[asterisk-users] Goto not jumping to current context
in a simple dialplan like follows:
[firstcontext]
include = secondcontext
include = thirdcontext
include = fourthcontext
[fourthcontext]
_03X.,1,Goto(${EXTEN:2},1)
_X.,1,DoSomething()
_X.,2,Hangup()
the Goto() for exten _03X. seems to start the search for the jump within
firstcontext, thus possibly matching an exten in secondcontext or
thirdcontext first before hitting the matchall in fourthcontext. obviously,
a simple fix would be to change it to Goto(fourthcontext,${EXTEN:2},1).
however, i dont remember Goto working this way. shouldn't a Goto search
within the current context first when the context parameter is ommitted ?
it's asterisk 1.2.14 in FreeBSD 6.1 though.
--
Regards, /\_/\ All dogs go to heaven.
[EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/
+==oOO--(_)--OOo==+
| for a in past present future; do|
| for b in clients employers associates relatives neighbours pets; do |
| echo The opinions here in no way reflect the opinions of my $a $b. |
| done; done |
+=+
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RE: [asterisk-users] AMAFlags alwaysDocumentation (or 3 inastcdrmysql) even after Set(CDR(amaflags)=bill) or SetAMAFlags(bill)
Already using the CDR(userfield), and overloading it with multiple variables will make some DB operations nastier to work with (I'm a little fuzzy and vague on exactly what... something to do with sql joins?). I'm digging deeper into how much pain it really causes us on the DB/App side to see if our work-arounds there are less painful than trying to get AMAFlags to work. It seems that AMAFlags are a really seldom used feature, considering that I'm asking same question that went unanswered several years ago. Thanks all, Scott From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jon Schøpzinsky Sent: Sunday, January 07, 2007 9:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] AMAFlags alwaysDocumentation (or 3 inastcdrmysql) even after Set(CDR(amaflags)=bill) or SetAMAFlags(bill) Hello Why not use the CDR(userfield) field instead. You can set that to any integer of your liking, and use that to identify the type of call. Jon From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Keagy Sent: 8. januar 2007 06:06 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] AMAFlags alwaysDocumentation (or 3 in astcdrmysql) even after Set(CDR(amaflags)=bill) or SetAMAFlags(bill) Is anyone out there using AMAFlags? I'd like to set this field as a marker to distinguish different types of calls in CDRs, but can't seem to make it respond to the documented commands Set(CDR(amaflags)=bill) or SetAMAFlags(bill). I've googled this issue, seen others have had this problem with IAX, with different DB drivers for CDR records, etc. I'm using SIP and LOCAL channels, asterisk-1.4beta2 release (I don't think upgrading to current release will fix this problem, it's been around for years based on trouble reports), both text .csv and mysql astcdr.cdr types. Seems like a problem with basic AMAflags support in CDR. They always show up as DOCUMENTATION in the .csv text file, and they always show up as '3' in mysql. I hurt my brain trying to follow the layers of indirection in the source code for where this is actually set. With verbosity turned on in asterisk console I can see the SetAMAFlags function being run. Any tips, tricks, or pointers in the right direction? Thanks, Scott ___ --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] MFC/R2 problems
Hi all, I have Asterisk 1.2.10, zaptel 1.2.7, spandsp-0.0.3pre22 compiled, and a Sangoma A101, and when I make a call I get this: Jan 8 13:04:06 DEBUG[12252]: chan_unicall.c:2000 unicall_exception: Exception on 19, channel 1 Jan 8 13:04:06 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1101 [1/ 40/Seize /Idle ] Jan 8 13:04:06 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 0 on - [2/ 40/Group I /Idle ] Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 R2 prot. err. [2/ 40/Group I /DNIS ] cause 32769 - T1 timed out Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 0 off - [1/ 1/Idle /Idle ] Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1001 - [1/ 1/Idle /Idle ] Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:2644 handle_uc_event: Unicall/1 event Protocol failure -- Unicall/1 protocol error. Cause 32769 Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Channel echo cancel Jan 8 13:04:11 DEBUG[12252]: chan_unicall.c:955 unicall_disable_ec: disabled echo cancellation on channel 1 Jan 8 13:04:11 WARNING[12250]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1001 [1/ 1/Idle /Idle ] Jan 8 13:04:11 WARNING[12250]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1001 - [1/ 1/Idle /Idle ] -- Hungup 'UniCall/1-1' What does - Unicall/1 protocol error. Cause 32769 mean, and can anyone help me. -- thanks, Yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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
Re: [asterisk-users] SIP/RTP Nat problem, can't solute it.
Still i cannot resolve this issue, please anyone can help me with this? Thanks in advance -- _ Facundo Agustin Barrera -- www.openlabs.com.ar Let the penguins do the work - Buenos Aires - Argentina _ ___ --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
Re: [asterisk-users] Interrupt rates and voip traffic
On Mon, 8 Jan 2007, Rajkumar S wrote: Hi, This is slightly off topic, but here I go any way... VoIP traffic has lot's of smaller packets, and since each packet can generate an interrupt, is there any way to determine the irq rates in a machine, and more importantly to know if I am hitting any of the limits in Linux or to determine how much interrupts per second can my box handle ? There seems to absolutely no information about his particular metric any where.. watch -n1 cat /proc/interrupts That'll give you the basic counters, but to generate graphs, etc. you'll need to use something else. Maybe MRTG. As to what the limits are , I've no idea - a lot will depend on your hardware, cpu, kernel compile options and so on. You may want to know how many packets a second that can pas through an interface - that may be more informative than interrupts though. You can query packets per second via SNMP or simple using ifconfig and looking at the numbers. Gordon ___ --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
Re: [asterisk-users] MusicOnHold Files
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Yuan LIU wrote: One item in my todo list is to make better sound quality whenever end point supports it. Wide-band codec's can already produce better sound than toll. So why do we still need to convert to 8 bit? Should be 16 bit, 8Khz, not 8 bit. However, your point stands, that it should be able to use 44.1Khz instead of 8Khz. The problem is, there is still quite a bit of work to support larger sample rates without simply doing passthrough. If I remember correctly, someone did have a patch and a bit of work done for 1.2. Maybe this made it into 1.4? - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://wap.sineapps.com (Daily Asterisk News for your cellphone) http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFoWeVS6d5vy0jeVcRAqx0AJ41+MMCToCDRvTUrJBipwKoyqdj6QCfeEuG 3INcPOyOmwAbBoNJMNQ8ku4= =Nm2h -END PGP SIGNATURE- ___ --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] Manage 'full' log file
Hi, I need some help on how to manage the full log file. It's getting quite large now and I'd like to clear it. Is there any simple command for this or should I just delete the file (need to be sure this won't affect the system). Also - how do I keep the log file from growing so large? Thanks! Regards, Jan ___ --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
Re: [asterisk-users] Some queries on g729 license.
/var/lib/asterisk/licenses :-) On 1/8/07, Xue Liangliang [EMAIL PROTECTED] wrote: Hi, leo, I will try the following solution that seperate /usr/lib/asterisk/modules in another patition other than drbd, then register the licenses on both server. not sure where the license key acutally lies in? Regards, Liangliang Leo Ann Boon wrote: Xue Liangliang wrote: Hi, actutally it is kind of shareing storage, because we use drbd and vserver technology, the fail over is at vserver level, and vserver is synced through drbd storage. drdb - that's what I suspected. Off the top of my head, the fastest way is to reactivate using the new master's MAC. The proper solution is to only use drdb for data that should be shared like the conf and database. The license key portion should not be on a device that's being mirrored by drdb. Leo ___ --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 ___ --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 ___ --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
Re: [asterisk-users] Manage 'full' log file
We've been using logrotate without any issue... We're using
the below quoted configuration. Notice the invocation of
Asterisk's CLI logger reload command so as to close the
old files and open new ones.
Cheers,
--
Ex Vito
/var/log/asterisk/messages /var/log/asterisk/queue_log
/var/log/asterisk/event_log {
weekly
rotate 52
dateext
compress
delaycompress
nocreate
missingok
sharedscripts
postrotate
/usr/sbin/asterisk -rx logger reload
endscript
}
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SV: [asterisk-users] Manage 'full' log file
Thanks for the quick response!
I read about logrotate at voip-info.org but I didn't quite understand it. I'm
no asterisk/linux expert unfortunately.
First of all. What exactly does happen when I run:
/usr/sbin/asterisk -rx 'logger rotate'
Does it clear the file and create a new one? Can I run this manually without
any interruption in the system?
And what does the script do? I understand it rotates the logs. But does it
delete the old files? Where do I put the script? How do I run it? As you can
see I'm really a newbie on this. Unfortunately the docs for asterisk are often
with the expectation that you know everything... :)
Regards,
Jan
-Ursprungligt meddelande-
Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För Ex Vitorino
Skickat: den 8 januari 2007 13:10
Till: Asterisk Users Mailing List - Non-Commercial Discussion
Ämne: Re: [asterisk-users] Manage 'full' log file
We've been using logrotate without any issue... We're using
the below quoted configuration. Notice the invocation of
Asterisk's CLI logger reload command so as to close the
old files and open new ones.
Cheers,
--
Ex Vito
/var/log/asterisk/messages /var/log/asterisk/queue_log
/var/log/asterisk/event_log {
weekly
rotate 52
dateext
compress
delaycompress
nocreate
missingok
sharedscripts
postrotate
/usr/sbin/asterisk -rx logger reload
endscript
}
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RE: [asterisk-users] Which is GUI to edit Asterisk IVR logic
Yes, trixbox includes more bloat and more problems. Let's clarify something, trixbox includes FOP which is part of FreePBX! -Original Message- From: Steve Sobol [mailto:[EMAIL PROTECTED] Sent: Sunday, January 07, 2007 12:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Which is GUI to edit Asterisk IVR logic On Fri, 5 Jan 2007, Olivier wrote: By Trixbox, do you mean FreePBX (formely AMP) ? Trixbox includes FreePBX but it also includes some other stuff like FOP and an open-source install of SugarCRM. FreePBX is also available as a standalone program. -- Steve Sobol, Professional Geek ** Java/VB/VC/PHP/Perl ** Linux/*BSD/Windows Victorville, California PGP:0xE3AE35ED It's all fun and games until someone starts a bonfire in the living room. ___ --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
Re: [asterisk-users] Which is GUI to edit Asterisk IVR logic
On Mon, Jan 08, 2007 at 04:40:14AM -0800, shadowym wrote: Yes, trixbox includes more bloat and more problems. Let's clarify something, trixbox includes FOP which is part of FreePBX! FOP is actually an independent application. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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
Re: [asterisk-users] jitterbuffer on sip.conf
but keep in mind, that jb for sip (generic jitterbuffer) is implemented differently, than iax, so it works only for SIP-SIP calls, or SIP-ZAP and, curious, eg. for SIP-ZAP call must be activated for (outgoing) ZAP channel :-\ yusuf wrote: [EMAIL PROTECTED] wrote: In iax.conf there is option jitterbuffer how about sip protocol ? Are jitterbuffer can configure in sip.conf ? Thanks, for your share If you upgrade to 1.4, there is a jitterbuffer available now for the SIP channel. ___ --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
Re: SV: [asterisk-users] Manage 'full' log file
[EMAIL PROTECTED] schrieb:
Thanks for the quick response!
I read about logrotate at voip-info.org but I didn't quite understand it. I'm
no asterisk/linux expert unfortunately.
First of all. What exactly does happen when I run:
/usr/sbin/asterisk -rx 'logger rotate'
Does it clear the file and create a new one? Can I run this manually without
any interruption in the system?
And what does the script do? I understand it rotates the logs. But does it
delete the old files? Where do I put the script? How do I run it? As you can see I'm
really a newbie on this. Unfortunately the docs for asterisk are often with the
expectation that you know everything... :)
Regards,
Jan
-Ursprungligt meddelande-
Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För Ex Vitorino
Skickat: den 8 januari 2007 13:10
Till: Asterisk Users Mailing List - Non-Commercial Discussion
Ämne: Re: [asterisk-users] Manage 'full' log file
We've been using logrotate without any issue... We're using
the below quoted configuration. Notice the invocation of
Asterisk's CLI logger reload command so as to close the
old files and open new ones.
Cheers,
--
Ex Vito
/var/log/asterisk/messages /var/log/asterisk/queue_log
/var/log/asterisk/event_log {
weekly
rotate 52
dateext
compress
delaycompress
nocreate
missingok
sharedscripts
postrotate
/usr/sbin/asterisk -rx logger reload
endscript
}
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This script goes into /etc/logrotate.d in a seperate file. It will
compress the log weekly and store it in the same directory the original
log was in.
-- F. Foerster
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Re: [asterisk-users] Manage 'full' log file
Hi Jan, You should use the logrotate in order to delete the log on periodic intervals. This article is meant to do exactly the opposite :) http://astrecipes.net/index.php?n=205 but you get an idea of how to setup log file rotation and how to notify Asterisk that it should open a new file after the log rotation. Hope this helps l. On Mon, 08 Jan 2007 13:00:52 +0100, [EMAIL PROTECTED] wrote: Hi, I need some help on how to manage the full log file. It's getting quite large now and I'd like to clear it. Is there any simple command for this or should I just delete the file (need to be sure this won't affect the system). Also - how do I keep the log file from growing so large? Thanks! Regards, Jan -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --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] IAX call path optimization with more than 3 legs
hi list, after connecting 3 asterisk servers via IAX in a line (+ 1 client at each end), i noticed that call path optimization happens only one time, i.e. only one node/leg in the path can be reduced. Does anybody know if this is the intended behaviour or if it's a bug? Can anyone confirm my observations? It seems that the first node, that sends the TXREQ Message, is optimized. For better understanding: before optimization: C1 A1 A2 A3 C2 after: C1 A1 A2 C2 C1/2: Client with Kiax application A1/2/3: Asterisk Server 1.2.12.1, on Ubuntu Linux ---.: Connection via IAX protocol thanx for reading, Ramon __ Do You Yahoo!? Sie sind Spam leid? Yahoo! Mail verfügt über einen herausragenden Schutz gegen Massenmails. http://mail.yahoo.com ___ --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
Re: [asterisk-users] answer machine detection
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Julian Lyndon-Smith wrote: Is there anyone with any experience of using the AMD app and the settings that worked for them in the UK ? Any help would be appreciated. Hi, I'm using it in New York, and we seem to be having good success (on this particular provider) with: [general] initial_silence = 3700 greeting = 2500 after_greeting_silence = 1200 total_analysis_time = 6000 min_word_length = 100 between_words_silence = 50 maximum_number_of_words = 4 silence_threshold = 860 Disclaimer: just use this as a starting point, go into the console with debug and verbose up, and make sure that for every word you speak, it recognises a word, then try again with cellphones instead of landlines. Remember that you're not going to get 100%, some answer machine messages may have been recorded quietly etc. You'll need to also make sure that the upstream provider doesn't answer the call and then provide ringing (as the stanaphone DID we were testing did). Report back how you go on this, maybe we should start a wiki page with settings in different places. - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://wap.sineapps.com (Daily Asterisk News for your cellphone) http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFoWTGS6d5vy0jeVcRAreVAJ0fyym4B/6vkDuesexxYNlTgt3RBgCcCL7W gTs3lOT7W406rmqhmxxqaqA= =3HcB -END PGP SIGNATURE- ___ --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
SV: [asterisk-users] Manage 'full' log file
Super! Thanks! Now I see how the script works a bit more clearly. :) I still don't understand what happens if I run: /usr/sbin/asterisk -rx 'logger rotate' Can I run the above without having the script? What will the command do? Regards, Jan -Ursprungligt meddelande- Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För Lenz Skickat: den 8 januari 2007 13:13 Till: Asterisk Users Mailing List - Non-Commercial Discussion Ämne: Re: [asterisk-users] Manage 'full' log file Hi Jan, You should use the logrotate in order to delete the log on periodic intervals. This article is meant to do exactly the opposite :) http://astrecipes.net/index.php?n=205 but you get an idea of how to setup log file rotation and how to notify Asterisk that it should open a new file after the log rotation. Hope this helps l. On Mon, 08 Jan 2007 13:00:52 +0100, [EMAIL PROTECTED] wrote: Hi, I need some help on how to manage the full log file. It's getting quite large now and I'd like to clear it. Is there any simple command for this or should I just delete the file (need to be sure this won't affect the system). Also - how do I keep the log file from growing so large? Thanks! Regards, Jan -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --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 ___ --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
Re: [asterisk-users] MFC/R2 problems
Hi Yusuf, how are you? It orders in the list its configurations, so that let us can help. Best Regards Josue 2007/1/8, yusuf [EMAIL PROTECTED]: Hi all, I have Asterisk 1.2.10, zaptel 1.2.7, spandsp-0.0.3pre22 compiled, and a Sangoma A101, and when I make a call I get this: Jan 8 13:04:06 DEBUG[12252]: chan_unicall.c:2000 unicall_exception: Exception on 19, channel 1 Jan 8 13:04:06 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1101 [1/ 40/Seize /Idle ] Jan 8 13:04:06 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 0 on - [2/ 40/Group I /Idle ] Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 R2 prot. err. [2/ 40/Group I /DNIS ] cause 32769 - T1 timed out Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 0 off - [1/ 1/Idle /Idle ] Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1001 - [1/ 1/Idle /Idle ] Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:2644 handle_uc_event: Unicall/1 event Protocol failure -- Unicall/1 protocol error. Cause 32769 Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Channel echo cancel Jan 8 13:04:11 DEBUG[12252]: chan_unicall.c:955 unicall_disable_ec: disabled echo cancellation on channel 1 Jan 8 13:04:11 WARNING[12250]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1001 [1/ 1/Idle /Idle ] Jan 8 13:04:11 WARNING[12250]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1001 - [1/ 1/Idle /Idle ] -- Hungup 'UniCall/1-1' What does - Unicall/1 protocol error. Cause 32769 mean, and can anyone help me. -- thanks, Yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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 ___ --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
Re: SV: [asterisk-users] Manage 'full' log file
On Mon, Jan 08, 2007 at 02:28:19PM +0100, [EMAIL PROTECTED] wrote: Super! Thanks! Now I see how the script works a bit more clearly. :) I still don't understand what happens if I run: /usr/sbin/asterisk -rx 'logger rotate' Can I run the above without having the script? What will the command do? If you use that, make sure you don't have logrotate configured. I generally find logrotate to be more robust and tunable. It is not Asterisk's job to mess with the policy of log rotation on the system. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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
Re: [asterisk-users] answer machine detection
great help. Thanks for that Matt. One thing that is really confusing me at this point: if I want to leave an automated answer machine message, and amd tells me it's a machine, how do I know when to start leaving the message ? Some intros are long (thanks for calling, me and mine are not here right now, please leave a message after the beep) and some are short Leave a message. Is there a way of waiting in the dialplan for a beep or something like that ? Julian. Matt Riddell (NZ) wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Julian Lyndon-Smith wrote: Is there anyone with any experience of using the AMD app and the settings that worked for them in the UK ? Any help would be appreciated. Hi, I'm using it in New York, and we seem to be having good success (on this particular provider) with: [general] initial_silence = 3700 greeting = 2500 after_greeting_silence = 1200 total_analysis_time = 6000 min_word_length = 100 between_words_silence = 50 maximum_number_of_words = 4 silence_threshold = 860 Disclaimer: just use this as a starting point, go into the console with debug and verbose up, and make sure that for every word you speak, it recognises a word, then try again with cellphones instead of landlines. Remember that you're not going to get 100%, some answer machine messages may have been recorded quietly etc. You'll need to also make sure that the upstream provider doesn't answer the call and then provide ringing (as the stanaphone DID we were testing did). Report back how you go on this, maybe we should start a wiki page with settings in different places. - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://wap.sineapps.com (Daily Asterisk News for your cellphone) http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFoWTGS6d5vy0jeVcRAreVAJ0fyym4B/6vkDuesexxYNlTgt3RBgCcCL7W gTs3lOT7W406rmqhmxxqaqA= =3HcB -END PGP SIGNATURE- ___ --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 ___ --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
Re: RE : [asterisk-users] Happy 2007!!!
I am an addict to teasin. Takes one to know on ;) - Original Message - From: Tom Lynn To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, January 01, 2007 7:12 PM Subject: Re: RE : [asterisk-users] Happy 2007!!! Dovid, you're killing me. This after asking if we can't all just be nice to each other. On 1/1/07, Dovid B [EMAIL PROTECTED] wrote: Adam and bill are both wrong. The world revolves around me. Geeez cant we cut the crap (i.e. Happy new year is followed by a response that hey it isnt the new year here yet) If you need the attention find a place where there is a live TV feed (report) and say I am a tool, I need attention Geez.. (As a disclamer don't do it. I just hope you get my point) - Original Message - From: Bill Hackensack To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, January 01, 2007 6:08 PM Subject: Re: RE : [asterisk-users] Happy 2007!!! On 12/31/06, Adam Jacob Muller [EMAIL PROTECTED] wrote: It's still 2006 here -Adam Well, Adam, I guess it is all about you. What does the rest of the world look like as it revolves around you? -- ___ --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 ___ --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 -- ___ --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 ___ --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
Re: [asterisk-users] Some queries on g729 license.
I connect to a PSTN carrier over SIP which requires me to connect with a g729 codec. I'm using them for just robocalling: Asterisk server originates calls which play a prerecorded file. Can I pre-encode those stored files in g729 so they don't need to be encoded for each call? If so, do I need a g729 license for each call, or just a license for the preencoder? If the robocalls accept incoming DTMF, do I need g729 licenses for those calls? On Mon, 2007-01-08 at 04:08 -0700, [EMAIL PROTECTED] wrote: Date: Mon, 08 Jan 2007 13:47:39 +0800 From: Leo Ann Boon [EMAIL PROTECTED] Subject: Re: [asterisk-users] Some queries on g729 license. 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 Xue Liangliang wrote: Hi, all I am a pabx vendor from Singapore. Recently we are going to implement a failover solution for our customers using heartbeat, the asterisk server can failover perfectly, however the g729 codec canot work, because it is binded the mac address, we have bought two set of licenses, can you provide us some workaround for this scenario? It shouldn't be a problem if you're only doing IP takeover and have bound the licenses to each server separately. If you're sharing the storage, then that could pose a problem. Leo DatVoiz Singapore Pte Ltd -- (C) Matthew Rubenstein ___ --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
Re: [asterisk-users] Some queries on g729 license.
Matthew Rubenstein wrote: I connect to a PSTN carrier over SIP which requires me to connect with a g729 codec. I'm using them for just robocalling: Asterisk server originates calls which play a prerecorded file. Can I pre-encode those stored files in g729 so they don't need to be encoded for each call? Yes, if you are using asterisk 1.4 then in the CLI you can type: convert filename-including-path-if-not-in-asterisk-sounds-folder.original extension filename-including-path-if-not-in-asterisk-sounds-folder.g729 so convert recording.ulaw recording.g729 Will make a permanent copy not requireing transcoding again. If you are using asterisk 1.2, there is a tool on the asteriskguru site to transcode the file for you. http://www.asteriskguru.com/tools/audio_conversion.php If so, do I need a g729 license for each call, or just a license for the preencoder? You will need a license for when the file is encoded, after that if it is played back on a g729 call you will not need a license. Asterisk will automatically choose the lowest cost file to playback (which one in natvie format will be). If the robocalls accept incoming DTMF, do I need g729 licenses for those calls? You only need a license when you are transcoding, if you have an incoming call that is g729 and you terminate the call to a device that is configured to use g729 then you will not need a license. If you are recording the call then you will need (possibly 2) llicenses. DTMF signals do not require a license (although the device generating them needs to be configured to use RFC 2833 or Out of Band for DMTF encoding). ___ --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
Re: [asterisk-users] Some queries on g729 license.
Thank you, that is excellent advice. I understand that Intel has a free g729 codec, and that there might be others. Free g729 codecs cheat Digium of some income that helps keep them producing free Asterisk (and hosting lists like this one), but what other reasons (quality, performance, missing features) would make the Digium (or other $) license worth paying for? On Mon, 2007-01-08 at 14:40 +, Thomas Kenyon wrote: Matthew Rubenstein wrote: I connect to a PSTN carrier over SIP which requires me to connect with a g729 codec. I'm using them for just robocalling: Asterisk server originates calls which play a prerecorded file. Can I pre-encode those stored files in g729 so they don't need to be encoded for each call? Yes, if you are using asterisk 1.4 then in the CLI you can type: convert filename-including-path-if-not-in-asterisk-sounds-folder.original extension filename-including-path-if-not-in-asterisk-sounds-folder.g729 so convert recording.ulaw recording.g729 Will make a permanent copy not requireing transcoding again. If you are using asterisk 1.2, there is a tool on the asteriskguru site to transcode the file for you. http://www.asteriskguru.com/tools/audio_conversion.php If so, do I need a g729 license for each call, or just a license for the preencoder? You will need a license for when the file is encoded, after that if it is played back on a g729 call you will not need a license. Asterisk will automatically choose the lowest cost file to playback (which one in natvie format will be). If the robocalls accept incoming DTMF, do I need g729 licenses for those calls? You only need a license when you are transcoding, if you have an incoming call that is g729 and you terminate the call to a device that is configured to use g729 then you will not need a license. If you are recording the call then you will need (possibly 2) llicenses. DTMF signals do not require a license (although the device generating them needs to be configured to use RFC 2833 or Out of Band for DMTF encoding). -- (C) Matthew Rubenstein ___ --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
Re: [asterisk-users] MFC/R2 problems
Hi, if that means I should post my config, here goes: zaptel: span=1,1,3,cas,hdb3,crc4 cas=1-15:1101 cas=17-31:1101 unicall.conf: protocolvariant=id,10,10 protocolend=cpe group=1 channel = 1-15 channel = 17-31 wanpipe1.conf FE_MEDIA= E1 FE_LCODE= HDB3 FE_FRAME= CRC4 FE_LINE = 1 TE_CLOCK= NORMAL TE_REF_CLOCK= 0 TE_HIGHIMPEDANCE= NO LBO = 120OH TE_SIG_MODE = CAS FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 1 TDMV_DCHAN = 16 [w1g1] ACTIVE_CH = ALL TDMV_ECHO_OFF = NO TDMV_HWEC = NO Josué Conti wrote: Hi Yusuf, how are you? It orders in the list its configurations, so that let us can help. Best Regards Josue 2007/1/8, yusuf [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: Hi all, I have Asterisk 1.2.10, zaptel 1.2.7, spandsp-0.0.3pre22 compiled, and a Sangoma A101, and when I make a call I get this: Jan 8 13:04:06 DEBUG[12252]: chan_unicall.c:2000 unicall_exception: Exception on 19, channel 1 Jan 8 13:04:06 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1101 [1/ 40/Seize /Idle ] Jan 8 13:04:06 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 0 on - [2/ 40/Group I /Idle ] Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 R2 prot. err. [2/ 40/Group I /DNIS ] cause 32769 - T1 timed out Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 0 off - [1/ 1/Idle /Idle ] Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1001 - [1/ 1/Idle /Idle ] Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:2644 handle_uc_event: Unicall/1 event Protocol failure -- Unicall/1 protocol error. Cause 32769 Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Channel echo cancel Jan 8 13:04:11 DEBUG[12252]: chan_unicall.c:955 unicall_disable_ec: disabled echo cancellation on channel 1 Jan 8 13:04:11 WARNING[12250]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1001 [1/ 1/Idle /Idle ] Jan 8 13:04:11 WARNING[12250]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1001 - [1/ 1/Idle /Idle ] -- Hungup 'UniCall/1-1' What does - Unicall/1 protocol error. Cause 32769 mean, and can anyone help me. -- -- thanks, Yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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
RE: [asterisk-users] Block some number outgoing from joust oneextention
Hi! Unfortunately did this stop Asterisk to register ny phones and trunk. Did I put tit in the wrong place? //Mattias Hi! Exactly what I needed. It was the 209 part that I did not figure put. Thanks! //Mattias At 03:53 2007-01-05, you wrote: exten = _9070X./209,1,NoOP,SORRY CHARLIE exten = _9070X./209,2,Congestion This would block any call from 209 to 070X as long as 9 was your outside digit. I use the NoOP to help me out with the CLI and debugging :) Hope this helps Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mattias Andersson Sent: Thursday, January 04, 2007 5:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Block some number outgoing from joust oneextention Thanks! I can´t rely figure out how to block for only one extension. Eg. Extension 209 need to be blocked from making calls starting with 070 (eg. 9070). Some clues did I get bout would it men a new form-internal-blocked dialplan? Regards Mattias On 04/01/07, C F mailto:[EMAIL PROTECTED][EMAIL PROTECTED] wrote: The easiest way is thru using contexts. On 1/3/07, Mattias Andersson mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi all! I am shore someone have writing about it bout I cant find it. I have a extension that I need to block from making expansive mobil calls. Everyone else should be aloud to do the calls. I am shore it is possible to be done sens I had a commercial asterisk based PBX that I did that on. However I have switch to Trixbox because I need some custom functions not supported by the commercial product. I would appreciate all help. Regards Mattias Adress: Mattias Andersson Storskiftesvägen 6 S-145 60 Norsborg Mobil: +46-70-799 44 41 Email: mailto:[EMAIL PROTECTED][EMAIL PROTECTED] Skype: eskes1 ___ --Bandwidth and Colocation provided by http://Easynews.comEasynews.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://Easynews.comEasynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- Mattias Andersson Storskiftesvägen 6 145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38 97 Email: mailto:[EMAIL PROTECTED][EMAIL PROTECTED] -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.432 / Virus Database: 268.16.4/615 - Release Date: 1/3/2007 1:34 PM ___ --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 Adress: Mattias Andersson Storskiftesvägen 6 S-145 60 Norsborg Mobil: +46-70-799 44 41 Email: [EMAIL PROTECTED] Skype: eskes1 ___ --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
Re: [asterisk-users] Some queries on g729 license.
Biggest feature: You need a patent license to use the codec. The intel software does not include a patent license. Matthew Rubenstein wrote: Thank you, that is excellent advice. I understand that Intel has a free g729 codec, and that there might be others. Free g729 codecs cheat Digium of some income that helps keep them producing free Asterisk (and hosting lists like this one), but what other reasons (quality, performance, missing features) would make the Digium (or other $) license worth paying for? On Mon, 2007-01-08 at 14:40 +, Thomas Kenyon wrote: Matthew Rubenstein wrote: I connect to a PSTN carrier over SIP which requires me to connect with a g729 codec. I'm using them for just robocalling: Asterisk server originates calls which play a prerecorded file. Can I pre-encode those stored files in g729 so they don't need to be encoded for each call? Yes, if you are using asterisk 1.4 then in the CLI you can type: convert filename-including-path-if-not-in-asterisk-sounds-folder.original extension filename-including-path-if-not-in-asterisk-sounds-folder.g729 so convert recording.ulaw recording.g729 Will make a permanent copy not requireing transcoding again. If you are using asterisk 1.2, there is a tool on the asteriskguru site to transcode the file for you. http://www.asteriskguru.com/tools/audio_conversion.php If so, do I need a g729 license for each call, or just a license for the preencoder? You will need a license for when the file is encoded, after that if it is played back on a g729 call you will not need a license. Asterisk will automatically choose the lowest cost file to playback (which one in natvie format will be). If the robocalls accept incoming DTMF, do I need g729 licenses for those calls? You only need a license when you are transcoding, if you have an incoming call that is g729 and you terminate the call to a device that is configured to use g729 then you will not need a license. If you are recording the call then you will need (possibly 2) llicenses. DTMF signals do not require a license (although the device generating them needs to be configured to use RFC 2833 or Out of Band for DMTF encoding). ___ --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
Re: [asterisk-users] Some queries on g729 license.
What about the free open source G729 Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Matthew Rubenstein wrote: I connect to a PSTN carrier over SIP which requires me to connect with a g729 codec. I'm using them for just robocalling: Asterisk server originates calls which play a prerecorded file. Can I pre-encode those stored files in g729 so they don't need to be encoded for each call? If so, do I need a g729 license for each call, or just a license for the preencoder? If the robocalls accept incoming DTMF, do I need g729 licenses for those calls? On Mon, 2007-01-08 at 04:08 -0700, [EMAIL PROTECTED] wrote: Date: Mon, 08 Jan 2007 13:47:39 +0800 From: Leo Ann Boon [EMAIL PROTECTED] Subject: Re: [asterisk-users] Some queries on g729 license. 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 Xue Liangliang wrote: Hi, all I am a pabx vendor from Singapore. Recently we are going to implement a failover solution for our customers using heartbeat, the asterisk server can failover perfectly, however the g729 codec canot work, because it is binded the mac address, we have bought two set of licenses, can you provide us some workaround for this scenario? It shouldn't be a problem if you're only doing IP takeover and have bound the licenses to each server separately. If you're sharing the storage, then that could pose a problem. Leo DatVoiz Singapore Pte Ltd ___ --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] Adding 4000 Lines to asteriskdb via asterisk -rx ?
Hi there, I want to add 4000 Callerids and Callernames to my asterisk-db. (/var/lib/asterisk/astdb) I do not want an external database or an sql-database because I do not want asterisk to depend on external processes. However, when I do 4000 database put number name via a shellscript and asterisk -rx I only have 600 entries later in my asterisk- database. The asterisk sockets seems not to be designed for bulk updates to the asterisk-db. I also don't want to add 4000 sleep 1 to my shell-script. Does anybody have an Idea how to add these lines to asterisk ? I managed to Build a Perl DB_File Module for db1.8.5 but I do not have the know how how to use DB_file and db1 databases. Are there some external utilities to lock and update the asteriskdb ? Is there a better way ? Thanks Christoph -- Two hours of trial and error can save ten minutes of manual reading. GATWORKS GmbH [EMAIL PROTECTED] Internetloesungen vom Feinsten Fon. +49 2166 9149-32 Fax. +49 2166 9149-10 ___ --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
Re: [asterisk-users] Some queries on g729 license.
Al Bochter wrote: What about the free open source G729 To use a g729 codec you must pay a license fee to the patent holder. It is immaterial as to whether the implementation is open/closed source. ___ --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
Re: SV: [asterisk-users] Manage 'full' log file
You know that if you rename an open Unix file, it will stay open - i.e. if you rename the logfile full to full.1, Asterisk will continue writing to full.1 thinking it was full. The logger rotate command forces all log files to be closed and reopened with their canonical names, so your file is actually rotated. Hope this helps l. In data Mon, 08 Jan 2007 14:28:19 +0100, [EMAIL PROTECTED] ha scritto: Super! Thanks! Now I see how the script works a bit more clearly. :) I still don't understand what happens if I run: /usr/sbin/asterisk -rx 'logger rotate' Can I run the above without having the script? What will the command do? Regards, Jan -Ursprungligt meddelande- Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För Lenz Skickat: den 8 januari 2007 13:13 Till: Asterisk Users Mailing List - Non-Commercial Discussion Ämne: Re: [asterisk-users] Manage 'full' log file Hi Jan, You should use the logrotate in order to delete the log on periodic intervals. This article is meant to do exactly the opposite :) http://astrecipes.net/index.php?n=205 but you get an idea of how to setup log file rotation and how to notify Asterisk that it should open a new file after the log rotation. Hope this helps l. On Mon, 08 Jan 2007 13:00:52 +0100, [EMAIL PROTECTED] wrote: Hi, I need some help on how to manage the full log file. It's getting quite large now and I'd like to clear it. Is there any simple command for this or should I just delete the file (need to be sure this won't affect the system). Also - how do I keep the log file from growing so large? Thanks! Regards, Jan -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --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 ___ --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 -- Home of QueueMetrics - http://queuemetrics.loway.it ___ --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] ARA extensions ordering
Hello List, I am curious how the ordering of the extensions are determined for an ARA dial-plan. For example, if I have these: _9X. _9011. Which is selected first? Any number dialed starting with 9011 is matched by either rule here and I don't remember seeing any ORDER BY clauses when I had debugged the ARA queries. I'm sure I just missed some critical documentation here. Thoughts? THanks, - Jesse ___ --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: ARA extensions ordering
I am sot sure, but you can use the following to make sure: _9. _9011. -- -- Steven http://www.glimasoutheast.org Jesse Peterson [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Hello List, I am curious how the ordering of the extensions are determined for an ARA dial-plan. For example, if I have these: _9X. _9011. Which is selected first? Any number dialed starting with 9011 is matched by either rule here and I don't remember seeing any ORDER BY clauses when I had debugged the ARA queries. I'm sure I just missed some critical documentation here. Thoughts? THanks, - Jesse ___ --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 ___ --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] Realtime Voicemail Table Column Name Question
Hi All, In the realtime voicemail table the column 'customer_id' is used, for my purpose, to specify the customers accountcode. The column name 'accountcode' is used in the iax and sip tables. To keep this consistent throughout the tables, is there any reason I should NOT switch the column name 'customer_id' to 'accountcode' in the voicemail table? Does Asterisk read from the 'customer_id' column for anything? Is the name particular and need to remain 'customer_id' for any reason? Thanks. JR -- JR Richardson Engineering for the Masses ___ --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: Adding 4000 Lines to asteriskdb via asterisk -rx ?
How much would all of that data slow down asterisk? Is astdb made to handle that much data? -- -- Steven http://www.glimasoutheast.org Christoph Adomeit [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Hi there, I want to add 4000 Callerids and Callernames to my asterisk-db. (/var/lib/asterisk/astdb) I do not want an external database or an sql-database because I do not want asterisk to depend on external processes. However, when I do 4000 database put number name via a shellscript and asterisk -rx I only have 600 entries later in my asterisk- database. The asterisk sockets seems not to be designed for bulk updates to the asterisk-db. I also don't want to add 4000 sleep 1 to my shell-script. Does anybody have an Idea how to add these lines to asterisk ? I managed to Build a Perl DB_File Module for db1.8.5 but I do not have the know how how to use DB_file and db1 databases. Are there some external utilities to lock and update the asteriskdb ? Is there a better way ? Thanks Christoph -- Two hours of trial and error can save ten minutes of manual reading. GATWORKS GmbH [EMAIL PROTECTED] Internetloesungen vom Feinsten Fon. +49 2166 9149-32 Fax. +49 2166 9149-10 ___ --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 ___ --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
Re: [asterisk-users] Some queries on g729 license.
As far as I know, the g729 patent requires buying a license to operate any implementation of it, whether Digium's, Intel's, or any other. Digium is set up to collect royalties (perhaps at a favorable rate) as part of their license from the patent holder. I don't know about Intel or any other. Or what the mechanics are for enforcing the patent on someone who operates a codec without a license. On Mon, 2007-01-08 at 10:51 -0500, Al Bochter wrote: What about the free open source G729 Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Matthew Rubenstein wrote: I connect to a PSTN carrier over SIP which requires me to connect with a g729 codec. I'm using them for just robocalling: Asterisk server originates calls which play a prerecorded file. Can I pre-encode those stored files in g729 so they don't need to be encoded for each call? If so, do I need a g729 license for each call, or just a license for the preencoder? If the robocalls accept incoming DTMF, do I need g729 licenses for those calls? On Mon, 2007-01-08 at 04:08 -0700, [EMAIL PROTECTED] wrote: Date: Mon, 08 Jan 2007 13:47:39 +0800 From: Leo Ann Boon [EMAIL PROTECTED] Subject: Re: [asterisk-users] Some queries on g729 license. 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 Xue Liangliang wrote: Hi, all I am a pabx vendor from Singapore. Recently we are going to implement a failover solution for our customers using heartbeat, the asterisk server can failover perfectly, however the g729 codec canot work, because it is binded the mac address, we have bought two set of licenses, can you provide us some workaround for this scenario? It shouldn't be a problem if you're only doing IP takeover and have bound the licenses to each server separately. If you're sharing the storage, then that could pose a problem. Leo DatVoiz Singapore Pte Ltd -- (C) Matthew Rubenstein ___ --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] G729 license counting
Hello, How many licenses to buy?? : From what we understood from digium website, we must buy as many licenses as the number of maximum simultaneous calls using G729 Codec we wish to make. For example, If we want to be able to make a maximum of 10 simultaneous calls using G729 Codec, we must buy 10 licenses. Is it right? Thanks you ___ --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
Re: [asterisk-users] Problems with park
Leo Ann Boon wrote: Are you doing a blind transfer or attended transfer? I'm assuming you're using the phone's transfer button. You may need to press transfer a second time to complete the transfer. Attended. I have decided my problem is with transfer, not specifically parking. I just noticed I'm having a similar problem transferring on another system I recently upgraded to 1.4. When I try to complete the transfer, I get Transfer Failed on my LCD, and the following on my console: [Jan 8 10:32:31] WARNING[30406]: chan_sip.c:12317 handle_response: Notify answer on an owned channel? I have set the transfer context (did I do that right? internal context contains the extension I'm transferring to) exten = 101,1,Set(*__TRANSFER_CONTEXT*=internal) I have even caused Asterisk to crash a couple of times. I'm beginning to wonder if I'm hitting a bug in 1.4. I've noticed a few other people have got the notify answer... message. I'm going to try downgrading to 1.2.14 to see if it works there. In the meantime, and additional wisdom would still be appreciated. ttyl srw ___ --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
Re: [asterisk-users] Re: ARA extensions ordering
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf+sorting Consult the wiki! -- George Pajari, netVOICE communications604 484 VOIP (484 8647 x102) Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) Hosted IP PBX Services for SOHO Small Businesses - www.ip-centrex.ca VoIP Service, Equipment, Systems, and Consulting - www.netvoice.ca ___ --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
Re: [asterisk-users] G729 license counting
Yes Zoa Michel wrote: Hello, How many licenses to buy?? : From what we understood from digium website, we must buy as many licenses as the number of maximum simultaneous calls using G729 Codec we wish to make. For example, If we want to be able to make a maximum of 10 simultaneous calls using G729 Codec, we must buy 10 licenses. Is it right? Thanks you ___ --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 ___ --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
Re: [asterisk-users] Some queries on g729 license.
On Mon, Jan 08, 2007 at 10:51:03AM -0500, Al Bochter wrote: What about the free open source G729 There's no such thing ... g.729 (as per the ITU specification) is patent encumbered. Anyone USING the codec has to pay a license to the patent holders. Digium have negotiated a bulk-buying agreement and can sub-license (or relicense - however they've worded their agreement) the codec to end users. The same is true for several other codecs like AMR etc. even though there are open source implementations of them. MP3 is also patent encumbered, but since so many people were using it they changed the licensing so that freeware players could continue giving away the implementation. Any commercial software (or hardware) has to pay license fees (for encoding or decoding). Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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
Re: [asterisk-users] Problems with park
Scott Walde wrote: I have even caused Asterisk to crash a couple of times. I'm beginning to wonder if I'm hitting a bug in 1.4. I've noticed a few other people have got the notify answer... message. I'm going to try downgrading to 1.2.14 to see if it works there. In the meantime, and additional wisdom would still be appreciated. Well, it works in 1.2.14. I guess I'll have to decide how badly I want MWI lights. ttyl srw ___ --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
Re: [asterisk-users] Some queries on g729 license.
What about the free open source G729 There's no such thing ... g.729 (as per the ITU specification) is patent encumbered. Anyone USING the codec has to pay a license to the patent holders. I believe (this may have changed) that ANY patented technology can be used for free educationally. The idea is that people can study and play with the technology for no charge. I'm not sure if this means that a University can use this in their phone system without paying the patent fees, though. Now, certainly there can be open source versions of the G.729 codec. They can even be free in the sense that the author is not charging. But the author can't waive the patent rights. Intel has a freely downloadable codec for educational use, but they have a long legalese document which explains the patent obligations. If you are using G.729 commercially, there is no question you have a legal obligation to pay the patent holder for his rights. J ___ --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
RE: [asterisk-users] answer machine detection
One thing that is really confusing me at this point: if I want to
leave
an automated answer machine message, and amd tells me it's a machine,
how do I know when to start leaving the message ? Some intros are
long
(thanks for calling, me and mine are not here right now, please leave
a
message after the beep) and some are short Leave a message.
Is there a way of waiting in the dialplan for a beep or something like
that ?
Excellent question. I've been experimenting with the 'WaitForSilence'
app. I've not tried to detect a beep since answering machines and
voicemail systems will not be uniform in their beep sounds.
I've used this with reasonably good success:
[lmtc]
; if detect ans machine, come here and leave a msg to call back
exten = s,1,Answer
exten = s,n,Wait(5)
exten = s,n,WaitForSilence(1000,2)
exten = s,n,Playback(Not-right-party-live-Eng)
exten = s,n,Wait(.3)
exten = s,n,SayDigits(${dnum}) ; Supplied by Originate action
exten = s,n,Wait(1)
exten = s,n,Playback(Not-right-party-live-Eng)
exten = s,n,Wait(.3)
exten = s,n,SayDigits(${dnum}) ; Supplied by Originate action
exten = s,n,Wait(1)
exten = s,n,AppendCDRUserField(${cdrdelim}Y)
exten = s,n,Hangup
As soon as I detect AMD, I goto lmtc,s,1. I wait 5 seconds, then do
wait for silence. I've experimented with various settings, and I
settled on wait for two occurrences of 1000ms of silence. This is a
reasonable balance between having a two second pause at the very
beginning of the message that I leave and accidentally starting my
message playback too early because of silence detected during the target
machine's outbound message. Sometimes you have a message like, High
this is so-and-so. pause Please leave me a message. That pause can
sometimes trip up your WaitForSilence app if you don't wait long enough
for silence.
In my case, I'm leaving a message that says, Please call us at phone
number and provide reference number dnum. I repeat the message just
in case I started playing it too soon the first time. Thus far I've had
pretty good success. YMMV, so tinker with the WaitForSilence settings.
If you're okay with a two second pause at the beginning of the message
that you leave on the target answering machine then these settings will
probably work for you.
Let us know how it goes.
-MC
___
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RE: [asterisk-users] G729 license counting
That's not correct. You need one G729 license for each transcoding instance. If you have two SIP channels and both are G729, then no license is required. If you have two SIP channels, and one is G729 and the other is ulaw, then a license is required. Doug. -Original Message- From: Zoa [mailto:[EMAIL PROTECTED] Sent: Monday, January 08, 2007 10:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] G729 license counting Yes Zoa Michel wrote: Hello, How many licenses to buy?? : From what we understood from digium website, we must buy as many licenses as the number of maximum simultaneous calls using G729 Codec we wish to make. For example, If we want to be able to make a maximum of 10 simultaneous calls using G729 Codec, we must buy 10 licenses. Is it right? Thanks you ___ --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 ___ --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 ___ --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] Strange error
Someone know why my asterisk gives me the following msgs? Thank you - Got SIP response 603 Declined (no dialog) back from X.X.X.Xhttp://82.51.224.34/ -- Got SIP response 603 Declined (no dialog) back from X.X.X.Yhttp://82.51.224.34/http://82.104.4.192/http://82.104.4.192/ -- Got SIP response 603 Declined (no dialog) back from X.X.X.Zhttp://82.51.224.34/http://82.51.224.34/ -- Got SIP response 603 Declined (no dialog) back from X.X.X.Xhttp://82.51.224.34/http://82.104.4.192/ -- Got SIP response 603 Declined (no dialog) back from X.X.X.Xhttp://82.51.224.34/http://82.51.224.34/ ___ --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] OT:spa942 provisioning
Hello! Sorry for the OT-thread, but i don't know where else too ask... Has anyone done http-provisioning of a Linksys SPA942 with client side ssl-authentication? Where do i get the CA from? I'm aware of the Sipura mass deployment howto on voip-info.org, but it doesn't cover the authentification part. Thanks Christian ___ --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
Re: [asterisk-users] Some queries on g729 license.
First point to tackle in any case involving patent, copyright or trademark infringement is whether or not the infringing party would have been qualified to buy any usage rights at all. In a case where you license the Intel source(read the terms, it's not really that free), you would be applying for a license under some plan that includes certain minimum payments. Even if you wrote new source from scratch you would be in the same boat. Last time I looked at the plans, I didn't see anything with low minimums. So even if you wrote code from scratch and never used it on more than 6 channels, you might have done something that normally requires a large upfront payment. Use $10k as an example. In such a case owner of the patent might have an attorney initiate contact. If you are willing to communicate they might allow you to pay the minimum and be licensed. If you can't do that, they might offer a settlement where you stop using the codec and pay them some lesser amount. If the patent holder can easily prove the violation you might as well try to deal with them and get things settled fast. If you sell or give away the codec it is easier for them to dig up proof. If you have unhappy employees that might be the way they hear about the violation in the first place. Important consideration: Bankruptcy law generally excludes debts created by things like malicious or criminal acts. Matthew Rubenstein wrote: As far as I know, the g729 patent requires buying a license to operate any implementation of it, whether Digium's, Intel's, or any other. Digium is set up to collect royalties (perhaps at a favorable rate) as part of their license from the patent holder. I don't know about Intel or any other. Or what the mechanics are for enforcing the patent on someone who operates a codec without a license. On Mon, 2007-01-08 at 10:51 -0500, Al Bochter wrote: What about the free open source G729 Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Matthew Rubenstein wrote: I connect to a PSTN carrier over SIP which requires me to connect with a g729 codec. I'm using them for just robocalling: Asterisk server originates calls which play a prerecorded file. Can I pre-encode those stored files in g729 so they don't need to be encoded for each call? If so, do I need a g729 license for each call, or just a license for the preencoder? If the robocalls accept incoming DTMF, do I need g729 licenses for those calls? On Mon, 2007-01-08 at 04:08 -0700, [EMAIL PROTECTED] wrote: Date: Mon, 08 Jan 2007 13:47:39 +0800 From: Leo Ann Boon [EMAIL PROTECTED] Subject: Re: [asterisk-users] Some queries on g729 license. 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 Xue Liangliang wrote: Hi, all I am a pabx vendor from Singapore. Recently we are going to implement a failover solution for our customers using heartbeat, the asterisk server can failover perfectly, however the g729 codec canot work, because it is binded the mac address, we have bought two set of licenses, can you provide us some workaround for this scenario? It shouldn't be a problem if you're only doing IP takeover and have bound the licenses to each server separately. If you're sharing the storage, then that could pose a problem. Leo DatVoiz Singapore Pte Ltd ___ --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] Looking for toll free in Italy
Hi, I am looking for tollfree number in italy. Anybody providing that? Charge per minute? It will connect to my asterisk pbx box. Thanks CM __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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
RE: [asterisk-users] Re: [Announce] Web-MeetMe 3.0.0 released
Buki wrote: Sorry I forgot to change the subject line in my last posting! I have been using the Web Meet-Me 2.1.0 on Asterisk 1.2.12.1 for many months now and I am a big fan and I have been very happy with it. I'm glad it's working well for you, positive feedback is always welcome. I want to try the v3.0.0 but I would like to know if there are specific steps I need to carry out to upgrade to the v3.0.0 on my current Asterisk 1.2.X? There are a couple answers here. First is that version 3.0.0 is NOT compatible with Asterisk 1.2.X, so there is no way to test or use it in your installation. There is a plan to release version 2.2.0 soon that has the features and bug fixes from version 3.0.0 that do not have a dependancy on Asterisk's version. The second answer is about the upgrade it self. Since the package is mostly php pages, there is not an 'upgrade'. Just rename the directory where Web-MeetMe is installed and extract the latest package. With the 3.0.0 and 2.2.0 releases we have further seperated the configuration settings from the actual code, so future upgrades should be able to re-use the ./lib/defines.php. With the 3.0.0 and 2.2.0 release it will be easiest to just edit the new defines.php to match your settings. Lastly you may need to add a couple columns to your database to take advantage of the improved recurring conference support. Refer to the sample tables in the ./cbmysql directory for details. Dan current document root and extract the package to ___ --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] MixMonitor write issue
Greetings, I am using MixMonitor to record my outgoing calls. It seems that MixMonitor will not write to a directory if it doesn't exist (ie - it doesn't create a new directory if needed). I have checked to ensure permissions are properly set, and if I manually create the directory, MixMonitor behaves normally. Rather than send several 'mkdir' commands each time I want to record a file, I was hoping someone knew an easier way to do this. It strikes me odd that directories are created when I record queue calls with 'monitor-join = yes', but can't do the same for outgoing calls. Any help would be much appreciated. Regards, Jay ___ --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
Re: [asterisk-users] Some queries on g729 license.
Jerry wrote: What about the free open source G729 There's no such thing ... g.729 (as per the ITU specification) is patent encumbered. Anyone USING the codec has to pay a license to the patent holders. I believe (this may have changed) that ANY patented technology can be used for free educationally. The idea is that people can study and play with the technology for no charge. I'm not sure if this means that a University can use this in their phone system without paying the patent fees, though. Now, certainly there can be open source versions of the G.729 codec. They can even be free in the sense that the author is not charging. But the author can't waive the patent rights. Intel has a freely downloadable codec for educational use, but they have a long legalese document which explains the patent obligations. If you are using G.729 commercially, there is no question you have a legal obligation to pay the patent holder for his rights. Whether it's a university or a megacorporation studying the technology, they have to be very careful. Suppose we are working on automotive fuel economy or emissions improvement. If we buy a new or used car we are reasonably sure that a multitude of patents involved are being legally used. If we build a cadillac clone for the research, I would be worried. Another factor to consider in some cases is when we sign a sales contract that includes things like no reverse engineering. It might be hard to prove that we did not reverse engineer the product in order to develop a patentable improvement. ___ --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
Re: [asterisk-users] Some queries on g729 license.
Mike, So tell me what this FREE open source G729 is I am told that you can use these Codecs with your Asterisk ! http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ You can do it Freely !! Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Mike wrote: Al Bochter wrote: What about the free open source G729 To use a g729 codec you must pay a license fee to the patent holder. It is immaterial as to whether the implementation is open/closed source. ___ --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 Inbound (clean). Database: 0701-4, 01/08/2007 - 1/8/2007 2:46:30 PM ___ --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
Re: [asterisk-users] Some queries on g729 license.
On Mon, Jan 08, 2007 at 02:53:39PM -0500, Al Bochter wrote: So tell me what this FREE open source G729 is I am told that you can use these Codecs with your Asterisk ! [1]http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ You can do it Freely !! No, Ready Technology have packaged the codecs based on Intel's IPP code. The codecs link against Intel's IPP libraries. The code here is a diff and other material to compile the codecs once you've downloaded the IPP libraries. It will then produce a binary. To download Intel's libraries you need to agree to their licensing terms. To utilise the codecs you still need to pay a royalty fee to Sipro (as is clearly stated on the site). There are some pre-built binaries held on servers were the patents don't apply, however utilising those binaries on a system in a country where they do apply means you have to pay royalties. If you look it's the patches which are distributed under GPL, not the actual code itself. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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
Re: [asterisk-users] G729 license counting
You need a license when ever you transcode the audio From any codec to G729. or G729 to any codec you will need a license for each instance. If you call into your system from a provider that uses G729 you don't need a license If you check your voicemail that is saved on your system in GSM format then you need a license to transcode the file from GSM to G729 Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Douglas Garstang wrote: That's not correct. You need one G729 license for each transcoding instance. If you have two SIP channels and both are G729, then no license is required. If you have two SIP channels, and one is G729 and the other is ulaw, then a license is required. Doug. -Original Message- From: Zoa [mailto:[EMAIL PROTECTED] Sent: Monday, January 08, 2007 10:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] G729 license counting Yes Zoa Michel wrote: Hello, How many licenses to buy?? : From what we understood from digium website, we must buy as many licenses as the number of maximum simultaneous calls using G729 Codec we wish to make. For example, If we want to be able to make a maximum of 10 simultaneous calls using G729 Codec, we must buy 10 licenses. Is it right? Thanks you ___ --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 ___ --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 ___ --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 Inbound (clean). Database: 0701-6, 01/08/2007 - 1/8/2007 2:47:33 PM ___ --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
Re: [asterisk-users] Some queries on g729 license.
On 1/8/07, Al Bochter [EMAIL PROTECTED] wrote: Mike, So tell me what this FREE open source G729 is I am told that you can use these Codecs with your Asterisk ! http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ You can do it Freely !! Al, I don't know if you're stupid, or you just like stirring things up. Once again, READ! Read the entire article before posting it. To quote: To use G.729 or G.723.1 you may need to pay a royalty fee. Please see http://www.sipro.com for details. Please note that this code is available for you to download for education purposes only and if a patent exists in your country for G.729 or G.723.1 then you should contact the owner of that patent and request their permission before executing the code. Now, Al, what does that say? I don't know what country you live in (and don't care), but if you live in a country (or possibly do business with a country) that honors patents, then you will have to pay to license this codec. Just because I _can_ break the law, does not mean that I should, or that I have the right to, or that it's ok to do so. ___ --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
Re: [asterisk-users] Some queries on g729 license.
Performance/Price wise which implementation of the codec is better? Digium or the Ready Tech/Intel IPP code? I'm looking at building a 4 PRI g.729 Asterisk box (Dell 2 x dual core, Digium 4 T1 + echo canceller). Which codec would provide the best audio quality? -- Matthew S. Crocker President Crocker Communications, Inc. Internet Division PO BOX 710 Greenfield, MA 01302-0710 http://www.crocker.com ___ --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
Re: [asterisk-users] Some queries on g729 license.
Al Bochter wrote: Mike, So tell me what this FREE open source G729 is I am told that you can use these Codecs with your Asterisk ! http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ You can do it Freely !! Please read the entire page. From the link you sent: Why NOT G.729? There are some reasons you might /not/ want or need to use G.729. * You don't want to pay the license fees or use the codec without the permission of the patent holder. ___ --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
Re: [asterisk-users] Some queries on g729 license.
Matthew Crocker wrote: I'm looking at building a 4 PRI g.729 Asterisk box (Dell 2 x dual core, Digium 4 T1 + echo canceller). Which codec would provide the best audio quality? G.729 is G.729 (assuming same suffixes like B, C, etc.). Audio quality is exactly the same, or the implementations aren't compatible. ___ --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
Re: [asterisk-users] Some queries on g729 license.
Mike I understand that. but it states on there site and note the key words may need What I want to know is if you buy 10 licenses from digum can use the Open Souce code? As long as you don't transcode than 10 at a time. Is that legal? I see the note about the IPP license From what I have been told this is easier to get working than Digum's G729 Legal Stuff - Important, please read To use G.729 or G.723.1 _*you may need to pay a royalty fee.*_ Please see http://www.sipro.com for details. Please note that this code is available for you to download for education purposes only and if a patent exists in your country for G.729 or G.723.1 then you should contact the owner of that patent and request their permission before executing the code. To distribute Intel's IPP libraries with a commercial product, you may need to pay a once-off license fee to Intel (currently $US180). My patches to Intel's code are distributed free under the GPL. Most of the code is just Intel's sample code re-arranged a little bit to work the way Asterisk expects. Therefore, this work would not have been possible without Intel doing 99.9% of the work. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Mike wrote: Al Bochter wrote: Mike, So tell me what this FREE open source G729 is I am told that you can use these Codecs with your Asterisk ! http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ You can do it Freely !! Please read the entire page. From the link you sent: Why NOT G.729? There are some reasons you might /not/ want or need to use G.729. * You don't want to pay the license fees or use the codec without the permission of the patent holder. ___ --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 Inbound (clean). Database: 0701-6, 01/08/2007 - 1/8/2007 3:24:08 PM ___ --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
Re: [asterisk-users] Asterisk from Debian Packages
Tzafrir Cohen wrote: On Mon, Dec 11, 2006 at 12:11:34AM +0100, Andreas von Heydwolff wrote: I'm using 1.2.13~dfsg-2 from Debian unstable in a small SOHO environment, it's doing its job. However, the startup scripts seem to hose something and it's running but not working with /etc/init.d/asterisk start, but running it from commandline solved the problem. Asterisk has been up for a couple weeks again. Hadn't the time to look into that yet, perhaps a problem with old config files from previous versions. Please report bugs (reportbug asterisk) . Others may have the same problem as you. Have you modified /etc/init.d/asterisk ? What do you have in /etc/default/asterisk? Hi again. Sorry, was just too busy in th meantime. It's all working just as it should, must have been a temporary glitch. 1.2.13~dfsg-2 is doing fine on a sarge/etch mix with debian kernel 2.6.18-8. Had to install the self compiled zaptel modules with # dpkg -i --force-overwrite though as some config file is shared with the kernel's. --AvH ___ --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
Re: [asterisk-users] Some queries on g729 license.
Sorry again Al but you are way off on this one also. sipro licenses digium who licenses end users for the digium product they are buying. It's like saying if I buy an ATA with 2 g.729 licenses can I throw it away and use the licenses with my open source codec? No way! Al Bochter wrote: Mike I understand that. but it states on there site and note the key words may need What I want to know is if you buy 10 licenses from digum can use the Open Souce code? As long as you don't transcode than 10 at a time. Is that legal? I see the note about the IPP license From what I have been told this is easier to get working than Digum's G729 Legal Stuff - Important, please read To use G.729 or G.723.1 _*you may need to pay a royalty fee.*_ Please see http://www.sipro.com for details. Please note that this code is available for you to download for education purposes only and if a patent exists in your country for G.729 or G.723.1 then you should contact the owner of that patent and request their permission before executing the code. To distribute Intel's IPP libraries with a commercial product, you may need to pay a once-off license fee to Intel (currently $US180). My patches to Intel's code are distributed free under the GPL. Most of the code is just Intel's sample code re-arranged a little bit to work the way Asterisk expects. Therefore, this work would not have been possible without Intel doing 99.9% of the work. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Mike wrote: Al Bochter wrote: Mike, So tell me what this FREE open source G729 is I am told that you can use these Codecs with your Asterisk ! http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ You can do it Freely !! Please read the entire page. From the link you sent: Why NOT G.729? There are some reasons you might /not/ want or need to use G.729. * You don't want to pay the license fees or use the codec without the permission of the patent holder. ___ --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 Inbound (clean). Database: 0701-6, 01/08/2007 - 1/8/2007 3:24:08 PM ___ --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 ___ --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
Re: [asterisk-users] Some queries on g729 license.
Al Bochter wrote: Mike I understand that. but it states on there site and note the key words may need What I want to know is if you buy 10 licenses from digum can use the Open Souce code? That is not what you said or asked. You were asserting that a free as in beer solution existed. If something says may it is incumbent upon you to decide if the rules/requirements in question are applicable to you, nobody else knows your situation. To answer your new question, as I am not an expert in patent law I haven't a clue. I see the note about the IPP license From what I have been told this is easier to get working than Digum's G729 I use Digium's codec and found it very easy to install. ___ --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
Re: [asterisk-users] Some queries on g729 license.
All of which hassle and expense can be avoided by buying a license for Digium's codec, which is tested to work well with Asterisk (and might come with some support). And is pretty cheap per simul call. I wonder whether that per call means per codec instance, which could be multiple licenses on a single conference call, where multiple (even if not all) parties are getting de/encoded simultaneously. And whether there are other tools for editing (/mixing/transforming) g729 data, in realtime (streams) or not (files), and whether they require a license. Ideally sox or equivalent would work on g729, maybe with a codec plugin. On Mon, 2007-01-08 at 13:23 -0500, Paul wrote: First point to tackle in any case involving patent, copyright or trademark infringement is whether or not the infringing party would have been qualified to buy any usage rights at all. In a case where you license the Intel source(read the terms, it's not really that free), you would be applying for a license under some plan that includes certain minimum payments. Even if you wrote new source from scratch you would be in the same boat. Last time I looked at the plans, I didn't see anything with low minimums. So even if you wrote code from scratch and never used it on more than 6 channels, you might have done something that normally requires a large upfront payment. Use $10k as an example. In such a case owner of the patent might have an attorney initiate contact. If you are willing to communicate they might allow you to pay the minimum and be licensed. If you can't do that, they might offer a settlement where you stop using the codec and pay them some lesser amount. If the patent holder can easily prove the violation you might as well try to deal with them and get things settled fast. If you sell or give away the codec it is easier for them to dig up proof. If you have unhappy employees that might be the way they hear about the violation in the first place. Important consideration: Bankruptcy law generally excludes debts created by things like malicious or criminal acts. Matthew Rubenstein wrote: As far as I know, the g729 patent requires buying a license to operate any implementation of it, whether Digium's, Intel's, or any other. Digium is set up to collect royalties (perhaps at a favorable rate) as part of their license from the patent holder. I don't know about Intel or any other. Or what the mechanics are for enforcing the patent on someone who operates a codec without a license. On Mon, 2007-01-08 at 10:51 -0500, Al Bochter wrote: What about the free open source G729 Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Matthew Rubenstein wrote: I connect to a PSTN carrier over SIP which requires me to connect with a g729 codec. I'm using them for just robocalling: Asterisk server originates calls which play a prerecorded file. Can I pre-encode those stored files in g729 so they don't need to be encoded for each call? If so, do I need a g729 license for each call, or just a license for the preencoder? If the robocalls accept incoming DTMF, do I need g729 licenses for those calls? On Mon, 2007-01-08 at 04:08 -0700, [EMAIL PROTECTED] wrote: Date: Mon, 08 Jan 2007 13:47:39 +0800 From: Leo Ann Boon [EMAIL PROTECTED] Subject: Re: [asterisk-users] Some queries on g729 license. 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 Xue Liangliang wrote: Hi, all I am a pabx vendor from Singapore. Recently we are going to implement a failover solution for our customers using heartbeat, the asterisk server can failover perfectly, however the g729 codec canot work, because it is binded the mac address, we have bought two set of licenses, can you provide us some workaround for this scenario? It shouldn't be a problem if you're only doing IP takeover and have bound the licenses to each server separately. If you're sharing the storage, then that could pose a problem. Leo DatVoiz Singapore Pte Ltd -- (C) Matthew Rubenstein ___ --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
RE: [asterisk-users] Some queries on g729 license.
per call means per terminating channel where encoding/decoding is required. Termination could be to translate to another codec (with another peer) or to Asterisk itself to handle menus, voicemail, conference calls. In the conference call setup, each caller uses a license. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Rubenstein Sent: Monday, January 08, 2007 12:56 To: Paul Cc: Asterisk-Users Subject: Re: [asterisk-users] Some queries on g729 license. All of which hassle and expense can be avoided by buying a license for Digium's codec, which is tested to work well with Asterisk (and might come with some support). And is pretty cheap per simul call. I wonder whether that per call means per codec instance, which could be multiple licenses on a single conference call, where multiple (even if not all) parties are getting de/encoded simultaneously. And whether there are other tools for editing (/mixing/transforming) g729 data, in realtime (streams) or not (files), and whether they require a license. Ideally sox or equivalent would work on g729, maybe with a codec plugin. On Mon, 2007-01-08 at 13:23 -0500, Paul wrote: First point to tackle in any case involving patent, copyright or trademark infringement is whether or not the infringing party would have been qualified to buy any usage rights at all. In a case where you license the Intel source(read the terms, it's not really that free), you would be applying for a license under some plan that includes certain minimum payments. Even if you wrote new source from scratch you would be in the same boat. Last time I looked at the plans, I didn't see anything with low minimums. So even if you wrote code from scratch and never used it on more than 6 channels, you might have done something that normally requires a large upfront payment. Use $10k as an example. In such a case owner of the patent might have an attorney initiate contact. If you are willing to communicate they might allow you to pay the minimum and be licensed. If you can't do that, they might offer a settlement where you stop using the codec and pay them some lesser amount. If the patent holder can easily prove the violation you might as well try to deal with them and get things settled fast. If you sell or give away the codec it is easier for them to dig up proof. If you have unhappy employees that might be the way they hear about the violation in the first place. Important consideration: Bankruptcy law generally excludes debts created by things like malicious or criminal acts. Matthew Rubenstein wrote: As far as I know, the g729 patent requires buying a license to operate any implementation of it, whether Digium's, Intel's, or any other. Digium is set up to collect royalties (perhaps at a favorable rate) as part of their license from the patent holder. I don't know about Intel or any other. Or what the mechanics are for enforcing the patent on someone who operates a codec without a license. On Mon, 2007-01-08 at 10:51 -0500, Al Bochter wrote: What about the free open source G729 Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Matthew Rubenstein wrote: I connect to a PSTN carrier over SIP which requires me to connect with a g729 codec. I'm using them for just robocalling: Asterisk server originates calls which play a prerecorded file. Can I pre-encode those stored files in g729 so they don't need to be encoded for each call? If so, do I need a g729 license for each call, or just a license for the preencoder? If the robocalls accept incoming DTMF, do I need g729 licenses for those calls? On Mon, 2007-01-08 at 04:08 -0700, [EMAIL PROTECTED] wrote: Date: Mon, 08 Jan 2007 13:47:39 +0800 From: Leo Ann Boon [EMAIL PROTECTED] Subject: Re: [asterisk-users] Some queries on g729 license. 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 Xue Liangliang wrote: Hi, all I am a pabx vendor from Singapore. Recently we are going to implement a failover solution for our customers using heartbeat, the asterisk server can failover perfectly, however the g729 codec canot work, because it is binded the mac address, we have bought two set of licenses, can you provide us some workaround for this scenario? It shouldn't be a problem if you're only doing IP takeover and have bound the licenses to each server separately. If you're sharing the storage, then that could pose a problem. Leo DatVoiz Singapore Pte Ltd -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com --
[asterisk-users] delete=yes is not working
Hey guys. This is the setup that I have for a voicemail account. 1509 = 1509,Mark Greene,[EMAIL PROTECTED],attach=yes|delete=yes It emails me the voicemail, but it does not delete it from the system afterwards. I have also tried 1509 = 1509,Mark Greene,[EMAIL PROTECTED],attach=yes|delete=1 Any ideas on this? - Mark ___ --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] snom 190 (etc.?) dialscript for * debugging and kaddressbook
Thought I might just as well share these scripts, they may work with other phones too: *1)* Dialing from the KDE 3.5.5 address book works with a script that gets triggered from the kaddressbook (Settings - Script Hooks - Phone) with my command snom_dial_number %N The script snom_dial_number itself goes like this: - #!/bin/sh ENCODEDNUMBER=$(echo $@ | sed 's/\+/00/g' | sed 's/\///g' | sed \ 's/-//g' | sed 's/\#/\%23/g' |sed 's/ //g' | sed 's/0043/0/g') konqueror -geometry 700x30+350-810 \ http://172.16.0.2/command.htm?DIAL=$ENCODEDNUMBERDIAL #EOF Substitute 172.16.0.2 with your phone's IP number and 0043 with your country code. The format of numbers in my address book is +CC AC NUMBER which works also for exporting via gnokii to my Nokia mobile. The script handles the empty spaces and eventual hyphens. (BTW, for SMS sending via bluetooth I added to the script hooks cat %F | gnokii --sendsms %N) *2)* When working on the dialplan on the office asterisk server via ssh from home I needed to test outgoing calls - but nobody was physically there. What to do? Being logged in on a shell on my remote asterisk machine I used the following script to trigger outgoing calls from an office snom 190 phone to my phone beside me on the desk. A timeout of 3 secs for POTS or 15 secs for my mobile guaranteed that no voicebox would take over but I heard a short ring when calls got through, to add a real life ringtone to remote visual feedback from asterisk -rv. httpsnom-dialtest - #!/bin/bash # Created 070107 by AvH # $1 is the extension to dial if [ $1 = ] then echo enter number please ; exit fi # command for snom 190 phone, taken from # http://80.237.155.31/kb/index.php?View=entryCategoryID=21EntryID=40 SOURCE=command.htm?number=$1 # origin EXT=2 # IP number of phone echo Dialing from $EXT # the actual command, -w is a timeout echo -e GET $SOURCE HTTP/1.0\n\n | nc -w 1 $EXT 80 /dev/null #EOF I guess the second script can be put into use for KDE as well. Any ideas for improvements? Cheers, --AvH ___ --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
Re: [asterisk-users] delete=yes is not working
On 15:17, Mon 08 Jan 07, Mark Greene wrote: Hey guys. This is the setup that I have for a voicemail account. 1509 = 1509,Mark Greene,[EMAIL PROTECTED],attach=yes|delete=yes try this: 1509 = 1509,Mark Greene,[EMAIL PROTECTED],,attach=yes|delete=yes You forgot about the pager field. It emails me the voicemail, but it does not delete it from the system afterwards. I have also tried 1509 = 1509,Mark Greene,[EMAIL PROTECTED],attach=yes|delete=1 Any ideas on this? -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --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] SIP rt load from db
Anyone know the command that tells * to load a sipfriend from the realtime db rather than saying no such host? I've tried various combinations of the rt commands: rtcachefriends=yes; ;rtcache=yes ;rtAutoClear=yes ;rtautoreg=yes ;rtIgnoreRegExpire=yes ;rtupdate=yes rtfromcontact=yes Basically I have a group of 4 * servers all routing calls, but only two are hearing the phones registration. I'd like the other two to load the sipfriends entry from mysql when a channel for that sipfriends is requested. Any ideas? ___ --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
Re: [asterisk-users] delete=yes is not working
I have googled and I do not understand how the pager field is what is causing the problem. Could you explain? ___ --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
Re: [asterisk-users] Some queries on g729 license.
G.729 is G.729 (assuming same suffixes like B, C, etc.). Audio quality is exactly the same, or the implementations aren't compatible. Yes, but depending on the implementation the CPU resources between two could be quite different. Audio quality could be adversely affected by inadequate CPU resources with a bad implementation. So, in other words, which asterisk g.729 is better on CPU utilization Digium or Ready Tech/Intel IPP? Digium certainly knows Asterisk but I'm sure Intel knows their CPUs pretty well too -Matt -- Matthew S. Crocker President Crocker Communications, Inc. Internet Division PO BOX 710 Greenfield, MA 01302-0710 http://www.crocker.com ___ --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
Re: [asterisk-users] delete=yes is not working
On 15:51, Mon 08 Jan 07, Mark Greene wrote: I have googled and I do not understand how the pager field is what is causing the problem. Could you explain? If you dont provide it the parser will think the pager address is 'delete=yes|attach=yes' -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --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
Re: [asterisk-users] delete=yes is not working
Mark Greene wrote: I have googled and I do not understand how the pager field is what is causing the problem. Could you explain? Think of it as a CSV file. The ,, entry for pager is just a placeholder saying that for pager there is nothing. Omitting means that the next field will be treated as pager info, rather then for whatever it is actually intended. ___ --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
Re: [asterisk-users] Some queries on g729 license.
Mike What I was looking to do is use the easier to install one and the better one. I was asked by a customer about using G729 and I told the customer that they would have to pay for the G729 licenses. The customer pointed out the open source G729 code and I was not sure if I could use that. Then I was told by others that work on Asterisk that the open G729 was a cracked ver of Digum G729 and don't use it without buying the Digum licenses. So that is what I am tring to found out. And Paul did point that out that the open G729 and Digums code is not the same. I don't have Open G729 or Digum G729 installed in the Asterisk server. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Mike wrote: Al Bochter wrote: Mike I understand that. but it states on there site and note the key words may need What I want to know is if you buy 10 licenses from digum can use the Open Souce code? That is not what you said or asked. You were asserting that a free as in beer solution existed. If something says may it is incumbent upon you to decide if the rules/requirements in question are applicable to you, nobody else knows your situation. To answer your new question, as I am not an expert in patent law I haven't a clue. I see the note about the IPP license From what I have been told this is easier to get working than Digum's G729 I use Digium's codec and found it very easy to install. ___ --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 Inbound (clean). Database: 0701-6, 01/08/2007 - 1/8/2007 3:57:10 PM ___ --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
Re: [asterisk-users] Some queries on g729 license.
Matthew I agree. I only know what I have told by others so I do need this input I have been told that Digum G729 is a big pain the the butt to get working with Asterisk and it is very hard on the CPU Keep in mind I have never used any Ver. of G 729 So tell me what you think. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Matthew Rubenstein wrote: All of which hassle and expense can be avoided by buying a license for Digium's codec, which is tested to work well with Asterisk (and might come with some support). And is pretty cheap per simul call. I wonder whether that per call means per codec instance, which could be multiple licenses on a single conference call, where multiple (even if not all) parties are getting de/encoded simultaneously. And whether there are other tools for editing (/mixing/transforming) g729 data, in realtime (streams) or not (files), and whether they require a license. Ideally sox or equivalent would work on g729, maybe with a codec plugin. On Mon, 2007-01-08 at 13:23 -0500, Paul wrote: First point to tackle in any case involving patent, copyright or trademark infringement is whether or not the infringing party would have been qualified to buy any usage rights at all. In a case where you license the Intel source(read the terms, it's not really that free), you would be applying for a license under some plan that includes certain minimum payments. Even if you wrote new source from scratch you would be in the same boat. Last time I looked at the plans, I didn't see anything with low minimums. So even if you wrote code from scratch and never used it on more than 6 channels, you might have done something that normally requires a large upfront payment. Use $10k as an example. In such a case owner of the patent might have an attorney initiate contact. If you are willing to communicate they might allow you to pay the minimum and be licensed. If you can't do that, they might offer a settlement where you stop using the codec and pay them some lesser amount. If the patent holder can easily prove the violation you might as well try to deal with them and get things settled fast. If you sell or give away the codec it is easier for them to dig up proof. If you have unhappy employees that might be the way they hear about the violation in the first place. Important consideration: Bankruptcy law generally excludes debts created by things like malicious or criminal acts. Matthew Rubenstein wrote: As far as I know, the g729 patent requires buying a license to operate any implementation of it, whether Digium's, Intel's, or any other. Digium is set up to collect royalties (perhaps at a favorable rate) as part of their license from the patent holder. I don't know about Intel or any other. Or what the mechanics are for enforcing the patent on someone who operates a codec without a license. On Mon, 2007-01-08 at 10:51 -0500, Al Bochter wrote: What about the free open source G729 Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Matthew Rubenstein wrote: I connect to a PSTN carrier over SIP which requires me to connect with a g729 codec. I'm using them for just robocalling: Asterisk server originates calls which play a prerecorded file. Can I pre-encode those stored files in g729 so they don't need to be encoded for each call? If so, do I need a g729 license for each call, or just a license for the preencoder? If the robocalls accept incoming DTMF, do I need g729 licenses for those calls? On Mon, 2007-01-08 at 04:08 -0700, [EMAIL PROTECTED] wrote: Date: Mon, 08 Jan 2007 13:47:39 +0800 From: Leo Ann Boon [EMAIL PROTECTED] Subject: Re: [asterisk-users] Some queries on g729 license. 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 Xue Liangliang wrote: Hi, all I am a pabx vendor from Singapore. Recently we are going to implement a failover solution for our customers using heartbeat, the asterisk server can failover perfectly, however the g729 codec canot work, because it is binded the mac address, we have bought two set of licenses, can you provide us some workaround for this scenario? It shouldn't be a problem if you're only doing IP takeover and have bound the licenses to each server separately. If you're sharing the storage, then that could pose a problem. Leo DatVoiz Singapore Pte Ltd ___ --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
Re: [asterisk-users] Block some number outgoing from joust oneextention
post here your extensions.conf On 1/8/07, Mattias Andersson [EMAIL PROTECTED] wrote: Hi! Unfortunately did this stop Asterisk to register ny phones and trunk. Did I put tit in the wrong place? //Mattias Hi! Exactly what I needed. It was the 209 part that I did not figure put. Thanks! //Mattias At 03:53 2007-01-05, you wrote: exten = _9070X./209,1,NoOP,SORRY CHARLIE exten = _9070X./209,2,Congestion This would block any call from 209 to 070X as long as 9 was your outside digit. I use the NoOP to help me out with the CLI and debugging :) Hope this helps Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mattias Andersson Sent: Thursday, January 04, 2007 5:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Block some number outgoing from joust oneextention Thanks! I can´t rely figure out how to block for only one extension. Eg. Extension 209 need to be blocked from making calls starting with 070 (eg. 9070). Some clues did I get bout would it men a new form-internal-blocked dialplan? Regards Mattias On 04/01/07, C F mailto:[EMAIL PROTECTED][EMAIL PROTECTED] wrote: The easiest way is thru using contexts. On 1/3/07, Mattias Andersson mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi all! I am shore someone have writing about it bout I cant find it. I have a extension that I need to block from making expansive mobil calls. Everyone else should be aloud to do the calls. I am shore it is possible to be done sens I had a commercial asterisk based PBX that I did that on. However I have switch to Trixbox because I need some custom functions not supported by the commercial product. I would appreciate all help. Regards Mattias Adress: Mattias Andersson Storskiftesvägen 6 S-145 60 Norsborg Mobil: +46-70-799 44 41 Email: mailto:[EMAIL PROTECTED][EMAIL PROTECTED] Skype: eskes1 ___ --Bandwidth and Colocation provided by http://Easynews.comEasynews.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://Easynews.comEasynews.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users -- Mattias Andersson Storskiftesvägen 6 145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38 97 Email: mailto:[EMAIL PROTECTED][EMAIL PROTECTED] -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.432 / Virus Database: 268.16.4/615 - Release Date: 1/3/2007 1:34 PM ___ --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 Adress: Mattias Andersson Storskiftesvägen 6 S-145 60 Norsborg Mobil: +46-70-799 44 41 Email: [EMAIL PROTECTED] Skype: eskes1 ___ --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 ___ --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
Re: [asterisk-users] Some queries on g729 license.
I did some tests a long time ago and the speed was roughly the same. ( I think digium's was slightly faster). I think the IPP version also doesn't work on AMD out of the box. It's just 10$ a channel, that's not even worth the hassle of trying something else. Joachim Al Bochter wrote: Matthew I agree. I only know what I have told by others so I do need this input I have been told that Digum G729 is a big pain the the butt to get working with Asterisk and it is very hard on the CPU Keep in mind I have never used any Ver. of G 729 So tell me what you think. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Matthew Rubenstein wrote: All of which hassle and expense can be avoided by buying a license for Digium's codec, which is tested to work well with Asterisk (and might come with some support). And is pretty cheap per simul call. I wonder whether that per call means per codec instance, which could be multiple licenses on a single conference call, where multiple (even if not all) parties are getting de/encoded simultaneously. And whether there are other tools for editing (/mixing/transforming) g729 data, in realtime (streams) or not (files), and whether they require a license. Ideally sox or equivalent would work on g729, maybe with a codec plugin. On Mon, 2007-01-08 at 13:23 -0500, Paul wrote: First point to tackle in any case involving patent, copyright or trademark infringement is whether or not the infringing party would have been qualified to buy any usage rights at all. In a case where you license the Intel source(read the terms, it's not really that free), you would be applying for a license under some plan that includes certain minimum payments. Even if you wrote new source from scratch you would be in the same boat. Last time I looked at the plans, I didn't see anything with low minimums. So even if you wrote code from scratch and never used it on more than 6 channels, you might have done something that normally requires a large upfront payment. Use $10k as an example. In such a case owner of the patent might have an attorney initiate contact. If you are willing to communicate they might allow you to pay the minimum and be licensed. If you can't do that, they might offer a settlement where you stop using the codec and pay them some lesser amount. If the patent holder can easily prove the violation you might as well try to deal with them and get things settled fast. If you sell or give away the codec it is easier for them to dig up proof. If you have unhappy employees that might be the way they hear about the violation in the first place. Important consideration: Bankruptcy law generally excludes debts created by things like malicious or criminal acts. Matthew Rubenstein wrote: As far as I know, the g729 patent requires buying a license to operate any implementation of it, whether Digium's, Intel's, or any other. Digium is set up to collect royalties (perhaps at a favorable rate) as part of their license from the patent holder. I don't know about Intel or any other. Or what the mechanics are for enforcing the patent on someone who operates a codec without a license. On Mon, 2007-01-08 at 10:51 -0500, Al Bochter wrote: What about the free open source G729 Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Matthew Rubenstein wrote: I connect to a PSTN carrier over SIP which requires me to connect with a g729 codec. I'm using them for just robocalling: Asterisk server originates calls which play a prerecorded file. Can I pre-encode those stored files in g729 so they don't need to be encoded for each call? If so, do I need a g729 license for each call, or just a license for the preencoder? If the robocalls accept incoming DTMF, do I need g729 licenses for those calls? On Mon, 2007-01-08 at 04:08 -0700, [EMAIL PROTECTED] wrote: Date: Mon, 08 Jan 2007 13:47:39 +0800 From: Leo Ann Boon [EMAIL PROTECTED] Subject: Re: [asterisk-users] Some queries on g729 license. 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 Xue Liangliang wrote: Hi, all I am a pabx vendor from Singapore. Recently we are going to implement a failover solution for our customers using heartbeat, the asterisk server can failover perfectly, however the g729 codec canot work, because it is binded the mac address, we have bought two set of licenses, can you provide us some workaround for this scenario? It shouldn't be a problem if you're only doing IP takeover and have bound the licenses to each server separately. If you're sharing the storage, then that could pose a problem. Leo DatVoiz Singapore Pte Ltd
Re: [asterisk-users] Some queries on g729 license.
Did you find any operations trouble installing/using the Digium codec with Asterisk? I'd be surprised if Digium's were hard to use with Asterisk, considering they wrote and support both. Also can their codec be used to pre-encode data to files from a Linux command/line? Or just the Asterisk CLI mentioned earlier in this thread? On Tue, 2007-01-09 at 00:31 +0200, Zoa wrote: I did some tests a long time ago and the speed was roughly the same. ( I think digium's was slightly faster). I think the IPP version also doesn't work on AMD out of the box. It's just 10$ a channel, that's not even worth the hassle of trying something else. Joachim Al Bochter wrote: Matthew I agree. I only know what I have told by others so I do need this input I have been told that Digum G729 is a big pain the the butt to get working with Asterisk and it is very hard on the CPU Keep in mind I have never used any Ver. of G 729 So tell me what you think. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Matthew Rubenstein wrote: All of which hassle and expense can be avoided by buying a license for Digium's codec, which is tested to work well with Asterisk (and might come with some support). And is pretty cheap per simul call. I wonder whether that per call means per codec instance, which could be multiple licenses on a single conference call, where multiple (even if not all) parties are getting de/encoded simultaneously. And whether there are other tools for editing (/mixing/transforming) g729 data, in realtime (streams) or not (files), and whether they require a license. Ideally sox or equivalent would work on g729, maybe with a codec plugin. On Mon, 2007-01-08 at 13:23 -0500, Paul wrote: First point to tackle in any case involving patent, copyright or trademark infringement is whether or not the infringing party would have been qualified to buy any usage rights at all. In a case where you license the Intel source(read the terms, it's not really that free), you would be applying for a license under some plan that includes certain minimum payments. Even if you wrote new source from scratch you would be in the same boat. Last time I looked at the plans, I didn't see anything with low minimums. So even if you wrote code from scratch and never used it on more than 6 channels, you might have done something that normally requires a large upfront payment. Use $10k as an example. In such a case owner of the patent might have an attorney initiate contact. If you are willing to communicate they might allow you to pay the minimum and be licensed. If you can't do that, they might offer a settlement where you stop using the codec and pay them some lesser amount. If the patent holder can easily prove the violation you might as well try to deal with them and get things settled fast. If you sell or give away the codec it is easier for them to dig up proof. If you have unhappy employees that might be the way they hear about the violation in the first place. Important consideration: Bankruptcy law generally excludes debts created by things like malicious or criminal acts. Matthew Rubenstein wrote: As far as I know, the g729 patent requires buying a license to operate any implementation of it, whether Digium's, Intel's, or any other. Digium is set up to collect royalties (perhaps at a favorable rate) as part of their license from the patent holder. I don't know about Intel or any other. Or what the mechanics are for enforcing the patent on someone who operates a codec without a license. On Mon, 2007-01-08 at 10:51 -0500, Al Bochter wrote: What about the free open source G729 Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Matthew Rubenstein wrote: I connect to a PSTN carrier over SIP which requires me to connect with a g729 codec. I'm using them for just robocalling: Asterisk server originates calls which play a prerecorded file. Can I pre-encode those stored files in g729 so they don't need to be encoded for each call? If so, do I need a g729 license for each call, or just a license for the preencoder? If the robocalls accept incoming DTMF, do I need g729 licenses for those calls? On Mon, 2007-01-08 at 04:08 -0700, [EMAIL PROTECTED] wrote: Date: Mon, 08 Jan 2007 13:47:39 +0800 From: Leo Ann Boon [EMAIL PROTECTED] Subject: Re: [asterisk-users] Some queries on g729 license. 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 Xue Liangliang wrote: Hi, all I am a pabx vendor from Singapore.
Re: [asterisk-users] jitterbuffer on sip.conf
[EMAIL PROTECTED] wrote: In iax.conf there is option jitterbuffer how about sip protocol ? Are jitterbuffer can configure in sip.conf ? In 1.2 and 1.0 there is no jitter buffer for SIP. I think 1.4 might have a SIP jitter buffer, but I'm not sure. Check sip.conf.sample in 1.4. ___ --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
Re: [asterisk-users] Hanging up a 3-way conference when middle user hangs up
Lex Lethol wrote: As far as I know when I setup a 3-way on something like a cisco will disconnect everyone when the middle (person who setup the conference) hangs up. The problem I describe happens on ATAs and the like that uses flash to put on hold while setting up the second call. I am not sure about other phones other than cisco, polycom and a few others. Then the issue is with the ATA config, not Asterisk. Make sure the ATA is set up for 3-way calling, not 3-way conference or 3-way transfer. ___ --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
Re: [asterisk-users] Some queries on g729 license.
On 1/8/07, Al Bochter [EMAIL PROTECTED] wrote: I agree. I only know what I have told by others so I do need this input I have been told that Digum G729 is a big pain the the butt to get working with Asterisk and it is very hard on the CPU Keep in mind I have never used any Ver. of G 729 I only know what I have been told... I have been told... I have never used... All common phrases with this person. I have never seen somebody spread as much third party information as this person spreads. He knows nothing, yet informs all. It's real simple to give it a shot and see what happens. If you can't afford $10 for testing, maybe you are in the wrong business. You claim to have clients using Asterisk. I'd hate to be one of your clients. I would hope that my consultant has at least tried the things he/she is suggesting to me. FWIW, I have used Digium's g729 and it works great, and is about as simple to install as you can get. ___ --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
Re: [asterisk-users] How to get dial tone back
The DISA application (show application disa at the CLI) will let you do this. Mojo On Saturday, January 06, 2007 10:19 pm, Yuan LIU wrote: After the user navigated some voice menus, how do I give him another (fake) dial tone? Yuan Liu ___ --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 ___ --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] Call Sound Volume Low : between extensions and over ZAP.
Hi All, I am starting to get complains from users that the call volume is very low and people are having problems haring what is said. This is for internal calls (between extensions) and over ZAP. The problem seems to be with the caller and callee, no matter if it is an incoming or outgoing call. I know you can increase the gain levels on the digium cards but is it possible to increase the volume of voice overall? ___ --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
Re: [asterisk-users] Call Sound Volume Low : between extensions and over ZAP.
Klaverstyn, David C wrote: Hi All, I am starting to get complains from users that the call volume is very low and people are having problems haring what is said. This is for internal calls (between extensions) and over ZAP. The problem seems to be with the caller and callee, no matter if it is an incoming or outgoing call. I know you can increase the gain levels on the digium cards but is it possible to increase the volume of voice overall? No. Gain is done at the interface. i.e. gains for Zap are in /etc/asterisk/zapata.conf and the gains for SIP are in the SIP device. ___ --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
RE: [asterisk-users] Call Sound Volume Low : between extensions andover ZAP.
Thanks Eric, That's what I figured but I wanted to make sure that it was not possible to increase VoIP volume levels. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Tuesday, 9 January 2007 9:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call Sound Volume Low : between extensions andover ZAP. Klaverstyn, David C wrote: Hi All, I am starting to get complains from users that the call volume is very low and people are having problems haring what is said. This is for internal calls (between extensions) and over ZAP. The problem seems to be with the caller and callee, no matter if it is an incoming or outgoing call. I know you can increase the gain levels on the digium cards but is it possible to increase the volume of voice overall? No. Gain is done at the interface. i.e. gains for Zap are in /etc/asterisk/zapata.conf and the gains for SIP are in the SIP device. ___ --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 ___ --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
Re: [asterisk-users] delete=yes is not working
OK that makes a lot more sense. Thanks for the explanation. - Mark ___ --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
Re: [asterisk-users] Some queries on g729 license.
It sounds fairly easy to me. If I had a 1.4 system built I would write something in perl to do that and put it under gpl. You could also do it in php or anything else that can be run from command line and has an asterisk manager interface available. I don't even need the g729 codec to get it running and delivered. If it works for the other codecs it just needs to be tested by someone who has 1.4 with g729 up and running. Matthew Rubenstein wrote: Did you find any operations trouble installing/using the Digium codec with Asterisk? I'd be surprised if Digium's were hard to use with Asterisk, considering they wrote and support both. Also can their codec be used to pre-encode data to files from a Linux command/line? Or just the Asterisk CLI mentioned earlier in this thread? On Tue, 2007-01-09 at 00:31 +0200, Zoa wrote: I did some tests a long time ago and the speed was roughly the same. ( I think digium's was slightly faster). I think the IPP version also doesn't work on AMD out of the box. It's just 10$ a channel, that's not even worth the hassle of trying something else. Joachim Al Bochter wrote: Matthew I agree. I only know what I have told by others so I do need this input I have been told that Digum G729 is a big pain the the butt to get working with Asterisk and it is very hard on the CPU Keep in mind I have never used any Ver. of G 729 So tell me what you think. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Matthew Rubenstein wrote: All of which hassle and expense can be avoided by buying a license for Digium's codec, which is tested to work well with Asterisk (and might come with some support). And is pretty cheap per simul call. I wonder whether that per call means per codec instance, which could be multiple licenses on a single conference call, where multiple (even if not all) parties are getting de/encoded simultaneously. And whether there are other tools for editing (/mixing/transforming) g729 data, in realtime (streams) or not (files), and whether they require a license. Ideally sox or equivalent would work on g729, maybe with a codec plugin. On Mon, 2007-01-08 at 13:23 -0500, Paul wrote: First point to tackle in any case involving patent, copyright or trademark infringement is whether or not the infringing party would have been qualified to buy any usage rights at all. In a case where you license the Intel source(read the terms, it's not really that free), you would be applying for a license under some plan that includes certain minimum payments. Even if you wrote new source from scratch you would be in the same boat. Last time I looked at the plans, I didn't see anything with low minimums. So even if you wrote code from scratch and never used it on more than 6 channels, you might have done something that normally requires a large upfront payment. Use $10k as an example. In such a case owner of the patent might have an attorney initiate contact. If you are willing to communicate they might allow you to pay the minimum and be licensed. If you can't do that, they might offer a settlement where you stop using the codec and pay them some lesser amount. If the patent holder can easily prove the violation you might as well try to deal with them and get things settled fast. If you sell or give away the codec it is easier for them to dig up proof. If you have unhappy employees that might be the way they hear about the violation in the first place. Important consideration: Bankruptcy law generally excludes debts created by things like malicious or criminal acts. Matthew Rubenstein wrote: As far as I know, the g729 patent requires buying a license to operate any implementation of it, whether Digium's, Intel's, or any other. Digium is set up to collect royalties (perhaps at a favorable rate) as part of their license from the patent holder. I don't know about Intel or any other. Or what the mechanics are for enforcing the patent on someone who operates a codec without a license. On Mon, 2007-01-08 at 10:51 -0500, Al Bochter wrote: What about the free open source G729 Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Matthew Rubenstein wrote: I connect to a PSTN carrier over SIP which requires me to connect with a g729 codec. I'm using them for just robocalling: Asterisk server originates calls which play a prerecorded file. Can I pre-encode those stored files in g729 so they don't need to be encoded for each call? If so, do I need a g729 license for each call, or just a license for the preencoder? If the robocalls accept incoming DTMF, do I need g729 licenses for those calls? On Mon, 2007-01-08 at 04:08 -0700, [EMAIL PROTECTED] wrote: Date: Mon, 08 Jan 2007 13:47:39 +0800 From: Leo Ann Boon [EMAIL PROTECTED] Subject: Re: [asterisk-users] Some
Re: [asterisk-users] Call Sound Volume Low : between extensions andover ZAP.
All SIP devices I have used allow you to increase the volume levels. But it is done in the DEVICE, not in Asterisk. Klaverstyn, David C wrote: Thanks Eric, That's what I figured but I wanted to make sure that it was not possible to increase VoIP volume levels. ___ --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
Re: [asterisk-users] Asterisk and MiniITX setups
Looks like indeed I was underclocked, new results: [EMAIL PROTECTED]:~# cat /proc/cpuinfo processor : 0 vendor_id : CentaurHauls cpu family : 6 model : 10 model name : VIA Esther processor 1200MHz stepping: 9 cpu MHz : 1197.301 cache size : 128 KB fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge cmov pat clflush acpi mmx fxsr sse sse2 tm pni est tm2 rng rng_en ace ace_en bogomips: 2398.50 pbx*CLI show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - - - - - - - - - - - gsm - - 4 410 4 316 - -61 ulaw -11 - 1 8 2 114 - -59 alaw -11 1 - 8 2 114 - -59 g726 -16 7 7 - 7 619 - -64 adpcm -11 2 2 8 - 114 - -59 slin -10 1 1 7 1 -13 - -58 lpc10 -473838443837 - - -95 g729 - - - - - - - - - - - speex - - - - - - - - - - - ilbc -20111117111023 - - - Thanks again On 12/30/06, C F [EMAIL PROTECTED] wrote: Thanks guys, looks like that is the problem I am wondering why I didn't notice it. Will try to change it in the bios and see if that helps. I'll report back On 12/29/06, Gordon Henderson [EMAIL PROTECTED] wrote: On Fri, 29 Dec 2006, Paul wrote: Doug Lytle wrote: C F wrote: Gordon, how did you get such good numbers? model name : VIA Esther processor 1200MHz cpu MHz : 399.054 Is this accurate? A 1200mhz cpu running at 399? Under clocked? Maybe it is intentionally underclocked to run without a cpu fan. My money would be on accidental incorrect BIOS settings - eg. Load Safety defaults setting or something. I did have one of hese boards do this on me after I had to do a NV-RAM clear after screwing up the video system! These VIA processors are quite hardy - I could probably overclock them if needed - the 1GHz ones I'm using run cool to the point of being cold, as do the 533MHz ones I'm using. I have a couple of 1.2GHz ones with fans, (in a router application) and the fans are temperature controlled directly by the motherboard, and I've tried my hardest to get them to spin up when in-use and they stubbornly stay off! (they do spin up at boot time, so I know they do work! but for other applications - eg. totally fanless set-top boxes, these are the works and some of them have hardware Mpeg decode - which is OT for asterisk though!) Gordon ___ --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 ___ --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
Re: [asterisk-users] Adding 4000 Lines to asteriskdb via asterisk -rx ?
On Mon, Jan 08, 2007 at 04:59:58PM +0100, Christoph Adomeit wrote: Hi there, I want to add 4000 Callerids and Callernames to my asterisk-db. (/var/lib/asterisk/astdb) I do not want an external database or an sql-database because I do not want asterisk to depend on external processes. However, when I do 4000 database put number name via a shellscript and asterisk -rx I only have 600 entries later in my asterisk- database. The asterisk sockets seems not to be designed for bulk updates to the asterisk-db. I also don't want to add 4000 sleep 1 to my shell-script. If you have a 'sleep 1', you have a badly written script. You should wait for Asterisk to finish. However 4000 invocations of Asterisk can be a pain. Asterisk takes quite a while to start. I consider Asterisk's behaviour here a bug. Sadly others don't agree. Does anybody have an Idea how to add these lines to asterisk ? I managed to Build a Perl DB_File Module for db1.8.5 but I do not have the know how how to use DB_file and db1 databases. Option A: Use the manager interface. Option B: write directly to the socket: ere: usint a small utility called socat (there's that package in Debian) http://threebit.net/mail-archive/asterisk-users/msg27519.html The downside is that you can't tell of you had an error. Are there some external utilities to lock and update the asteriskdb ? Grab the sources of the berkely DB from the Asterisk source. It should have a utility for manipulating a databae. Nothing about locking, though, I guess. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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
Re: [asterisk-users] Some queries on g729 license.
The Intel IPP-based G.729 codec does work with AMD processors out of the box, both with the 32 bit and 64 bit versions. On Mon January 8 2007 19:31, Zoa wrote: I did some tests a long time ago and the speed was roughly the same. ( I think digium's was slightly faster). I think the IPP version also doesn't work on AMD out of the box. It's just 10$ a channel, that's not even worth the hassle of trying something else. Joachim Al Bochter wrote: Matthew I agree. I only know what I have told by others so I do need this input I have been told that Digum G729 is a big pain the the butt to get working with Asterisk and it is very hard on the CPU Keep in mind I have never used any Ver. of G 729 So tell me what you think. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Matthew Rubenstein wrote: All of which hassle and expense can be avoided by buying a license for Digium's codec, which is tested to work well with Asterisk (and might come with some support). And is pretty cheap per simul call. I wonder whether that per call means per codec instance, which could be multiple licenses on a single conference call, where multiple (even if not all) parties are getting de/encoded simultaneously. And whether there are other tools for editing (/mixing/transforming) g729 data, in realtime (streams) or not (files), and whether they require a license. Ideally sox or equivalent would work on g729, maybe with a codec plugin. On Mon, 2007-01-08 at 13:23 -0500, Paul wrote: First point to tackle in any case involving patent, copyright or trademark infringement is whether or not the infringing party would have been qualified to buy any usage rights at all. In a case where you license the Intel source(read the terms, it's not really that free), you would be applying for a license under some plan that includes certain minimum payments. Even if you wrote new source from scratch you would be in the same boat. Last time I looked at the plans, I didn't see anything with low minimums. So even if you wrote code from scratch and never used it on more than 6 channels, you might have done something that normally requires a large upfront payment. Use $10k as an example. In such a case owner of the patent might have an attorney initiate contact. If you are willing to communicate they might allow you to pay the minimum and be licensed. If you can't do that, they might offer a settlement where you stop using the codec and pay them some lesser amount. If the patent holder can easily prove the violation you might as well try to deal with them and get things settled fast. If you sell or give away the codec it is easier for them to dig up proof. If you have unhappy employees that might be the way they hear about the violation in the first place. Important consideration: Bankruptcy law generally excludes debts created by things like malicious or criminal acts. Matthew Rubenstein wrote: As far as I know, the g729 patent requires buying a license to operate any implementation of it, whether Digium's, Intel's, or any other. Digium is set up to collect royalties (perhaps at a favorable rate) as part of their license from the patent holder. I don't know about Intel or any other. Or what the mechanics are for enforcing the patent on someone who operates a codec without a license. On Mon, 2007-01-08 at 10:51 -0500, Al Bochter wrote: What about the free open source G729 Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Matthew Rubenstein wrote: I connect to a PSTN carrier over SIP which requires me to connect with a g729 codec. I'm using them for just robocalling: Asterisk server originates calls which play a prerecorded file. Can I pre-encode those stored files in g729 so they don't need to be encoded for each call? If so, do I need a g729 license for each call, or just a license for the preencoder? If the robocalls accept incoming DTMF, do I need g729 licenses for those calls? On Mon, 2007-01-08 at 04:08 -0700, [EMAIL PROTECTED] wrote: Date: Mon, 08 Jan 2007 13:47:39 +0800 From: Leo Ann Boon [EMAIL PROTECTED] Subject: Re: [asterisk-users] Some queries on g729 license. 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 Xue Liangliang wrote: Hi, all I am a pabx vendor from Singapore. Recently we are going to implement a failover solution for our customers using heartbeat, the asterisk server can failover perfectly, however the g729 codec canot work, because it is binded the mac address, we have bought two set of licenses, can you provide us some workaround for this scenario? It shouldn't be a problem
[asterisk-users] Where is this hilarious Allison Smith file? (Also: Interview with Allison)
Listen to the first 30 (or so) seconds of http://pod-serve.com/audiofile/filename/4353/interviews-podcast-allison-smith.mp3 There's a queue-related sound file recorded by Allison: Thank you for holding... You are the umm, you are SO FAR DOWN the list we won't get your call today, probably the Cleaning Lady will notice the phone off the hook at 11 PM and go 'HAL?'. I cannot find this file anywhere, despite thorough searching. Certainly not in the two usual big sound tarfiles. I have a great place for this file in my extensions.conf, no doubt. I know somebody on the list will embarrass me by pointing out the obvious location In any event, many Allison groupies will enjoy the interview which follows the 'Cleaning Lady' introduction. ___ --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
Re: [asterisk-users] Some queries on g729 license.
When I first noticed that this thread has over 20 messages i was sure it is interesting. When I read it I realized that I havn't noticed that Al Bochter has posted to it. Plain old stuff, just someone making sure to put a new twist on it. On 1/8/07, Juan Jose Comellas [EMAIL PROTECTED] wrote: The Intel IPP-based G.729 codec does work with AMD processors out of the box, both with the 32 bit and 64 bit versions. On Mon January 8 2007 19:31, Zoa wrote: I did some tests a long time ago and the speed was roughly the same. ( I think digium's was slightly faster). I think the IPP version also doesn't work on AMD out of the box. It's just 10$ a channel, that's not even worth the hassle of trying something else. Joachim Al Bochter wrote: Matthew I agree. I only know what I have told by others so I do need this input I have been told that Digum G729 is a big pain the the butt to get working with Asterisk and it is very hard on the CPU Keep in mind I have never used any Ver. of G 729 So tell me what you think. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Matthew Rubenstein wrote: All of which hassle and expense can be avoided by buying a license for Digium's codec, which is tested to work well with Asterisk (and might come with some support). And is pretty cheap per simul call. I wonder whether that per call means per codec instance, which could be multiple licenses on a single conference call, where multiple (even if not all) parties are getting de/encoded simultaneously. And whether there are other tools for editing (/mixing/transforming) g729 data, in realtime (streams) or not (files), and whether they require a license. Ideally sox or equivalent would work on g729, maybe with a codec plugin. On Mon, 2007-01-08 at 13:23 -0500, Paul wrote: First point to tackle in any case involving patent, copyright or trademark infringement is whether or not the infringing party would have been qualified to buy any usage rights at all. In a case where you license the Intel source(read the terms, it's not really that free), you would be applying for a license under some plan that includes certain minimum payments. Even if you wrote new source from scratch you would be in the same boat. Last time I looked at the plans, I didn't see anything with low minimums. So even if you wrote code from scratch and never used it on more than 6 channels, you might have done something that normally requires a large upfront payment. Use $10k as an example. In such a case owner of the patent might have an attorney initiate contact. If you are willing to communicate they might allow you to pay the minimum and be licensed. If you can't do that, they might offer a settlement where you stop using the codec and pay them some lesser amount. If the patent holder can easily prove the violation you might as well try to deal with them and get things settled fast. If you sell or give away the codec it is easier for them to dig up proof. If you have unhappy employees that might be the way they hear about the violation in the first place. Important consideration: Bankruptcy law generally excludes debts created by things like malicious or criminal acts. Matthew Rubenstein wrote: As far as I know, the g729 patent requires buying a license to operate any implementation of it, whether Digium's, Intel's, or any other. Digium is set up to collect royalties (perhaps at a favorable rate) as part of their license from the patent holder. I don't know about Intel or any other. Or what the mechanics are for enforcing the patent on someone who operates a codec without a license. On Mon, 2007-01-08 at 10:51 -0500, Al Bochter wrote: What about the free open source G729 Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Matthew Rubenstein wrote: I connect to a PSTN carrier over SIP which requires me to connect with a g729 codec. I'm using them for just robocalling: Asterisk server originates calls which play a prerecorded file. Can I pre-encode those stored files in g729 so they don't need to be encoded for each call? If so, do I need a g729 license for each call, or just a license for the preencoder? If the robocalls accept incoming DTMF, do I need g729 licenses for those calls? On Mon, 2007-01-08 at 04:08 -0700, [EMAIL PROTECTED] wrote: Date: Mon, 08 Jan 2007 13:47:39 +0800 From: Leo Ann Boon [EMAIL PROTECTED] Subject: Re: [asterisk-users] Some queries on g729 license. 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 Xue Liangliang wrote: Hi, all I am a pabx vendor from Singapore. Recently we are going to implement a
Re: [asterisk-users] OT:spa942 provisioning
Good luck dealing with Linksys on that http://voxilla.com/tools/device-configuration-wizard/certificate-authority-service-for-linksys-analog-voip-adaptors-808.html On 1/8/07, Benko [EMAIL PROTECTED] wrote: Hello! Sorry for the OT-thread, but i don't know where else too ask... Has anyone done http-provisioning of a Linksys SPA942 with client side ssl-authentication? Where do i get the CA from? I'm aware of the Sipura mass deployment howto on voip-info.org, but it doesn't cover the authentification part. Thanks Christian ___ --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 ___ --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
Re: SV: [asterisk-users] Manage 'full' log file
On 1/8/07, lenz [EMAIL PROTECTED] wrote: You know that if you rename an open Unix file, it will stay open - i.e. if you rename the logfile full to full.1, Asterisk will continue writing to full.1 thinking it was full. The logger rotate command forces all log files to be closed and reopened with their canonical names, so your file is actually rotated. Hope this helps l. CORRECT about UNIX files, INCORRECT about logger rotate command. CORRECTION: logger rotate does: 1. Closes the files 2. Renames them (actually rotating them) 3. Reopnes the canonical named files logger reload does effectively work as you described: 1. Closes files 2. Reopens canonical named files This is the command that should be used with logrotate, for example. EXAMPLE: (consider asterisk running and writing to messages file) # cd /var/log/asterisk # mv messages messages.old (asterisk still running and now writing to messages.old) (there is no file named messages) # /usr/sbin/asterisk -rx logger reload (asterisk closed messages.old and created a new messages file) Hope this is clear enough as its really late now... Please correct me if I'm wrong Cheers -- Ex Vito ___ --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
Re: [asterisk-users] Looking for toll free in Italy
I do not think that there are some company that offer a toll free number (Numero verde in italian) But contact on of these three providers http://www.eutelia.it/tlc/ http://www.unidata.it/ http://messagenet.it/ If they have one of these should be able to give to you See you On 1/8/07, CM Rahman [EMAIL PROTECTED] wrote: *Hi,* ** *I am looking for tollfree number in italy. Anybody providing that? Charge per minute? It will connect to my asterisk pbx box.* ** *Thanks* ** *CM* __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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 ___ --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
RE: [asterisk-users] snom 360 auto answer
Thankyou David,
It works for Linksys,but not for snom 360.
Do I need to change someting using web UI ?
--- Klaverstyn, David C
[EMAIL PROTECTED] wrote:
This is my code (that I copied form somewhere) for
paging a group of
phones. By dialling 99 it will page phones 2101,
2102 and 2105.
Just include the context ext-paging in your dial
plan and modify the
extension numbers and all should be good.
This works on Linksys Phones but should also work on
Snoms.
I hope this helps you.
[ext-paging]
exten = PAGE2101,1,GotoIf($[ ${CALLERID(number)} =
2101 ]?skipself)
exten = PAGE2101,n,Set(__SIPADDHEADER=Call-Info:
\;answer-after=0)
exten = PAGE2101,n,Set(__ALERT_INFO=Ring Answer)
exten =
PAGE2101,n,Set(__SIP_URI_OPTIONS=intercom=true)
exten = PAGE2101,n,Dial(SIP/2101,5)
exten = PAGE2101,n(skipself),Noop(Not paging
originator)
exten = PAGE2102,1,GotoIf($[ ${CALLERID(number)} =
2102 ]?skipself)
exten = PAGE2102,n,Set(__SIPADDHEADER=Call-Info:
\;answer-after=0)
exten = PAGE2102,n,Set(__ALERT_INFO=Ring Answer)
exten =
PAGE2102,n,Set(__SIP_URI_OPTIONS=intercom=true)
exten = PAGE2102,n,Dial(SIP/2102,5)
exten = PAGE2102,n(skipself),Noop(Not paging
originator)
exten = PAGE2105,1,GotoIf($[ ${CALLERID(number)} =
2105 ]?skipself)
exten = PAGE2105,n,Set(__SIPADDHEADER=Call-Info:
\;answer-after=0)
exten = PAGE2105,n,Set(__ALERT_INFO=Ring Answer)
exten =
PAGE2105,n,Set(__SIP_URI_OPTIONS=intercom=true)
exten = PAGE2105,n,Dial(SIP/2105,5)
exten = PAGE2105,n(skipself),Noop(Not paging
originator)
exten = Debug,1,Noop(dialstr is
LOCAL/[EMAIL PROTECTED]LOCAL/[EMAIL PROTECTED]LOCAL/[EMAIL PROTECTED]
aging)
exten =
99,1,Page(LOCAL/[EMAIL PROTECTED]LOCAL/[EMAIL PROTECTED]LOCAL/PAGE
[EMAIL PROTECTED])
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Behalf Of Jason Kim
Sent: Monday, 8 January 2007 2:30 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] snom 360 auto answer
Hi,
I'm testing paging using snom 360.
Can someone correct my dialplan?
Regards,
Jason.
==
;exten = _99,1,SIPAddHeader(Call-Info:
Answer-After=0)
;exten = _99,n,SIPAddHeader(Call-Info:
sip:192.168.1.113\;answer-after=0)
;exten = _99,n,Dial(SIP/${EXTEN:2})
exten = _99,1,Set(__SIPADDHEADER=Call-Info:
answer-after=0)
exten =
_99,n,Set(__SIP_URI_OPTIONS=intercom=true)
exten = _99,n,Set(__ALERT_INFO=Ring Answer)
exten = _99,n,Dial(SIP/${EXTEN:2})
__
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Re: [asterisk-users] Some queries on g729 license.
Jerry wrote: I believe (this may have changed) that ANY patented technology can be used for free educationally. The idea is that people can study and play with the technology for no charge. I'm not sure if this means that a University can use this in their phone system without paying the patent fees, though. Now, certainly there can be open source versions of the G.729 codec. They can even be free in the sense that the author is not charging. But the author can't waive the patent rights. Intel has a freely downloadable codec for educational use, but they have a long legalese document which explains the patent obligations. If you are using G.729 commercially, there is no question you have a legal obligation to pay the patent holder for his rights. Whether it's a university or a megacorporation studying the technology, they have to be very careful. Suppose we are working on automotive fuel economy or emissions improvement. If we buy a new or used car we are reasonably sure that a multitude of patents involved are being legally used. If we build a cadillac clone for the research, I would be worried. I can't find any refs to it now, so it might well be that the law has changed. It was very specifically for research purposes only, and may have been limited to educational institutions. I'll dig around and see if I can locate the reference. Another factor to consider in some cases is when we sign a sales contract that includes things like no reverse engineering. It might be hard to prove that we did not reverse engineer the product in order to develop a patentable improvement. Since patents require the disclosure of the novel process in order to be considered, reverse engineering or violating the DMCA (or whatever other law may apply) wouldn't have to be done. That's the beauty of patents, as opposed to keeping something a trade secret -- anyone can see what is being done. (Reverse engineering is generally done to inter operate, not to recover something patentable) J. ___ --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
Fwd: [asterisk-users] Some queries on g729 license.
I knew I was doing the right thing, here is the proof, enjoy when you read it, and have a good laugh. -- Forwarded message -- From: Al Bochter [EMAIL PROTECTED] Date: Jan 8, 2007 8:22 PM Subject: Re: [asterisk-users] Some queries on g729 license. To: [EMAIL PROTECTED] (C)UNT (F)UCK! THIS IS OFF THE LIST FUCK YOU ASSHOLE! GET A JOB AND STOP LIVING OFF MY TAXES YOU DON'T KNOW WHAT YOU ARE DOING TRY AND STAY ON THE POINT. YOU ARE NOW BLOCKED I AM NOT GOING TO DEAL WITH JACKASSES LIKE YOU GOOD BYE Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email C F wrote: When I first noticed that this thread has over 20 messages i was sure it is interesting. When I read it I realized that I havn't noticed that Al Bochter has posted to it. Plain old stuff, just someone making sure to put a new twist on it. On 1/8/07, Juan Jose Comellas [EMAIL PROTECTED] wrote: The Intel IPP-based G.729 codec does work with AMD processors out of the box, both with the 32 bit and 64 bit versions. On Mon January 8 2007 19:31, Zoa wrote: I did some tests a long time ago and the speed was roughly the same. ( I think digium's was slightly faster). I think the IPP version also doesn't work on AMD out of the box. It's just 10$ a channel, that's not even worth the hassle of trying something else. Joachim Al Bochter wrote: Matthew I agree. I only know what I have told by others so I do need this input I have been told that Digum G729 is a big pain the the butt to get working with Asterisk and it is very hard on the CPU Keep in mind I have never used any Ver. of G 729 So tell me what you think. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Matthew Rubenstein wrote: All of which hassle and expense can be avoided by buying a license for Digium's codec, which is tested to work well with Asterisk (and might come with some support). And is pretty cheap per simul call. I wonder whether that per call means per codec instance, which could be multiple licenses on a single conference call, where multiple (even if not all) parties are getting de/encoded simultaneously. And whether there are other tools for editing (/mixing/transforming) g729 data, in realtime (streams) or not (files), and whether they require a license. Ideally sox or equivalent would work on g729, maybe with a codec plugin. On Mon, 2007-01-08 at 13:23 -0500, Paul wrote: First point to tackle in any case involving patent, copyright or trademark infringement is whether or not the infringing party would have been qualified to buy any usage rights at all. In a case where you license the Intel source(read the terms, it's not really that free), you would be applying for a license under some plan that includes certain minimum payments. Even if you wrote new source from scratch you would be in the same boat. Last time I looked at the plans, I didn't see anything with low minimums. So even if you wrote code from scratch and never used it on more than 6 channels, you might have done something that normally requires a large upfront payment. Use $10k as an example. In such a case owner of the patent might have an attorney initiate contact. If you are willing to communicate they might allow you to pay the minimum and be licensed. If you can't do that, they might offer a settlement where you stop using the codec and pay them some lesser amount. If the patent holder can easily prove the violation you might as well try to deal with them and get things settled fast. If you sell or give away the codec it is easier for them to dig up proof. If you have unhappy employees that might be the way they hear about the violation in the first place. Important consideration: Bankruptcy law generally excludes debts created by things like malicious or criminal acts. Matthew Rubenstein wrote: As far as I know, the g729 patent requires buying a license to operate any implementation of it, whether Digium's, Intel's, or any other. Digium is set up to collect royalties (perhaps at a favorable rate) as part of their license from the patent holder. I don't know about Intel or any other. Or what the mechanics are for enforcing the patent on someone who operates a codec without a license. On Mon, 2007-01-08 at 10:51 -0500, Al Bochter wrote: What about the free open source G729 Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Matthew Rubenstein wrote: I connect to a PSTN carrier over SIP which requires me to connect with a g729 codec. I'm using them for just robocalling: Asterisk server originates calls which play a prerecorded file. Can I pre-encode those stored files in g729 so they don't need to be encoded for each call? If so, do I need a g729 license for each call, or just a license
