Re: [asterisk-dev] Asterisk Core Dumps
/path/to/src/asterisk/doc/README.backtrace Kohler, Jeffrey wrote: I was able to eventually figure it out. For anyone as linux unsavvy as myself: - # gdb /usr/sbin/asterisk /tmp/core.21713 - # bt Sorry for the stupid question From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kohler, Jeffrey Sent: Friday, July 21, 2006 8:21 AM To: Asterisk Developers Mailing List Subject: [asterisk-dev] Asterisk Core Dumps I've been experiencing some Asterisk crashes and was trying to look at the core dumps to figure out what is going wrong. Unfortunately I'm not really even sure where to start. (I'm a Windows developer by trade so please excuse my ignorance) Here's what I've done so far: - run Asterisk with the safe_asterisk script - when Asterisk crashed it produced a dump file - core.21713 - From a command line, I ran gdb - # core-file /tmp/core.21713 - # bt - I did get a stack trace, however, it simply lists a series of addresses. How can I get symbolic information to make sense of this? Is there a better method of examining core files? -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] Feedback / handoff request for AGI "BACKGROUND" command...
For what it's worth, http://www.voip-info.org/wiki-Asterisk+cmd+Background is completely wrong. If you look at the history, it used to be right, but JazEzork in revision 11 changed it to it's current misleading form.--Tim ___ --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] [RFC] optimising RTP traffic?
Title: asterisk-dev Digest, Vol 24, Issue 53 Summing it up about optimizing RTP traffic - I get that "sendfile" is off-limits"because one of the file descriptions can't be a socket. splice - while einticing, is only avaialbe in new Lunix kernels, also it lacks user space library support (I'll take this back if someone explains me how I can do a syscall without library support) This leaves epoll and the only candiate to boost RTP performance on asterisk. Did I get it right? Thanks -c ___ --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] iax.conf and dtmfmode
On Fri, 21 Jul 2006, Russell Bryant wrote: On Fri, 2006-07-21 at 12:31 -0400, Peter Beckman wrote: Is dtmfmode a valid configuration variable in iax.conf? If so, can I request it be added to the sample config? "dtmfmode" is absolutely not a valid option for chan_iax2. There is only one way to send DTMF in the IAX2 protocol. That's what I understood, now I know for sure. This gives way to a followup question. Why would digit I enter NOT be passed or detected from an IAX2 connection? My guess is call quality between the caller and the VOIP DID provider. Caller Phone (MaBell, Mobile phone) --> DID Provider --iax2--> Asterisk If the connection is bad between the cell phone and did provider, tones may not be "heard" correctly, right? Once within an iax2 connection, DTMF tones are almost impossible to NOT work, correct? Thanks Russell! Beckman --- Peter Beckman Internet Guy [EMAIL PROTECTED] http://www.purplecow.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
Re: [asterisk-dev] iax.conf and dtmfmode
On Fri, 2006-07-21 at 12:31 -0400, Peter Beckman wrote: > Is dtmfmode a valid configuration variable in iax.conf? If so, can I > request it be added to the sample config? "dtmfmode" is absolutely not a valid option for chan_iax2. There is only one way to send DTMF in the IAX2 protocol. -- Russell Bryant Software Developer Digium, Inc. ___ --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] Solaris zaptel
Title: Re: [asterisk-dev] Solaris zaptel Hello, The Zaptel source code is completely different from the current TRUNK. It would be a large undertaking to ensure the same code base compiled on all platforms. I believe this is why the FreeBSD drivers are separate as well. Basically, Solaris != Linux on multiple levels. So, for Zaptel, I'd like my own module like: zaptel-solaris/trunk For Asterisk, I'd like two things. I'd like a branch for Solaris using the stable release branch. This is because the PKGs offered and used should be from the stable branch. Why the branch? Well because it'll take some time to get TRUNK truly able to have Solaris squared away and I need to be able to build packages now. Past dealings with patch submissions for TRUNK have taken a long time to be merged and I really can not wait for those merges to be realized. I already have a disclaimer on file with Digium. Thoughts, ideas, or suggestions? -Joe On Fri, 2006-07-21 at 10:07 -0700, John Todd wrote: Joe - Can you be more specific about what you'd like to see for SVN hosting? The Digium (main Asterisk/Zaptel) SVN server would be an optimal place for this in the asterisk-extras directory, unless your patches can go directly into SVN TRUNK, which would be the ideal method. Of course, you'd need to sign the release and license agreement on http://bugs.digium.com/ (at bottom of page) and send it back to Digium. Can you start a bug in the bugtracker on this? I'm sure that there are many people (including myself) who would be very interested in seeing the Solaris Zap patches make their way into TRUNK. JT ___ --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] Feedback / handoff request for AGI "BACKGROUND" command...
On Thursday 20 July 2006 18:15, Josh McAllister wrote: > Out of immediate neccessity I hacked together a new AGI command to > stream a file in the background (exec Background doesn't seem to > work... Does not return immediately). This has been briefly tested, > and it does indeed work. Usage: The Background app isn't supposed to return immediately. The usage of the Background app is to play a file, while waiting for DTMF input. > BACKGROUND [file] > > Yes, file is optional. Calling BACKGROUND with no parameters invokes > ast_stopstream which cuts off the stream, and makes us ready to > stream something else. You CAN NOT stream anything else until you've > called BACKGROUND with no params, or bad things may happen. > > One feature I started to plan for, but didn't finish is for > BACKGROUND by itself to return the offset (endpos) of the > backgrounded stream where it was stopped. I would prefer if you didn't call it BACKGROUND, since the functionality you've described conflicts with the functionality of the Background app. Perhaps a better name would be ASYNCPLAYBACK? -- Tilghman ___ --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] Solaris zaptel
Title: Re: [asterisk-dev] Solaris zaptel The wctdm driver has been posted to http://www.SolarisVoip.com/ in addition to updates to zaptel and ztdummy. I have tested the wctdm driver with FXO and FXS modules. It seems to work well. The full change log is available on the site. I've had a ton of people asking and demanding the source code. Some people, shesh... I don't mind people wanting to help, but demanding the source code? That's not friendly or nice. :) Digium never answered my question, nor Sun's, about hosting the SVN for the project - thus the source code won't be released to the public until I get the time to get SVN installed on a new IP address. I'm confused as to why they never responded. Comments, suggestions, or ideas welcomed! Thanks for your time, -Joe Thralling Penguin LLC Joe - Can you be more specific about what you'd like to see for SVN hosting? The Digium (main Asterisk/Zaptel) SVN server would be an optimal place for this in the asterisk-extras directory, unless your patches can go directly into SVN TRUNK, which would be the ideal method. Of course, you'd need to sign the release and license agreement on http://bugs.digium.com/ (at bottom of page) and send it back to Digium. Can you start a bug in the bugtracker on this? I'm sure that there are many people (including myself) who would be very interested in seeing the Solaris Zap patches make their way into TRUNK. JT ___ --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] iax.conf and dtmfmode
Dear Supergeeks -- I am confused. I've been having problems with receiving DTMF tones via different providers (I use three for inbound DID calling; junction networks, binfone and Vitelity/SixTel/EXGN). Today I read the docs for SIP dtmfmode, added "dtmfmode=auto" in my iax.conf, reloaded, and my problem went away. Dumbfounded, I tried to find documentation supporting the fact that setting dtmfmode in iax.conf is a valid config variable, but I couldn't. A search of Doxygen shows on chan_iax2.c, line 2045 there is mention of a dtmfmode, but it looks hard-coded to me. I found references to dtmfmode in chan_mgcp and chan_sip but not much else. Is dtmfmode a valid configuration variable in iax.conf? If so, can I request it be added to the sample config? http://www.asterisk.org/doxygen/Config_iax.html Thanks! Beckman --- Peter Beckman Internet Guy [EMAIL PROTECTED] http://www.purplecow.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
RE: [asterisk-dev] Asterisk Core Dumps
I was able to eventually figure it out. For anyone as linux unsavvy as myself: - # gdb /usr/sbin/asterisk /tmp/core.21713 - # bt Sorry for the stupid question From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kohler, Jeffrey Sent: Friday, July 21, 2006 8:21 AM To: Asterisk Developers Mailing List Subject: [asterisk-dev] Asterisk Core Dumps I’ve been experiencing some Asterisk crashes and was trying to look at the core dumps to figure out what is going wrong. Unfortunately I’m not really even sure where to start. (I’m a Windows developer by trade so please excuse my ignorance) Here’s what I’ve done so far: - run Asterisk with the safe_asterisk script - when Asterisk crashed it produced a dump file - core.21713 - From a command line, I ran gdb - # core-file /tmp/core.21713 - # bt - I did get a stack trace, however, it simply lists a series of addresses. How can I get symbolic information to make sense of this? Is there a better method of examining core files? Thanks ___ --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] Voicemail - Failed to lock path....
Hey List! I'm working on using the Distribution List Voicemail patch back from Jan2005 (v1.0.6) and trying to integrate it into 1.2.9 (and eventually, maybe HEAD?). http://bugs.digium.com/bug_view_page.php?bug_id=0002729 I've got the majority of the bugs worked out (not including Mark's suggestions). However, after leaving a message, I can no longer access my mailbox. "app.c:1171 ast_lock_path: Failed to lock path '/var/spool/asterisk/voicemail/default/1001/INBOX: file exists Again, this is against 1.2.9. This is the first time I've worked on the app_voicemail.c, and I may not fully understand how it locks/unlocks. Can someone give me a clue as to why this is happening, or some direction to look at? Thanks!! Aaron Paxson ___ --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] About implementation of poll() in poll.c
On Jul 21, 2006, at 10:40 AM, Pranav Peshwe wrote: Hello, In chan_sip.c , ast_io_wait() uses the poll() function defined in poll.c. poll() internally uses select() for fd multiplexing. Why is poll() implemented using select() in asterisk inspite of being present as a standard function already i.e POLL(2) - on linux atleast ? The semantics of the implemented poll() function look similar to POLL(2). Is there any specific reason ? like portability problems for poll () ? or is the implementation in any way different in functionality from POLL(2) ? It seems the emulation is only used when the underlying system doesn't support poll() itself. -SteveK ___ --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] Feedback / handoff request for AGI "BACKGROUND" command...
On Thu, 2006-07-20 at 16:15 -0700, Josh McAllister wrote: > Out of immediate neccessity I hacked together a new AGI command to > stream a file in the background (exec Background doesn't seem to work... > Does not return immediately). This has been briefly tested, and it does > indeed work. Usage: Please don't paste code to the mailing list -- post it on the bug tracker instead. You can find it at http://bugs.digium.com, along with submission guidelines and a disclaimer you'll need to fill out and send to Digium. -Jared ___ --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] About implementation of poll() in poll.c
Hello, In chan_sip.c , ast_io_wait() uses the poll() function defined in poll.c. poll() internally uses select() for fd multiplexing. Why is poll() implemented using select() in asterisk inspite of being present as a standard function already i.e POLL(2) - on linux atleast ? The semantics of the implemented poll() function look similar to POLL(2). Is there any specific reason ? like portability problems for poll() ? or is the implementation in any way different in functionality from POLL(2) ? TIA. Regards, Pranav ___ --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] RE: Sphinx and asterisk
Hello Kiran, Thanks for your message. It is fairly easy to integrate Asterisk with Sphinx, the only trouble is that you need to have an Acoustic Model (AM) for 8KHz, which are not (yet) readily available. There is a Language Model (LM) and AM included with Sphinx, but it is designed for a sampling rate of 16KHz and therefore does not work with Asterisk. Even if you use sophisticated upsampling techniques (sinc interpolation - sox) to create 16KHz for use with Sphinx, the recognition rate is absolutely dismal. Sphinx does a fine job on native 16KHz samples, just not samples created by upsampling. CMU does have an 8KHz LM and AM available which they created for their Communicator project (phone based airplane reservation system), but the AM is for Sphinx2, which has considerably poorer recognition speed and accuracy compared to Sphinx3 or Sphinx4. I haven't had a chance to try and convert the AM to Sphinx3, but I believe it can be done. If you want to spend the next 6 months developing and training a LM and AM for Sphinx for 8KHz, many many people will be very happy :) Regards, Stephan. --- Stephan A. Edelman, B.Eng. NewAce Corporation Toll Free: 1-877-463-9223 x221 Tel: +1 519 336 4837 x221 Fax: +1 519 336 4046 Pager: +1 519 333 3247 From: kiranSent: Fri 7/21/2006 8:31 AMTo: [EMAIL PROTECTED]Subject: Sphinx and asterisk Dear Stephan, Please let me know if you can help us in integrating sphinx with asterisk Regards Kiran ___ --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] Re: [RFC] optimisinf RTP traffic? (Roy Sigurd Karlsbakk)
On Fri, 2006-07-21 at 14:42 +0800, Dinesh Nair wrote: > On 07/21/06 01:48 Benny Amorsen said the following: > > If you use splice() instead of sendfile(), you can get the same > > splice() is not available on FreeBSD, and will hinder porting work there. > also, iianm, splice() needs one of its file descriptor arguments to be a > pipe, while in this context we're discussing two sockets. While reading the man page of sendfile() it says that the input file descriptor cannot be a socket. Jeff signature.asc Description: This is a digitally signed message part ___ --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
