI's get_data) Ignores DTMF if no timeout and max digits args are specified Can now wait for DTMF after text-to-speech processing is done if the timeout and max digits args are specified Entire DTMF input placed into channel variable Can be downloaded from http://www.darrensessions.com I promise, this is the last release notice. :) Thanks, - Darren ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ** app_swift v1.4.2 released for Asterisk 1.4.x code-base **
2008-07-08 - app_swift v1.4.2 released for Asterisk 1.4.x code-base --- Added support for handling multiple dtmf input Added support for input timeout and max input digits (similar to AGI's get_data) Ignores DTMF if no timeout and max digits args are specified Can now wait for DTMF after text-to-speech processing is done if the timeout and max digits args are specified Entire DTMF input placed into channel variable Internal cleanup Can be downloaded from http://www.darrensessions.com In addition, an Asterisk 1.6.x code-base version is almost complete. Thanks, - Darren ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] app_swift v1.6.1 released for Asterisk 1.6
Thought I'd let everyone know I've released app_swift v1.6.1 which is entirely based off of Will Orton's work he's placed in the public domain. Works great with Asterisk v1.6.0-beta7.1. In any case, can be downloaded from my site at: http://www.darrensessions.com Go easy on me, this is my first release of anything. Thanks, - Darren ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Need to retrieve Call-ID from dialed SIP channel (w/o CDRs)
Is there a way to retrieve the Call-ID from a call made using the 'Dial' command on a SIP channel without CDRs (i.e. variable) ? Thanks, - Darren ___ --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
[Asterisk-Users] Re: Need to retrieve Call-ID from dialed SIP channel (w/o CDRs)
I obviously didn't explain enough, sorry about that. Let me re-phrase the
problem/question.
Is there a way to retrieve the call-id from a call made using the 'Dial'
command on a SIP channel without CDRs if I've already answered an incoming
call, and the dial (SIP chan) was simply executed on that existing call.
If try and read in the SIPCALLID variable (which I already do on the incoming
call) after the dial, I still get the incoming call's call-id.
Make sense?
btw - Love the documentation . . read it all the time . .
Thanks,
- Darren
Message: 11
Date: Wed, 08 Feb 2006 11:37:13 -0600
From: Kevin P. Fleming [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Need to retrieve Call-ID from dialed SIP
channel (w/o CDRs)
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Darren Sessions wrote:
Is there a way to retrieve the Call-ID from a call made using the 'Dial'
command on a SIP channel without CDRs (i.e. variable) ?
(sometimes I wonder why we write documentation)
doc/README.variables has ${SIPCALLID} documented to be exactly that
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[Asterisk-Users] Re: Need to retrieve Call-ID from dialed number
Exactly. Message: 8 Date: Wed, 08 Feb 2006 13:41:29 -0600 From: Kevin P. Fleming [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Re: Need to retrieve Call-ID from dialed SIP channel (w/o CDRs) To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed Darren Sessions wrote: If try and read in the SIPCALLID variable (which I already do on the incoming call) after the dial, I still get the incoming call's call-id. Your explanation could have been much clearer. Are you saying that you initiate a dial, which succeeds, and then after the call bridge is over you want to know what the Call-ID of the outgoing call was? ___ --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
[Asterisk-Users] Local Channel Call Looping
*** If anyone has a better way of doing this, please post to the list. I
hadn't seen anything on this list or in channel.c/chan_local.c - which
prompted this email ***
I'm not sure how many VoIP providers out there are using Asterisk as a
service platform like we do, but I thought I'd share an experience with
call looping that was racking my brain with the list.
One of the features we offer our customers, is of course, call
forwarding. We take a call in and spit it back out to whatever the call
forward number is set by the customer.
With our particular proxy setup, if a call originates from * to the
proxy it will never loop back to *; this prevents SIP call loops.
In *, for an on-net call forward number, we would use the dial command
to call (if it was registered) the customer's device with SIP via the
proxy, and also dial a local channel to process any of the 'forward to'
customer's features; again, this is for an on-net call.
The problem was that if when we dial the local channel and that customer
had forwarded calls to the first number or calls were setup to forward
from cust1 to cust2 to cust3 to cust1, we were getting an infinite local
channel loop. As you can imagine, the load on * was off the charts.
The solution to the problem finally ended up being to set inherited
channel variables.
First, we'd read/parse the channel variable to determine if the call was
coming in anything other than a local channel. If it was, a variable
with that called number label was immediately set to a value of 1 - i.e.
the first in the chain.
Next, an addition variable with the 'call forward to' number was also
given a value of 1, and then the call was processed. When the new local
channel for the 'forward to' number was spawned, and assuming that call
forwarding was set on that number, the process would repeat with this
inherited variable label scheme.
The catch is that in each iteration at the same time the call forward to
number is being labeled, the system would check that variable for a
value before it tried to assign one. If the variable had a value, it was
safe to assume that it had already been processed in the call chain
somewhere and therefore the system would be looping the call if it
continued.
Here are some sanitized Perl based AGI excerpts that accomplish this:
sub callfwd_loop_check
{
my %v;
($v{callednum},$v{cfnum}) = @_;
$v{num} = $AGI-get_variable($v{cfnum});
if ($v{num})
{
debug( Call Loop Anaylsis for .$v{callednum}. =
LOOPING);
return(1);
}
else
{
debug( Call Loop Anaylsis for .$v{callednum}. = NO
LOOP);
$AGI-exec('Set',__.$v{cfnum}.=.$v{callednum})
}
return;
}
$AGI-exec('Set',__.$callednumber.=1) if ($calltype !~/^Local/);
if (callfwd_loop_check($callednumber,$callfwdtonum))
{
return;
}
$AGI-exec('Dial',Local/+.$callfwdtonum.[EMAIL PROTECTED]SIP/+.$callfwdtonum.[EMAIL PROTECTED]|180);
I hope this all makes sense! :)
Thanks,
- Darren
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[Asterisk-Users] Dial Cmd Outbound CLID Failure (* 1.2.1)
I've been doing AGI now for 2 years, and this problem is making me feel
like I just started. :) I don't have this problem on pre 1.2
installations, so I'm assuming either this is something new, or I've
missed something in the change logs or on wiki.
Scenario:
Customer disables caller id on their IAD. Customer calls in to * where a
perl AGI script reads in RPID info for the customer, and if
privacy=full, set's the callerid variables with RPID info.
Once callerid is set, the AGI script then dials out to a 3rd party,
however, the caller id info is set to 'Unknown'.
If the customer's IAD re-enables callerid, in the same scenario, the
callerid info is passed perfectly through to the 3rd party via *.
It's pretty obvious that * is honoring the privacy=full and/or
recognizing the 'Anonymous' tag in the 'From' field in the sip packet.
Is there a way to disable this behavior so that the callerid can be
forced when the call egresses the * server, regardless of what the
customer's IAD callerid is set to?
I've verified that my RPID parsing subroutine is completely functional
(by verbosing the variables the subroutine sets), and I've verified that
if I just enable the callerid on the IAD, without changing anything
else, that everything works just fine. I completely bypassed this
subroutine in desperation and just set the CLID stuff manually trying to
get it to work.
Any help would be appreciated; thanks in advance,
- Darren
Detailed info below . . .
AGI Excerpts:
Caller ID methods tried:
$AGI-set_variable('CALLERID(name)',\testing\);
$AGI-set_variable('CALLERID(num)',100);
$AGI-set_callerid(\testing\ 100);
$AGI-set_callerid(100);
$AGI-exec('SET',CALLERID 100);
Dial Command: (btw, I've tried using the pipe 'o' as well)
$AGI-exec('Dial',SIP/[EMAIL PROTECTED]|30);
SIP Excerpts (fields modified for protection :) ):
From the IAD to *:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP
192.168.1.100:5065;rport=5065;received=xxx.xxx.xxx.xxx;branch=z9hG4bK-5f1f01ef
From: Anonymous sip:[EMAIL PROTECTED];tag=df69fc0c312eb8bo0
To: sip:[EMAIL PROTECTED]
Remote-Party-ID: TEST
sip:[EMAIL PROTECTED];screen=yes;privacy=full;party=calling
From * out to terminate:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK140bc190;rport
From: Unknown sip:[EMAIL PROTECTED];tag=as335c51b2
To: sip:[EMAIL PROTECTED]
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[Asterisk-Users] GSM Audio Files on Windows w/o Quicktime
Is there a way to play gsm audio files on Windows Media Player ? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] De-Centralized / Distributed Conferencing App
Does such a thing exist? Here is my problem. I've got 300+ people that want to be on a single conference call. Not sure if a single Asterisk server could survive it. I was thinking of putting trunks in between the servers - but quickly realized I'm just giving the audio an extra HOP to traverse - and there is still one box that's going to get slammed. Anyone have any ideas? It would be nice if there was something that allowed you to host x amount of people on one server and x amount on another and created one rtp stream between them with all the combined audio to link the two servers together. An idea anyways. Thanks, - Darren ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hardware (and apple YDL G.729)
Or for that matter, is there a planned G729 binary for Mac OSX ?___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.729 licensing/patent?
Amen On Oct 22, 2004, at 1:26 PM, Kevin Walsh wrote: Kanuri, Seshu (Company IT) [EMAIL PROTECTED] lazily top-posted: Just my $0.02 Cents I propose that an Asterisk development fund be set up to hold all of these $0.02 donations. People who are not quite as cheap could donate a little bit more. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with $AGI-record_file for CVS-HEAD-10/18/04
When I execute the following AGI command in *, if the caller hangs up
during the record - it fails to run the callback sub -BUT- during any
other portion of the call, if the caller hangs up then it gets called
just fine.
Here are some code excerpts:
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
my %input = $AGI-ReadParse();
$AGI-setcallback(\mycallback);
...
$AGI-stream_file(beep);
$rc = $AGI-record_file(tmp_msgs/$sessionId, 'wav', '#*0', 7, 1);
...
sub mycallback {
my ($returncode) = @_;
print STDERR User Hungup ($returncode)\n;
exit($returncode);
}
Like I said - worked before. I'm going to update to the latest CVS and
see if that fixes it.
Any ideas would be appreciated.
Thanks,
- Darren
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Re: [Asterisk-Users] MWI - Sip phones
Noticed it here too. On Oct 21, 2004, at 10:58 AM, Joseph wrote: Using cvs build from CVS-HEAD-10/15/04-06:13:19 it seems the the mwi is randomly not lighting the phone when there is a message. Has any one else noticed this? Sometimes it works, sometimes it seems to *miss* messages. Using mostly cisco 79xx phones. -- respectfully, Joseph === -= ** = ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with $AGI-record_file for CVS-HEAD-10/18/04
Does the -EXACT- same thing if I do a straight print on the record
command.
$rc = print STDOUT RECORD FILE tmp_msgs/$sessionId wav #*0 7;
On Oct 21, 2004, at 11:00 AM, Darren Sessions wrote:
When I execute the following AGI command in *, if the caller hangs up
during the record - it fails to run the callback sub -BUT- during
any other portion of the call, if the caller hangs up then it gets
called just fine.
Here are some code excerpts:
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
my %input = $AGI-ReadParse();
$AGI-setcallback(\mycallback);
...
$AGI-stream_file(beep);
$rc = $AGI-record_file(tmp_msgs/$sessionId, 'wav', '#*0', 7, 1);
...
sub mycallback {
my ($returncode) = @_;
print STDERR User Hungup ($returncode)\n;
exit($returncode);
}
Like I said - worked before. I'm going to update to the latest CVS and
see if that fixes it.
Any ideas would be appreciated.
Thanks,
- Darren
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Re: [Asterisk-Users] SER or not to SER?
We use SER + Asterisk. One heck of a powerful combination. On Oct 21, 2004, at 11:59 AM, Nahuel Alejandro Ramos wrote: Hi everyone, I have some doubt about use or not to use SER. I need a solution using a single linux box that manages, aproximatly 500-1000 registred SIP users, but not more than 50 simultaneouly calls. My plan is to use between 3 to 5 Cisco 26xx PSTN gateway in diferents cities of my country (Argentine) connected through Internet (with public IP). I was searching for SER solutions (and works perfectly) but it does not support Prepaid Billing. So I post a message (on SerUsers maillist) and everybody said me to use Asterisk to use a Prepaid Billing App., so I install Asterisk. I googled, read this maillist (and post some message) and I receive some helpful answers recomending me to install ASTCC, so I install it too and work perfectly too. My questions (if someone could help me) are : 1) What platform (hardware) do I need to support my call flow (500-1000 registers and 50 simultaneouly calls)? 2) Do I need to install SER? 3) If YES, do I need to register my SIP clients on SER and forward all the calls to Asterisk? 3) If NO, do I need to register my SIP clients on Asterisk and forward all the calls to SER? 4) Can I use only Asterisk to REGISTER, AUTH, ACC, ROUTE CALLS, etc, SIP clients? 5) Could I use extension.conf file to route my calls to my diferents Cisco PSTN GW? 6) And how can I use MySQL instead of file? (I have created the DB and tables but I do not know how to make Asterisk use it instead the extension.conf file) 7) I found easy to use only Asterisk, but I have read that it uses to much CPU and memory, is that true? 8) Could anyone some me information about how to configure Asterisk to receive calls through Cisco PSTN GW? 9) THANK YOU VERY VERY MUCH!!! Nahuel Ramos. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SER or not to SER?
Have Asterisk register the same number as your end point/did - i.e. if you've got a carrier sending you a call everytime some calls 18885551212 and then you route that call to a cisco gateway to be terminated on a PBX or whatever, simply have asterisk register that same phone number with SER. Then, when the call comes in, it sends it to the cisco gateway and the asterisk box at the same time - who ever picks up first wins the call. - D On Oct 21, 2004, at 5:42 PM, Iqbal wrote: Hi i am stuck with the same dilemma, as the original poster I have setup ser now (with the helpful pointer from Girish..tks mate) and can do Ip --- Ip calls, and IP ---pstn (via cisco box), all via ser, however I also have asterisk installed, and now am wondering where I use asterisk, it was/is suggested I use it for all pbx functions such as voicemail etc, however I cant seem to see how on a call not answered howto get ser to send to asterisk. I also am looking at the prepaid billing option, and hence the main reason for asterisk, but unless all calls flow via asterisk instead of ser I cant see the point of astcc, and if they do all flow via asterisk, then why put ser infront... tks iqbal On 10/21/2004, Darren Sessions [EMAIL PROTECTED] wrote: We use SER + Asterisk. One heck of a powerful combination. On Oct 21, 2004, at 11:59 AM, Nahuel Alejandro Ramos wrote: Hi everyone, I have some doubt about use or not to use SER. I need a solution using a single linux box that manages, aproximatly 500-1000 registred SIP users, but not more than 50 simultaneouly calls. My plan is to use between 3 to 5 Cisco 26xx PSTN gateway in diferents cities of my country (Argentine) connected through Internet (with public IP). I was searching for SER solutions (and works perfectly) but it does not support Prepaid Billing. So I post a message (on SerUsers maillist) and everybody said me to use Asterisk to use a Prepaid Billing App., so I install Asterisk. I googled, read this maillist (and post some message) and I receive some helpful answers recomending me to install ASTCC, so I install it too and work perfectly too. My questions (if someone could help me) are : 1) What platform (hardware) do I need to support my call flow (500-1000 registers and 50 simultaneouly calls)? 2) Do I need to install SER? 3) If YES, do I need to register my SIP clients on SER and forward all the calls to Asterisk? 3) If NO, do I need to register my SIP clients on Asterisk and forward all the calls to SER? 4) Can I use only Asterisk to REGISTER, AUTH, ACC, ROUTE CALLS, etc, SIP clients? 5) Could I use extension.conf file to route my calls to my diferents Cisco PSTN GW? 6) And how can I use MySQL instead of file? (I have created the DB and tables but I do not know how to make Asterisk use it instead the extension.conf file) 7) I found easy to use only Asterisk, but I have read that it uses to much CPU and memory, is that true? 8) Could anyone some me information about how to configure Asterisk to receive calls through Cisco PSTN GW? 9) THANK YOU VERY VERY MUCH!!! Nahuel Ramos. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Transparent SIP Server
SER most definitely does CDR archiving via MySql database. It's a hellaciously fast and stable proxy - sounds like it'd be a good choice for the core of your network with all the different components. On Oct 19, 2004, at 10:01 AM, Andreas Anderson wrote: Hi Guys, i need to do some kind of CDR for all clients inside my network, but they do not register/use the same sip-server, some of them use iptel, others fwd and various other services. Can i somehow put asterisk in the (control-)path between my clients and the other services (iptable-redirect like with a squid-proxy), so the clients don't have to change their settings and still register with their respective service, but asterisk does a complete CDR on every call? If thats not possible, anyone knows a software that supports this? SER? Regards, Andreas _ Need more speed? Get Xtra JetStream @ http://xtra.co.nz/jetstream ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax over IP doesn't works
Someone should put a bounty on T38. We're using spandsp right now and have had success - but it was an absolute pain to get it to work. On Oct 19, 2004, at 12:38 PM, Steve Underwood wrote: Michael Loftis wrote: Just my $0.02 but seems to me the VoIP community as a whole needs to extend SIP (or IAX?) with a special 'fax data' mode wherein the gateways either act locally as the modem and queue/push bits (not audio data) for the remote end or transparently bridge them through in the case of a passthrough call. IMO faxes need to die, but business still loves them. As I said, just my $0.02. That is such a good idea they did it several years ago. Its called T.38 for H.323 and SIP. IAX doesn't yet have something similar, but its high on the list of things to do. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax over IP doesn't works
We used spandsp on the voip side. Our inbound vendor sets the call up G711 and spandsp answers. It's a bit slow as it seems to only negotiate v29 terbo, but it works. Finding the correct version of libtiff was a pain. One sub version off, and it wouldn't work. Wasn't so much that it was a hard to get it to work, just a lot of tweaking to get it to work. Tedious. On Oct 19, 2004, at 12:50 PM, Steve Underwood wrote: So what changes with T.38? You still need spandsp to interwork with the PSTN. What was so hard about getting spandsp to work? (I'm genuinely interested) Regards, Steve Darren Sessions wrote: Someone should put a bounty on T38. We're using spandsp right now and have had success - but it was an absolute pain to get it to work. On Oct 19, 2004, at 12:38 PM, Steve Underwood wrote: Michael Loftis wrote: Just my $0.02 but seems to me the VoIP community as a whole needs to extend SIP (or IAX?) with a special 'fax data' mode wherein the gateways either act locally as the modem and queue/push bits (not audio data) for the remote end or transparently bridge them through in the case of a passthrough call. IMO faxes need to die, but business still loves them. As I said, just my $0.02. That is such a good idea they did it several years ago. Its called T.38 for H.323 and SIP. IAX doesn't yet have something similar, but its high on the list of things to do. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Audio Files from a Database
Is there a way to stream or at least load into a variable with AGI, gsm or wav files out of a MySql database (contained in MySql as blob fields) directly from asterisk without having to write the files to disk first before you stream them out? I've seen a hack for mpg123 that lets you open MP3's from a database, but nothing for anything else. Seems like a way to make * a little more dynamic. Thanks, - Darren ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wonderful Success with PAP2-NA
The PAP2 is essentially a Sipura. Other than the different skin, a couple cool L.E.Ds, and an updated web interface - they might as well be the same box. Linksys's entire line of VoIP boxes are based on the Sipura technology. Our experience has been that the Sipura rules supreme in features for both the customers and for us, in terms of back-end service provider capability but that Linksys has tighter quality control for included lan cables, and power supplies. Not sure I follow the 4-line bit when you physically only have to lines.. On Oct 19, 2004, at 4:37 PM, Matthew Boehm wrote: Finally got authorized to purchase some PAP2-NA's from Linksys's. Works like a charm with Asterisk. Web configuration has TONS of options and looks nice. Able to put line1 and line2 on seperate asterisk servers. Hooked a 4-line vtech phone up to 2 PAP2-NAs and basically had created a 4 line ATA for $100. -Matthew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wonderful Success with PAP2-NA
Ok.. total brain fart.. sorry.. lol :) On Oct 19, 2004, at 5:55 PM, Matthew Boehm wrote: 2 PAP2NA's with 2 ports each = 4 lines Matthew - Original Message - From: Darren Sessions [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Tuesday, October 19, 2004 4:08 PM Subject: Re: [Asterisk-Users] Wonderful Success with PAP2-NA The PAP2 is essentially a Sipura. Other than the different skin, a couple cool L.E.Ds, and an updated web interface - they might as well be the same box. Linksys's entire line of VoIP boxes are based on the Sipura technology. Our experience has been that the Sipura rules supreme in features for both the customers and for us, in terms of back-end service provider capability but that Linksys has tighter quality control for included lan cables, and power supplies. Not sure I follow the 4-line bit when you physically only have to lines.. On Oct 19, 2004, at 4:37 PM, Matthew Boehm wrote: Finally got authorized to purchase some PAP2-NA's from Linksys's. Works like a charm with Asterisk. Web configuration has TONS of options and looks nice. Able to put line1 and line2 on seperate asterisk servers. Hooked a 4-line vtech phone up to 2 PAP2-NAs and basically had created a 4 line ATA for $100. -Matthew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Getting Call-ID w/o CDR platform
Is there a way to get the Call ID off of a call that runs through * without loading any kind of billing CDR platform? If not, I think it would be a great addition to * if the Call ID was passed as variable (in AGI). Thanks, - Darren ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Getting Call-ID w/o CDR platform
Call-ID as in SIP Call-ID *not* Caller ID. :) Thanks though Danny. On Oct 18, 2004, at 10:02 AM, Danny Froberg wrote: Hi Darren, It is today, check the variables CALLERID, CALLERIDNUM CALLERIDNAME /Danny At 15:58 2004-10-18, you wrote: Is there a way to get the Call ID off of a call that runs through * without loading any kind of billing CDR platform? If not, I think it would be a great addition to * if the Call ID was passed as variable (in AGI). Thanks, - Darren ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Getting Call-ID w/o CDR platform
Steve - it'd be really cool if you knew what you were talking about. There is a distinct difference between a Call ID and Caller ID. Guessing by your need to immediately label everyone a 'newbie' and the fact you don't know what a SIP Call-ID is, I can only speculate as to your technical expertise level. end of response This mailing list is getting out of control with people jumping all over everyone else at a whim. I'm at the point where I'm ready to go to daily digest just so I can weed through the ga-billion or so crap emails. Sheesh.. On Oct 18, 2004, at 10:04 AM, Steven Critchfield wrote: On Mon, 2004-10-18 at 09:58 -0400, Darren Sessions wrote: Is there a way to get the Call ID off of a call that runs through * without loading any kind of billing CDR platform? If not, I think it would be a great addition to * if the Call ID was passed as variable (in AGI). It would be really cool if you could read the documentation. Guessing by your impatience and the fact that you asked what should be obvious questions. You probably haven't waited for the CallerID to be processed before going into either AGI or whatever. Even on PRI lines you need to let a ring happen before going into your AGI. The CallerID will be there if it is available. If you see it in your Master.cdr, it is read and availble, just put a wait in before your AGI app. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Getting Call-ID w/o CDR platform
WORKS PERFECT !!!
THANK YOU !!!
:)
On Oct 18, 2004, at 3:55 PM, Olle E. Johansson wrote:
Darren Sessions wrote:
Call-ID as in SIP Call-ID *not* Caller ID.
In chan_sip2: ${SIPCALLID}
Very useful, indeed.
And looking at the chan_sip source code, I've obviously ported it to
standard Asterisk as well... :-)
/Olle
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Re: [Asterisk-Users] VoIP over 1xRTT
Works with Verizon and G729. I've got a Samsung i700 - works like a champ! If you're in a moving vehicle it can get choppy depending on signal strength - but works well. On Oct 18, 2004, at 5:01 PM, Brian McSpadden wrote: It kind of works...I've done it from my notebook. I wouldn't use it all the time, or for anything important, but it is good for testing and troubleshooting customer's systems while I'm on the road. I have used both X-Lite and Diax, with decent success. I can't say that one worked better than the other in this situation. The sound was a little bit choppy, but it was the variable latency (jitter) that kills you on a connection like that. Brian On Mon, 18 Oct 2004 15:49:38 -0500, Tim Jackson [EMAIL PROTECTED] wrote: Anybody ever tried doing voice over Sprint/Verizon 1xRTT cell service? 10-15KB/sec downloads/uploads with 400-1200MS latency is what I usually see on my service. Tim Jackson Network Engineer Angelina County, Texas (936)639-4827x101 office (936)414-6723 mobile ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quick question regarding daily restart of asterisk
I can tell you from first hand experience that unless you've got +1000 extensions completely configured, it's not a problem in the slightest. After that, you'll start getting to many files open messages (on a vanilla system install) and the server will go temporarily unresponsive (which can be semi-remedied by modifying your OS's max open files - but even then * had problems). On Oct 18, 2004, at 5:11 PM, Matt G wrote: Hi All, I have a quick question regarding restarting (and/or stopping/restarting) asterisk daily -- Should it be done? I've seen conflicting answers, some people have told me that the only reason for asterisk to be stopped/started daily was for mpg123 causing many childs, which has since been fixed using 'no buffer' or 'nb' appended to the line in musiconhold.conf. Others have told me there is no reason whatsoever to restart/stop it, yet there's instructions on how to do it on the wiki, are these just outdated? Is there any other reason why one would want to stop and restart asterisk daily? (or at any other scheduled time?) On a related note, is asterisk -rx restart now the equivalent of asterisk -rx stop now /usr/sbin/safe_asterisk (or whatever command is used to restart it)?. I have a job cronned on a slackware system to restart it daily using -rx restart now and it creates a new PID, and Process Time, but when I run the same thing on Redhat 9 I get an error saying that it exited on sig 13. I'm sure this is just a redhat specific thing as this isn't the only problem I'm running into, but it would be nice to find some answers. Thanks, Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G729 and Sipura.
I've use Sipuras with * using G729 - with no problems. Double check that G729 is turned on in the sipura and your sip.conf is correct - if anything post excerpts from your sip.conf. On Oct 16, 2004, at 6:27 AM, Jefferson Carvalho wrote: Hello All, I purchased yesterday two G729 licenses from Digium to my asterisk box. I used the register utility and i follow the installation procedures as describes the README. I forced my sipuras to use G729a protocol and on my sip.conf too. I get a message that there's no compatible codecs!!! What should i do? - When i use * show codecs , G.729a AUDIO is there! I tried use my X-Pro and i got the same error. I was looking at ITU website for more information if there's a diference between G729 and G729/a/b and compatibility issues between them. Best Regards , -Jefferson Carvalho Teresina-PI-Brazil ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Running Asterisk on Linksys Router
http://sourceforge.net/projects/wifi-box/ On Oct 14, 2004, at 3:43 PM, TC wrote: I run asterisk at my house on a linksys router. I have it sitting in the DMZ of the router so it acts like its outside. Works perfectly fine. is this a wrt54gs ? if so did you get this to compile with the openwrt54 tool chain uclib libaries ? can you share the tweaks you did to that the threads loading of the so modules for asterisk works ? .o---o. Brian Fertig Network Engineer Planet Telecom, Inc. Tampa, FL Office 813.864.3161x107 Office 813.864.3164 Direct 813.817.9961 Cellular 813.881.9762 Fax Web: www.planet-telecom.com email: [EMAIL PROTECTED] --IM's--- MSN: [EMAIL PROTECTED] AIM: ptelebrian Yahoo: ptele_brian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James H. Thompson Sent: Thursday, October 14, 2004 3:17 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Running Asterisk on Linksys Router At Astricon Mark mentioned that somone had Asterisk running on a Linksys Router. Anyone have more information on this? Jim James H. Thompson [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Running Asterisk on Linksys Router
Duh. Simply posting another interesting link. Smart guy. On Oct 14, 2004, at 5:03 PM, Jeremy McNamara wrote: Darren Sessions wrote: http://sourceforge.net/projects/wifi-box/ Yo, smart guy this thread is about running asterisk ON the WRT54GS. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] rfc3389 support in chan_sip?
Why not use an NTP timing source - go stratum 2 or 3. That should be plenty for a stable clock source. On Oct 12, 2004, at 9:52 AM, Eric Wieling wrote: Roy Sigurd Karlsbakk wrote: hi with silence suppression enabled I get these: Oct 12 15:45:55 NOTICE[1104014256]: rtp.c:289 process_rfc3389: RFC3389 support incomplete. Turn off on client if possible is rfc3389 support planned? I don't know if it's planned, but one of the features required to ever support RFC3389 is getting Asterisk to get it's timing for RTP from something other than the incoming RTP stream. I think there's a bounty for that. Check the mailing list archive. The messages are recent enough they may not have been indexed by google yet. eric.vcf___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] rfc3389 support in chan_sip?
We use NTP clock sources for a clock source on many of our physical T1 circuits. We use an outside stratum 1 clock source for our internal server (stratum 2) and because we have our own server, we clock everything else off of it (stratum 3). Maybe I'm not familiar enough with the internals of Asterisk to understand what kind of timing you're after. I assumed you were just after a reference clock. On Oct 12, 2004, at 10:12 AM, Christopher L. Wade wrote: Darren Sessions wrote: Why not use an NTP timing source - go stratum 2 or 3. That should be plenty for a stable clock source. *Timing* is what is needed, not _time_. Two different things. Besides the obvious problems with using a remote network resource as a timing device, I don't think many NTP server admins would enjoy you requesting a _time_ update on the order of 1000+ times a second? RTP not relying on incoming RTP stream is going to require ?hardware? on the machine. My $0.02. -- Christopher L. Wade Unistar-Sparco Computers, Inc. Senior Systems Administratordba Sparco.com Email: [EMAIL PROTECTED] 7089 Ryburn Drive Phone: (901) 872 2272 / (800) 840 8400Millington, TN 38053 Fax: (901) 872 8482 USA ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] System Hang Problem
I am getting some weird behavior and a rash of interesting messages in the log files. If anyone has some ideas, it would be appreciated. Using Asterisk v1.0.1 on Suse Enterprise Linux v8.0. HP DL380 Server. 4GB Ram - Dual 3.2ghz processors. This first entry is when asterisk simply goes unresponsive. We've got a script that automatically polls asterisk (via sip) and restarts it if it does not receive a response. Notice the 9:56 to 10:01 gap. Oct 11 09:53:29 WARNING[6427661]: Failed to write frame Oct 11 09:55:53 WARNING[6445068]: Failed to write frame Oct 11 09:56:10 WARNING[6449163]: Failed to write frame Oct 11 10:01:59 NOTICE[6478861]: Removed default indication country 'us' Oct 11 10:01:59 NOTICE[6150]: Cannot find extension context 'default' Oct 11 10:01:59 NOTICE[6150]: Cannot find extension context 'default' Oct 11 10:01:59 NOTICE[6150]: Cannot find extension context 'default' Oct 11 10:02:01 NOTICE[1024]: parking.conf is deprecated in favor of 'features.c We've started getting allot of these messages in our log files. Unlikely that this is not associated with the first problem. Oct 11 10:02:05 WARNING[6150]: Unable to create RTP session: Too many open files Oct 11 10:02:05 WARNING[6150]: Unable to build sip pvt data for MWI Oct 11 10:02:05 WARNING[6150]: Unable to allocate socket: Too many open files Oct 11 10:02:05 WARNING[6150]: Unable to create RTP session: Too many open files Oct 11 10:02:05 WARNING[6150]: Unable to build sip pvt data for MWI Oct 11 10:02:05 WARNING[6150]: Unable to allocate socket: Too many open files Oct 11 10:02:05 WARNING[6150]: Unable to create RTP session: Too many open files Oct 11 10:02:05 WARNING[6150]: Unable to build sip pvt data for MWI Oct 11 10:02:06 WARNING[6150]: Unable to allocate socket: Too many open files Oct 11 10:02:06 WARNING[6150]: Unable to create RTP session: Too many open files Oct 11 10:02:06 WARNING[6150]: Unable to build sip pvt data for MWI Oct 11 10:02:06 WARNING[6150]: Unable to allocate socket: Too many open files Oct 11 10:02:06 WARNING[6150]: Unable to create RTP session: Too many open files Oct 11 10:02:06 WARNING[6150]: Unable to build sip pvt data for MWI Oct 11 10:02:06 WARNING[6150]: Unable to allocate socket: Too many open files Oct 11 10:02:06 WARNING[6150]: Unable to create RTP session: Too many open files Oct 11 10:02:06 WARNING[6150]: Unable to build sip pvt data for MWI Oct 11 10:02:06 WARNING[6150]: Unable to allocate socket: Too many open files Oct 11 10:02:06 WARNING[6150]: Unable to create RTP session: Too many open files ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] G726 Codec Question
What is the rational for only supporting 32kbps G726 and not 16kbps? Thanks, - Darren ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with Unavailable Message Creation
I've changed the spool directory in asterisk.conf to point to a different directory. Everything works/gets created just fine with the exception of the unavailable messages. When a user tries to create one, I get this on the console (below). I changed the directory to /vm in asterisk.conf. Any help would be appreciated. Thanks, - Darren -- Playing 'beep' (language 'en') Jun 23 18:52:18 WARNING[7175]: file.c:852 ast_writefile: Unable to open file /var/lib/asterisk/sounds/voicemail/default/18037674315/unavail.WAV: No such file or directory -- x=0, open writing: voicemail/default/18037674315/unavail format: wav49, (nil) Jun 23 18:52:18 WARNING[7175]: app_voicemail.c:1463 play_and_record: Error creating writestream 'voicemail/default/18037674315/unavail', format 'wav49' -- Playing 'vm-review' (language 'en') ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk on Apple PPC with YDL
Fyi, Successfully compiled Asterisk on an Apple G4 PPC with Yellow Dog Linux - without any source modifications. Worked fast and smooth. - Darren ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk / SMP / Scalability
I've got Asterisk loading 100,000+ extensions in extensions.conf. This process is taking a little upwards of 10 minutes to complete on each of my dual 3.2Ghz HP DL380 with SuSE Linux Enterprise 8 boxes. Although asterisk creates child processes, it appears that it is only using a single processor to parse extensions.conf. I've turned off Hyper Threading on the servers which has increased the extensions.conf parsing speed, but not by more than a couple minutes. Is this a bug, or simply the way Asterisk works during startup? If it is the way Asterisk works during startup, would it be safe to say that once started - that the child processes would function? Thanks, - Darren ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbie Questions
Ill apologize right away for asking stupid questions. J System Setup: SER = Proxy Asterisk = Voicemail All sip based setup. What Is required to make asterisk NOT- accept inbound calls/signaling from an unknown host? I tried the peers in sip.conf but it still allows unknown hosts to send it calls. Does anyone have a suggestion or maybe some sample configs? Im trying to extensions.conf dynamic. Is there any other alternative to the DynamicDB program to do something like that at this time? Im trying to avoid having to restart * every time we make a change/addition. Im going to be rolling out a fairly large installation of Asterisk. What is the best way to have them all have the same configs/be synchronized? Does anyone have any good tips/advice on SER+Asterisk integration? I appreciate it. - Darren
