Re: [Asterisk-Dev] no mutex_assert call ?
On Dec 20, 2005, at 8:31 AM, Kevin P. Fleming wrote: That was my presentation from Astricon Fall 2005, and it's available on the Astricon web site. At one time that required a username/password (only for Astricon attendees), but I don't know if that is still true. The one from Astricon Europe is on digium.com: http://www.digium.com/downloads/AstriconEurope2005Tutorial.pdf Did you put the updated presentation on digium.com or asterisk.org? -- Russell Bryant ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[Asterisk-Dev] default value of ast_opt_priority_jumping
As we all already know, the n+101 priority jumping behavior of applications is being deprecated. For Asterisk 1.2, we made the default value of the global priority jumping option to be on. However, if it was a new installation, extensions.conf.sample has it turned off. I think it's time to make it off by default in the code in the trunk. Agree? -- Russell Bryant ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[Asterisk-Dev] meetme: codec_gsm.c errors when using user/admin menu
I'm getting the following errors when using the meetme user or admin
menu. Has anyone else seen this (or can reproduce it)? The following
is the sequence that starts this:
(get two people into conference)
Press * (for user or admin menu)
Press 8, 4, or 6
The following messages will start repeatedly streaming out on the console:
Dec 20 13:39:45 WARNING[29627]: codec_gsm.c:165 gsmtolin_framein: Huh? A GSM
frame that isn't a multiple of 33 or 65 bytes long from (null) (320)?
In addition, these keypresses do not stop the menu from being read (Allison
still continues to read the menu options).
On the admin menu with only one person in the conference, doing these
commands causes any further sound to mute along with the above warning
messages, which can't seem to be stopped without terminating the conference.
What seems to be happening is that the ast_streamfile is not stopped
after a key is pressed. When I change (in both places it occurs,
this one is for the admin menu) the following code:
if (!ast_streamfile(chan, conf-adminmenu, chan-language))
dtmf = ast_waitstream(chan, AST_DIGIT_ANY);
else
dtmf = 0;
-to-
if (!ast_streamfile(chan, conf-adminmenu, chan-language)) {
dtmf = ast_waitstream(chan, AST_DIGIT_ANY);
ast_stopstream(chan);
} else
dtmf = 0;
This appears to fix the trouble. It looks like the menu streaming
gets intermixed with the conference audio and confuses the GSM codec
(since the audio files that come with Asterisk are in GSM format).
My questions are:
(1) Is anyone else seeing this behavior in meetme?
(2) If so, does this fix look like the right one?
Here are the appropriate extensions.conf lines:
; Test conferencing user (657)
exten = 657,1,Answer
exten = 657,n,MeetMe(,iMs)
exten = 657,n,Hangup
; Test conferencing admin (658)
exten = 658,1,Answer
exten = 658,n,MeetMe(1234,aDiMs)
exten = 658,n,Hangup
This is Asterisk SVN-branch-1.2-r7351M on Debian, kernel 2.4.27.10,
zaptel as of 7-Dec-2005, TE410P card.
Thanks in advance for your feedback. Because of my last blunder, I
wanted to make sure I got someone else to review this first.
---
Gil Kloepfer
[EMAIL PROTECTED]
___
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[Asterisk-Dev] Called Number problem
Hello Guys, I couple of weeks ago we encountered a problem with two customers calling from abroad. The called number was received twice, at least from a certain point it became double. So every time they called (lets assume our number is ABCD) asterisk tried to jump to an extension ABCDABCD. We had a hard time digging around in the code and reading q931 specs trying to track down the reason for this. We assumed that this is an chan_zap.c issue, but after a time we tracked it down to q931.c from libpri. It seems that when the initialisation for the INFORMATION frame is made (actually there is none atm) the c-callednum stays the old one from the SETUP frame and is then again copied to pri-ev.ring.callednum even if the INFORMATION frame only contains a Sending Complete flag and no additional number. So we added a c-callednum[0] = '\0'; and now it works perfectly. If that the case I would propose to add this to the source. The diff is attached. Would be happy to get some feedback about this. Thanks in adavance. Bye, Dimitri Machen Sie aus 14 Cent spielend bis zu 100 Euro! Die neue Gaming-Area von Arcor - über 50 Onlinespiele im Angebot. http://www.arcor.de/rd/emf-gaming-1 q931_patch.diff Description: Binary data ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] default value of ast_opt_priority_jumping
On 12/20/05, Russell Bryant [EMAIL PROTECTED] wrote: As we all already know, the n+101 priority jumping behavior of applications is being deprecated. For Asterisk 1.2, we made the default value of the global priority jumping option to be on. However, if it was a new installation, extensions.conf.sample has it turned off. I think it's time to make it off by default in the code in the trunk. Agree? Agree. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[Asterisk-Dev] Asterisk::LDAP update
Hello All About six months ago I wrote to announce the release of my Perl module to manage Asterisk configuration in LDAP. I wanted to take a moment and send this follow-up to announce a newly updated version of Asterisk::LDAP, version 0.6.0. This new release is a significant improvement over the 0.5 series. Here's a quick summary of the new features: * Serial Numbers: Contexts are now written with a serial number and are not updated unless the serial number is incremented. This allows for granular and guaranteed consistent updates. Because LDAP updates are atomic the administrator or any external dialplan manager needs only to update the serial number after making any changes to the dialplan to guarantee the new dialplan will be loaded in its entirety. * API Simplicity: As mentioned above, the API has been dramatically cleaned up. Where before a number of calls were required to set up Asterisk::LDAP before getting any useful data out, only one call is required and three more optional calls can help the developer automate much of the configuration of the module. Also the internal data structures are made available to external programs at more points during the configuration generation so any kind of hooks or post-processing can be done more easily. * New Output Formats: The developer may now choose to have Asterisk::LDAP simply read the information from LDAP or have it write the contents to a set of files. Future improvements may include support for returning a string containing the entire data output instead of files written to disk. To find out more about this project or to download the new release please visit the pages at http://projects.alkaloid.net. Thanks again, /BAK/ -- Ben Klang [EMAIL PROTECTED] Alkaloid Networks http://projects.alkaloid.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] Asterisk::LDAP update
Ben Klang wrote: To find out more about this project or to download the new release please visit the pages at http://projects.alkaloid.net. Good work, but hopefully you support more than the obsolete v1.0 branch, as your website documents. Jeremy McNamara ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] no mutex_assert call ?
Russell Bryant wrote: Did you put the updated presentation on digium.com or asterisk.org? Neither; the Astricon people posted it on their site. I suppose at this point it would be safe to post it publicly, but I'll have to check with them first. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] default value of ast_opt_priority_jumping
BJ Weschke wrote: I think it's time to make it off by default in the code in the trunk. Agree? Yep. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] Asterisk::LDAP update
On Tue, 2005-12-20 at 23:42 -0500, Jeremy McNamara wrote: Ben Klang wrote: To find out more about this project or to download the new release please visit the pages at http://projects.alkaloid.net. Good work, but hopefully you support more than the obsolete v1.0 branch, as your website documents. Since I do not have 1.2 anywhere in production yet I have not coded to that. To be honest I haven't even looked into the differences. I will certainly add support for 1.2 as time permits and demand allows. /BAK/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] Asterisk::LDAP update
Ben Klang wrote: On Tue, 2005-12-20 at 23:42 -0500, Jeremy McNamara wrote: Ben Klang wrote: To find out more about this project or to download the new release please visit the pages at http://projects.alkaloid.net. Good work, but hopefully you support more than the obsolete v1.0 branch, as your website documents. Since I do not have 1.2 anywhere in production yet I have not coded to that. To be honest I haven't even looked into the differences. I will certainly add support for 1.2 as time permits and demand allows. /BAK/ As a datapoint from this quarter, until it supports 1.2 it won't be of any use to me; that's all I run. I know there are a lot of 1.0.x systems still in production (hell, I've got a friend with a pre-1.0 system still up somewhere out there. . ) but 1.2 is such a radical improvement, IMO it would be well worth it for the eventual acceptance of the product (if in fact that's a goal) for you to investigate the port. Thanks. B. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
