Re: [asterisk-users] issues.asterisk.org/jira not working
A number of people are reporting that Safari is not working properly with JIRA. Use Firefox or Chrome for now. -- Russell Bryant Digium, Inc. | Engineering Manager, Open Source Software 445 Jan Davis Drive NW- Huntsville, AL 35806 - USA www.digium.com -=- www.asterisk.org -=- blogs.asterisk.org - Original Message - I get this on my Mac: Safari can’t open the page. Safari can’t open the page “https://issues.asterisk.org/jira/browse/ASTERISK-17984” because Safari can’t establish a secure connection to the server “issues.asterisk.org”. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jun 8, 2011, at 11:38 AM, William Stillwell wrote: You mean this one? https://issues.asterisk.org/jira/browse/ASTERISK-17984 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Wednesday, June 08, 2011 2:17 PM To: asterisk-users Subject: [asterisk-users] issues.asterisk.org/jira not working Bad day today. Why this new JIRA system not working. I have created issue and submit and i got blank page.. Please someone help me to create BUG!!! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] issues.asterisk.org
On 06/06/2011 12:08 AM, Jeremy Kister wrote: i'm trying to review issues that i'm monitoring and/or have reported at http://issues.asterisk.org when I click on 'My View' or 'View Issues' I get an error: APPLICATION ERROR #401 Database query failed. Error received from database was #1142: DELETE command denied to user 'mantisreadonly'@'localhost' for table 'mantis_tokens_table' for the query: DELETE FROM mantis_tokens_table WHERE '2011-06-06 00:03:56' expiry. Please try this again. Are tickets that I had set up for monitoring on mantis going to be automatically monitored in jira ? No. We migrated as much as we could. This was one minor thing that was not migrated over. -- Russell Bryant Digium, Inc. | Engineering Manager, Open Source Software 445 Jan Davis Drive NW- Huntsville, AL 35806 - USA www.digium.com -=- www.asterisk.org -=- blogs.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Migration from Mantis to JIRA
On 06/02/2011 06:46 AM, Terry Brummell wrote: We use Jira at work. I hate it. Hope you have a better experience than I've had! We've been using it for years internally to Digium. We've been happy with it. -- Russell Bryant Digium, Inc. | Engineering Manager, Open Source Software 445 Jan Davis Drive NW- Huntsville, AL 35806 - USA www.digium.com -=- www.asterisk.org -=- blogs.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Migration from Mantis to JIRA
Greetings, A few weeks ago I posted a message about the upcoming migration from Mantis to JIRA for issues.asterisk.org [1]. A lot of testing has been done and all known issues have been resolved. We have scheduled the migration for Sunday, June 5th. The issue tracker will be down most of the day as the migration takes place. Once the migration is complete, the issue tracker will be: https://issues.asterisk.org/jira/ Mantis will still be available for some time, but will be read-only. If you have an account on Mantis, you will be able to log in to JIRA using the same username. All of your history will have been migrated. This account can also be used on wiki.asterisk.org. IMPORTANT NOTE: You will have to click the forgot my password link to reset your password before you can log in, though. It is not possible to migrate passwords from one to the other as they use a different hashing algorithm. For more information about how to use JIRA, see the JIRA user's guide: http://confluence.atlassian.com/display/JIRA042/JIRA+User%27s+Guide If you run into any problems after the migration has taken place, please report them in the JIRA Help project. If you would rather report something via email, email espiceland at digium dot com and me. Thanks, [1] http://lists.digium.com/pipermail/asterisk-dev/2011-May/049088.html -- Russell Bryant Digium, Inc. | Engineering Manager, Open Source Software 445 Jan Davis Drive NW- Huntsville, AL 35806 - USA www.digium.com -=- www.asterisk.org -=- blogs.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AJAM XML output not valid xml
- Original Message - On Mon, 2011-05-23 at 15:41 +0100, Ishfaq Malik wrote: Hi I'm using asterisk 1.8.3.2 and have been implementing AJAM. I've noticed the final '' is missing from every response I've had so far. Here is an example ajax-response response type='object' id='unknown'generic response='Success' message='Authentication accepted' //response /ajax-response Has anyone else noticed this? Is it a bug in the code or possibly a config setting I've missed? Thanks Ish Can someone else please verify that this is happening to them as well and if so I'll raise the issue... Try the latest code from the 1.8 branch. This sounds very familiar. I think it has already been fixed. $ svn co http://svn.asterisk.org/svn/asterisk/branches/1.8 asterisk-1.8 -- Russell Bryant Digium, Inc. | Engineering Manager, Open Source Software 445 Jan Davis Drive NW- Huntsville, AL 35806 - USA www.digium.com -=- www.asterisk.org -=- blogs.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Discussion: 1.8 quality issues
- Original Message - 1) We have adopted peer code reviews as common practice for all non-trivial changes going into Asterisk. This alone has _greatly_ increased the quality of the code going in. It is rare that a patch goes up for review where someone doesn't point out some sort of problem. These problems are found and fixed _much_ faster in the up front review process than if it had been many months later when someone encountered it as a bug in the field. Agree. But it also puts a significant delay on the process. We have to be very careful about that. Having too many branches open in addition to this was a pain. With fewer branches I hope it will get better. Fewer branches should help, but the fact the bar is raised on getting patches in due to the peer code review process is no different. There will always be problems with the code developers write. I view it as if there is a problem in the code, it is _much_ less expensive to get it resolved in up front peer review as much as possible than later on once users encounter a bug, report it, developers debug, fix, and test. That's the tradeoff. -- Russell Bryant Digium, Inc. | Engineering Manager, Open Source Software 445 Jan Davis Drive NW- Huntsville, AL 35806 - USA www.digium.com -=- www.asterisk.org -=- blogs.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?
- Original Message - PS. Please don't start a discussion about 1.8 quality in this thread, that's a separate issue. I just want to know what you think about closing 1.4 support now. If you want to discuss 1.8 quality, start a new thread. Thanks. I don't think it's a separate issue at all. I would like to see discussion of exactly which issues are preventing users from using Asterisk 1.8. We're trying to shift focus to those issues and get them resolved as quickly and as efficiently as we can so that we can all move forward. Resources are limited. What is the best use of our time to help ensure the best future? Where do we want to see the project in the next 6 months to a year? A primary focus on further solidifying Asterisk 1.8 is what gets us there in my mind. Asterisk 1.4 was released 4.5 years ago. It mostly just works, and I fully expect many to keep using it until they see a need to migrate. This process has been likened to when the community moved from Asterisk 1.2 to 1.4. Asterisk 1.8 has been much more stable out of the gate than 1.4, due to many things we have done over the years to increase quality, including: 1) We have adopted peer code reviews as common practice for all non-trivial changes going into Asterisk. This alone has _greatly_ increased the quality of the code going in. It is rare that a patch goes up for review where someone doesn't point out some sort of problem. These problems are found and fixed _much_ faster in the up front review process than if it had been many months later when someone encountered it as a bug in the field. 2) We have placed an increased emphasis on automated testing efforts. In addition to building up a lot of test environments inside of Digium, there is now an open source automated testing effort for Asterisk. There are over 200 test cases that run every time anyone touches the code. This includes complex call scenarios such as transfers and call parking. These open source test cases touch about 25% of the code (and what it does touch are things we considered some of the most important parts). That is a huge step forward from where we started. We are continuing to place more and more resources on this effort to move it forward. Despite comments in this thread, there _are_ many people using Asterisk 1.8 in production, including large installations. The ones with systems working perfectly fine don't tend to make as much noise. :-) For those still getting hit by problems, I hope that you can make the time to report them so that we can work with you to get them resolved. I started my work on Asterisk as a volunteer 7 years ago and even though it is now my full time job, I still put many personal hours into the project. I care very deeply about the success of Asterisk. I truly believe that the steps we have taken with release management are in the best interest of the project. Thanks, -- Russell Bryant Digium, Inc. | Engineering Manager, Open Source Software 445 Jan Davis Drive NW- Huntsville, AL 35806 - USA www.digium.com -=- www.asterisk.org -=- blogs.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?
- Original Message - On Apr 28, 2011, at 9:53 AM, Russell Bryant wrote: I don't think it's a separate issue at all. I would like to see discussion of exactly which issues are preventing users from using Asterisk 1.8. We're trying to shift focus to those issues and get them resolved as quickly and as efficiently as we can so that we can all move forward. For us the biggest issue is multi-tenant parking not working. We've really given up testing anything beyond that point because without that feature there really isn't any way we could use it. Broken as compared to 1.6.2? I ask since that feature wasn't in 1.4. Can you point to a bug report? I'd like to understand better what's not working. -- Russell Bryant Digium, Inc. | Engineering Manager, Open Source Software 445 Jan Davis Drive NW- Huntsville, AL 35806 - USA www.digium.com -=- www.asterisk.org -=- blogs.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?
- Original Message - On Apr 28, 2011, at 10:21 AM, Russell Bryant wrote: For us the biggest issue is multi-tenant parking not working. We've really given up testing anything beyond that point because without that feature there really isn't any way we could use it. Broken as compared to 1.6.2? I ask since that feature wasn't in 1.4. As compared to 1.6.1.x. We were using it precisely because we had to have multi-tenant parking. Can you point to a bug report? I'd like to understand better what's not working. https://issues.asterisk.org/view.php?id=18553 Basically for several versions of 1.6.2.x and all 1.8.x that we've tested, when you park a call it gets parked in the first parking lot regardless of what context the call is in when it is parked. Thanks! I will take a look at this one and see what we can do. -- Russell Bryant Digium, Inc. | Engineering Manager, Open Source Software 445 Jan Davis Drive NW- Huntsville, AL 35806 - USA www.digium.com -=- www.asterisk.org -=- blogs.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?
- Original Message - Sure. Please follow the 2 next stories: - had a customer running 1.4.26 We upgraded to a new server and installed 1.4.39, last version at this time. Bang: voicemail doesn't work as it should, had to fallback to 1.4.26 Customer is still running this version. - have 1.4.41 and 1.6.16 which are no more able to use auth keys in iax since we update one server from 1.4 to 1.6 Now imagine that 1.4 stays at only security level. For first case we have 2 options: upgrading for security reasons to last version but then no more voicemail, or staying with 1.4.26. In the second case, upgrading both servers to test with 1.8. If it's still not working, it was time loose beside other problems. If there are obvious regressions in major functionality such as voicemail, I'm more than happy to still consider making fixes for those problems during the security maintenance period. It has to be pretty clear, though, and in this particular case, it is. Can you point to the bug number please? I want to make sure this voicemail problem is resolved as soon as possible. -- Russell Bryant Digium, Inc. | Engineering Manager, Open Source Software 445 Jan Davis Drive NW- Huntsville, AL 35806 - USA www.digium.com -=- www.asterisk.org -=- blogs.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?
- Original Message - Thanks Matt. There seems to be an unresolved deadlock since the birth of 1.8. Using the most basic feature of a PBX, try to pickup some elses ringing extension - DEADLOCK. But I'm on to it, https://issues.asterisk.org/view.php?id=18654 and it's more uptodate review https://reviewboard.asterisk.org/r/1185/ Thanks, Alec. I have added this to the roadmap for the next 1.8 update. I'll make sure it gets resolved before then. -- Russell Bryant Digium, Inc. | Engineering Manager, Open Source Software 445 Jan Davis Drive NW- Huntsville, AL 35806 - USA www.digium.com -=- www.asterisk.org -=- blogs.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?
- Original Message - I would comment that I've been complaining about this since RC1 or 2 and if you just fixed it in 2 hours that there is something seriously wrong with the bug tracking system. I mean, I reported it a long time ago and while it was probably not the best bug report ever, I would have been more than willing to do almost anything to help fix it. I know what beta tester means, I've beta tested disk defraggers and disk caches and lost everything when they had the wrong bug and I know it can take a few tries to both fix the bug and for someone to help me identify it so they have an idea of where to look. Personally I'd just assumed that 1.8 was going to stay broken as no one seemed to care and was really happy when trunk worked as that meant I could move on. I like the bleeding edge and will always run the current beta on my small system unless I find a problem. I know it's dangerous, but it gives me the best chance of influencing where the product is going. Not much chance with this, but old habits die hard. I don't think there's anything inherently wrong with the bug tracking system. It's more of a resource issue with many conflicting priorities. Officially letting off some of the pressure from older branches does help. I would like to be making faster progress through bug reports and patches. I do have an open position for another full time Asterisk developer at Digium in case anyone is interested. :-) -- Russell Bryant Digium, Inc. | Engineering Manager, Open Source Software 445 Jan Davis Drive NW- Huntsville, AL 35806 - USA www.digium.com -=- www.asterisk.org -=- blogs.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 Packages for Debian and Ubuntu
- Original Message - Thanks for providing these - can you just clarify your policy on the following: - file locations - same layout as the regular Debian packages? Yes, same layout. - upgrade policy - is it intended that someone who has Debian 6 with the existing Asterisk 1.6 packages (from Debian's maintainer) can just upgrade to the Digium package without moving or changing any config? There is nothing specific about the packages that is going to make this situation any better or worse than any method of upgrading from Asterisk 1.6.X to Asterisk 1.8. Issues related to version compatibility can be found in the UPGRADE*.txt files in the Asterisk source. http://svn.asterisk.org/view/asterisk/trunk/UPGRADE-1.8.txt?view=markup -- Russell Bryant Digium, Inc. | Engineering Manager, Open Source Software 445 Jan Davis Drive NW- Huntsville, AL 35806 - USA www.digium.com -=- www.asterisk.org -=- blogs.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.8 Packages for Debian and Ubuntu
Greetings, Digium has been providing rpm packages for the latest versions of Asterisk that are compatible with RHEL 5 and CentOS 5 for quite some time now. We are pleased to announce that we will now be providing deb packages for both Debian and Ubuntu. As of now, we have Asterisk 1.8 packages available for the following distribution versions: * Debian 6.0 (squeezy) * Ubuntu 10.04 (lucid) * Ubuntu 10.10 (maverick) * Ubuntu 11.04 (natty) This effort is not intended to replace packaging of Asterisk in the official Debian or Ubuntu repositories. Our repositories are for providing access to major versions of Asterisk that are newer than what is included. We are exploring ways to work as closely as possible with the Debian and Ubuntu package maintainers to ensure that we do not duplicate efforts and that we provide the best possible result for users of Asterisk. For information on how to set up your system to use our repositories, please refer to the following page on the Asterisk wiki: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages If you have any problems related to the repositories or the packages themselves, please report them in the AsteriskNOW and Packages project on the Asterisk issue tracker, http://issues.asterisk.org/. Thanks, -- Russell Bryant Digium, Inc. | Engineering Manager, Open Source Software 445 Jan Davis Drive NW- Huntsville, AL 35806 - USA www.digium.com -=- www.asterisk.org -=- blogs.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Erroneous email from JIRA
Greetings, If you have an account on issues.asterisk.org and just received an email about having an account created on a JIRA server, please ignore and accept our apologies for the accidental message. We were doing some test migrations of data from issues.asterisk.org into a newer issue tracking system and forgot to disable email first. Thanks, -- Russell Bryant Digium, Inc. | Engineering Manager, Open Source Software 445 Jan Davis Drive NW -Huntsville, AL 35806 - USA jabber: rbry...@digium.com-=-skype: russell-bryant www.digium.com -=- www.asterisk.org -=- blogs.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DISA problem in 1.8.0
- Original Message - When I call into my Asterisk box via my VoIP line (using gsm codec) and then try to make an outgoing DISA call over PSTN I get the following: [Nov 1 15:12:54] WARNING[17694]: chan_dahdi.c:8930 dahdi_write: Cannot handle frames in gsm format [Nov 1 15:12:54] WARNING[17694]: app_dial.c:1401 wait_for_answer: Unable to forward voice or dtmf Obviously, it looks like asterisk is not converting the gsm frames to whatever it needs to send over the PSTN. I never had this problem with the 1.6.x series but it started as soon as I upgraded to 1.8.0 and dahdi-2.4.0. My Asterisk machine has a TDM-410 card installed for the interface to the PSTN. Any ideas? It just looks like a bug to me. Please report it on https://issues.asterisk.org/. Please include a call log, as well as the relevant configuration files so that we can try to reproduce the problem. Thanks! -- Russell Bryant Digium, Inc. | Engineering Manager, Open Source Software 445 Jan Davis Drive NW -Huntsville, AL 35806 - USA jabber: rbry...@digium.com-=-skype: russell-bryant www.digium.com -=- www.asterisk.org -=- blogs.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Remote Unix Connection
- Original Message - Sounds like either FreePBX or some other script using astmanproxy or just the AMI in general. Another possible cause is a script (or terminal) constantly accessing asterisk -r or rasterisk (+ any other arguments) to either run an Asterisk CLI command, or to just watch' the console output. It's asterisk -r or asterisk -rx. The message that says remote unix connection means a remote connection to Asterisk over the UNIX domain socket, which is what the remote console uses. AMI connections have a different message associated with them. -- Russell Bryant Digium, Inc. | Engineering Manager, Open Source Software 445 Jan Davis Drive NW -Huntsville, AL 35806 - USA jabber: rbry...@digium.com-=-skype: russell-bryant www.digium.com -=- www.asterisk.org -=- blogs.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.0-beta1 is Now Available!
- Original Message - Well, looking at messages turned up these, maybe it will help? WARNING[28505] loader.c: Error loading module 'chan_dahdi.so': /usr/lib/asterisk/modules/chan_dahdi.so: undefined symbol: ast_smdi_interface_unref WARNING[28505] loader.c: Error loading module 'app_stack.so': /usr/lib/asterisk/modules/app_stack.so: undefined symbol: ast_agi_unregister WARNING[28505] loader.c: Error loading module 'func_aes.so': /usr/lib/asterisk/modules/func_aes.so: undefined symbol: ast_aes_set_decrypt_key WARNING[28505] loader.c: Error loading module 'app_voicemail.so': /usr/lib/asterisk/modules/app_voicemail.so: undefined symbol: ast_smdi_mwi_message_destroy Thank you for posting information about the problem that you hit. These look related to some changes in 1.8 that were made to make modules that were previously hard requirements optional. For example, chan_dahdi required res_smdi, but now it's supposed to work regardless of whether or not it is loaded. With those changes in place, you are _never_ supposed to get these errors. Please open an issue on http://issues.asterisk.org/ for this problem. Please include your config.log file, as well as what distro you are using, and the version of gcc that you have (gcc --version). Thanks again for the feedback, -- Russell Bryant Digium, Inc. | Engineering Manager, Open Source Software 445 Jan Davis Drive NW -Huntsville, AL 35806 - USA jabber: rbry...@digium.com-=-skype: russell-bryant www.digium.com -=- www.asterisk.org -=- blogs.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Chanspy - Meetme
- Original Message - On 07/12/2010 05:36 PM, Xavier wrote: I've got a question about chanspy and meetme. I'd like to transfer all the persons involved in a chanspy (the guy spying, the guy that is spied and the guy that is speaking to the spied one - total: 3) in a conference room. Is there a way to do it quickly without especially knowing each channels ? It's a bit tricky to know and remember each channels, no ? You may not need to do this at all. ChanSpy (in Asterisk 1.6.2, at least) has a barge mode that allows the spying channel to speak to both parties. There is also the ability to enable DTMF key presses to swap between spy, whisper, and barge modes. -- Russell Bryant Digium, Inc. | Engineering Manager, Open Source Software 445 Jan Davis Drive NW -Huntsville, AL 35806 - USA jabber: rbry...@digium.com-=-skype: russell-bryant www.digium.com -=- www.asterisk.org -=- blogs.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6.2: Using hints on multiple parking lots
- Original Message - How do I specify to which parking lot the hints refer to? For exemple, on 1.4 I use this to see whether a call is parked in 800: exten = 800,hint,park:8...@parkedcalls But on 1.6 I have multiple parking lots working apparently sucessfully. How do I build the hint for parkinglot1 and parkingloit2 so that my phone , which is subscribing to 800, only see parkinglot1 and NOT parkinglot2? I tried the obvious answer exten = 800,hint,park:8...@parkinglot1 but that didnt seem to do anything. Instead of parkinglot1, use the name of the configured context for that parking lot. For example, if you set up the parking lot in features.conf as: [parkinglot1] context = parkedcalls_custom parkpos=800-850 the hint should be: exten = 800,hint,park:8...@parkedcalls_custom -- Russell Bryant Digium, Inc. | Engineering Manager, Open Source Software 445 Jan Davis Drive NW -Huntsville, AL 35806 - USA jabber: rbry...@digium.com-=-skype: russell-bryant www.digium.com -=- www.asterisk.org -=- blogs.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Remote-Party-ID party=called
On Mon, 2010-07-12 at 10:48 -0400, unsero...@aol.com wrote: No, the receiving side shows name and number as it should. But as calling person I only see the number of the called person instead of name and number. So we seem to struggle with the same issue. This is something that is not supported in any current version of Asterisk. However, a large amount of work has gone into connected party ID support which will be included in Asterisk 1.8. I expect the first beta of 1.8 to be available this month. -- Russell Bryant Digium, Inc. | Engineering Manager, Open Source Software 445 Jan Davis Drive NW -Huntsville, AL 35806 - USA jabber: rbry...@digium.com-=-skype: russell-bryant www.digium.com -=- www.asterisk.org -=- blogs.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Definite app_jack trouble - unsolvable
- Original Message - Hello Motiejus! My jack setup is fine. I'm a musician, so I use it regularly for much more demanding tasks than simple one-channel I/O. I just installed asterisk 1.6.2.9-rc1, it looked newer and fix-richer than 1.6.2.8. So any other ideas? Do you remember, when you had these problems? Perhaps the asterisk version you used? The app_jack code hasn't changed in a long time. I doubt a different version is going to help. I know you posted a bug about this a long time ago, and I'm sorry I haven't worked on it. I just haven't had time to work on this module again. It was something I had done in my free time. I got it working for me and never got back to it to address problems other people have reported. Hopefully another developer will take interest and try to help. -- Russell Bryant Digium, Inc. | Engineering Manager, Open Source Software 445 Jan Davis Drive NW -Huntsville, AL 35806 - USA jabber: rbry...@digium.com-=-skype: russell-bryant www.digium.com -=- www.asterisk.org -=- blogs.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Release Time Frames
Greetings, Asterisk 1.6.2.0 was released last week. It's time to revisit release plans for both current and future Asterisk releases. For the past few months, there have been discussions regarding some updates to Asterisk release policies. You can find my original -dev list post on this topic here [1]. I also spoke about this as part of my presentation at AstriCon, which you can find a text version of on the Asterisk project blog [2]. After much positive feedback, we have proceeded with implementing these policy updates. I made some additional comments on this topic and noted plans for next major release yesterday [3]. The key things to note are that all current Asterisk releases now have a specified end of life date. Future releases will have EOL dates from their initial release. For additional details regarding maintenance time frames for Asterisk releases, please see the project web site [4]. Thank you all very much for your continued support of Asterisk! [1] http://lists.digium.com/pipermail/asterisk-dev/2009-October/040082.html [2] http://blogs.asterisk.org/2009/11/10/asterisk-project-update-astricon-2009/ [3] http://lists.digium.com/pipermail/asterisk-dev/2009-December/041336.html [4] http://www.asterisk.org/asterisk-versions -- Russell Bryant Digium, Inc. | Engineering Manager, Open Source Software 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Showing name of extension when calling
On 12/22/09 6:00 PM, Kevin P. Fleming wrote: Doug Lytle wrote: Kevin P. Fleming wrote: This is called Connected Party information display, and it will be in Asterisk 1.8. Wasn't this scheduled for 1.6.2? I don't believe so, but I could be mistaken. It's certainly not in 1.6.2 as best I can tell from looking over the source code :-) Nope, it's not in 1.6.2. It went into trunk after the 1.6.2 feature freeze. -- Russell Bryant Digium, Inc. | Engineering Manager, Open Source Software 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Crash Hard, Crash Often
On Feb 5, 2009, at 9:32 AM, Josiah Bryan wrote: I've ran with verbose quite high lately, but havn't left debug on. Well, I just opened console and turned debug on to 100 so we'll wait and see what it shows next time it crashes. It's due for another any time now... If it's crashing, the first thing I would do is upgrade from 1.4.41.2 to the latest version, which is 1.4.23.1. Quite a number of issues have been fixed since the version you're using. If you're still having a problem with 1.4.23, then try to get a backtrace of the crash. First, build Asterisk without optimizations enabled by running make menuselect and turning on the DONT_OPTIMIZE flag in the Compiler Flags section. Then, if you start Asterisk with -g, it will generate a core dump on a crash. Finally, use gdb to get a backtrace. $ gdb asterisk core.12345 (gdb) bt (gdb) bt full Then, post this information on http://bugs.digium.com/ and we'll help you resolve the issue. Thanks, -- Russell Bryant Digium, Inc. | Senior Software Engineer, Open Source Team Lead 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6.1: iax trunk needs dahdi timing ??
Michiel van Baak wrote: On 20:24, Sun 14 Dec 08, sean darcy wrote: starting 161.1-beta3: chan_iax2.c:10925 build_user: Unable to support trunking on user 'iax-out' without DAHDI timing But I have these timing modules: ls /usr/lib/asterisk/modules/res_tim* /usr/lib/asterisk/modules/res_timing_dahdi.so /usr/lib/asterisk/modules/res_timing_pthread.so Do I need to do some magic to get these loaded? modules.conf is set to auto. Is this what iax is looking for? If you dont have any dahdi hardware installed and configured, make sure to load dahdi_dummy. That will provide you the timers. In 1.6.1, this should not be required. It's probalby a check in the code that shouldn't be there anymore. If you post this on bugs.digium.com, I'll remove it. -- Russell Bryant Digium, Inc. | Senior Software Engineer, Open Source Team Lead 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Subversion Mirror Down for Maintenance
Greetings, We recently moved our public subversion mirror to a new server. It is currently down for maintenance while we resolve some unforeseen problems. It should be back up by the end of the day. I apologize for the inconvenience, -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tired of midget packet received warnings
On Nov 8, 2008, at 1:30 PM, Atis Lezdins wrote:
Asterisk offers very much the same flexibility. You can disable
specific log levels (for example warnings) in logger.conf or you can
log everything to syslog, where filter out this specific message.
Of course, there is always this method, which is an even easier way to
disable this specific message:
Index: channels/chan_iax2.c
===
--- channels/chan_iax2.c(revision 155670)
+++ channels/chan_iax2.c(working copy)
@@ -7058,7 +7058,6 @@
memcpy(sin, thread-iosin, sizeof(sin));
if (res sizeof(*mh)) {
- ast_log(LOG_WARNING, midget packet received (%d of %zd
min)\n,
res, sizeof(*mh));
return 1;
}
if ((vh-zeros == 0) (ntohs(vh-callno) 0x8000)) {
--
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Open Source Team Lead
Digium, Inc.
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Re: [asterisk-users] Asterisk 1.6.1 + openais
On Oct 19, 2008, at 5:35 PM, Edgar Guadamuz wrote: I enabled the subscribe_event in the ais.conf and restarted aisexec. After that I restarted asterisk and the only warning I got in console was Oct 11 6:38:04.340485 [CLM ] nodeget: trying to find node If I disable the subscribe_event, asterisk starts as normal. What version of openais are you using? The versions listed in the stable section of openais.org do not include a bug fix to the event service that prevent a crash. Try one of the newer versions listed on the site. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk SIP and SRTP
Artem Makhutov wrote: are there any plans in including SRTP into Asterisk? Yes. The patches in http://bugs.digium.com/view.php?id=5413 are pretty old and do not work with asterisk 1.6.0. Correct. There is still work to be done, but it's getting much higher on our list of things that need some development effort dedicated to getting completed. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6.1 + openais
Edgar Guadamuz wrote: Hello, I followed the steps by Russell_ http://www.venturevoip.com/news.php?rssid=1980_ and I got it working for publish_event only. As soon as I add subscribe_event, Asterisk doesn't start and I just get the following message: *Oct 11 6:38:04.340485 [CLM ] nodeget: trying to find node * I have no idea what's wrong. There is not very much information about this issue. That message is normal. It is not an indication of an error. You said Asterisk doesn't start. Does it hang or does it just fail to start? If it is hanging, then I'll need a backtrace from Asterisk when it is hanging. If it is just failing to start, I would need to see the full Asterisk console output on startup to see what happens when it decides to stop. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] about application Jack and its runtime
On Sep 12, 2008, at 10:17 AM, Julien Claassen wrote:
exten = NUM,1,System(ast_picker ring.wav)
exten = NUM,2,Answer()
exten = NUM,3,GotoIf($[${SYSTEMSTATUS} = SUCCESS]?4:7)
exten = \
NUM,
4,Set(JACK_HOOK(manipulate,i(sstem:playback_1)o(system:capture_1)=on)
exten = NUM,5,System(ast_connect)
exten = NUM,6,Goto(8)
exten = NUM,7,VoiceMail(MBOX)
exten = NUM,8,Hangup()
JACK_HOOK is not going to do exactly what you want it to do here.
JACK_HOOK does the following:
Incoming call audio - audio in to jack, audio out from jack -
current Asterisk application
Outgoing call audio - current Asterisk application
When you use the JACK application, you get:
Incoming call audio - Jack Application
Outgoing call audio - Jack Application
It looks like the Jack application is really what you want here, and
that you're just trying to figure out how to connect to more than one
port. I would be happy to add that as an option. Until then, here is
something that should allow you to run your script to do more
connections.
It takes advantage of some features of the Dial application to
accomplish what you need. It does a dial back into the local
dialplan, which is just an extension that answers and runs JACK. It
uses the M() option to Dial() to run a macro after the called channel
(the jack application) answers. The macro waits a few seconds to
ensure the jack application has had time to get set up, and then runs
your ast_connect script to do the rest of your connections.
[default]
exten = NUM,1,System(ast_picker ring.wav)
exten = NUM,n,Answer()
exten = NUM,n,GotoIf($[${SYSTEMSTATUS} = SUCCESS]?4:7)
exten = NUM,n,Dial(Local/[EMAIL PROTECTED],,M(connect))
exten = jack-exten,1,Answer()
exten = jack-exten,n,JACK(i(system:playback_1)o(system:capture_1))
[macro-connect]
exten = s,1,Wait(3)
exten = s,n,System(ast_connect)
--
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Senior Software Engineer
Open Source Team Lead
Digium, Inc.
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Re: [asterisk-users] about application Jack and its runtime
Julien Claassen wrote: Something I find even stranger is that jack_lsp shows, that the asterisk input AND output ports do exist and ARE CORRECTLY connected. So I should get audio from my microphone and still I don't. Hope that helps... Can you share the dialplan that you're using? That may help me understand the audio path involved .. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] about application Jack and its runtime
Jay R. Ashworth wrote: A bit of nomenclature: is Jack the name of an Asterisk application? Or are you referring to JACK, the Jack Audio Connection Kit, whose name is all-caps, directly? And if not, of course, is Jack something that connects JACK to Asterisk? Sorry for the confusion. There is a JACK() application, and JACK_HOOK() function, which both connect Asterisk to JACK, the Jack Audio Connection Kit. And why should I know all of this already? :-) You should be psychic. It's a new 1.6 thing, though, so I don't expect many people to already know all about it. As posted earlier, more info here ... http://www.russellbryant.net/blog/2008/01/13/jack-interfaces-for-asterisk/ -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Video on Hold?
[EMAIL PROTECTED] wrote: Is the idea to switch to another video source or stay with the callers camera? An option for both would be nice. I could see a help desk placing a caller in que and a 1-2 min video coming on showing some simple video of how to hook it up. What I had in mind was to play a video stream that went along with the on hold audio. I was going to make it so if a video file was found with the same name as the audio file being played, it would play it. If I understand you correctly, you're also suggesting echoing back the caller's video. I'm not sure why that's useful, and it's also more difficult to implement with the current code. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] about application Jack and its runtime
On Sep 11, 2008, at 2:18 PM, Julien Claassen wrote: Does application Jack run the whole time, the conversation is going? If so: is there a SIMPLE extensions.conf-only-based way to put it in the background? I know AGI and other applications... :-( Yes. When you use the Jack application, it runs as blocking. It will not exit until the call is over. However, there is another option, which is the JACK_HOOK function. The Jack application acts as an endpoint for a call. The JACK_HOOK function is different. It allows you to hook into the audio stream for a channel while the channel goes off and executes other Asterisk applications. Your hook can be listen-only, write-only (which would replace the channel's audio with something), or as a manipulate audiohook. The manipulate version is the most common because you have full access to the audio stream for a channel and can do whatever you want with it. When I wrote JACK_HOOK, the idea I had in mind was being able to use some external application using jack to add sound effects, like a vocoder. Some more information about these interfaces can be found here: http://www.russellbryant.net/blog/2008/01/13/jack-interfaces-for-asterisk/ -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Video on Hold?
On Sep 8, 2008, at 9:15 AM, Gordon Henderson wrote: Does/Will asterisk support video streaming on hold? Been playing with videphones as of late, and a client asked about video on hold - standard MoH works fine - but on the target video phone the image just freezes - any way to inject a video? This is not something that is supported right now. However, it would be relatively straight forward to add for a developer interested in adding it. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Video on Hold?
On Sep 8, 2008, at 7:31 PM, Russell Bryant wrote: On Sep 8, 2008, at 9:15 AM, Gordon Henderson wrote: Does/Will asterisk support video streaming on hold? Been playing with videphones as of late, and a client asked about video on hold - standard MoH works fine - but on the target video phone the image just freezes - any way to inject a video? This is not something that is supported right now. However, it would be relatively straight forward to add for a developer interested in adding it. I just went and wrote a first draft in http://svn.digium.com/svn/asterisk/team/russell/video_on_hold/ . I haven't tested it, yet, though. However, as soon as I can get this tested and any issues fixed, it will be merged into Asterisk 1.6. This would be a fun project to finish up in the code zone at Astricon. :) -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bridge 2 incoming calls
On Sep 5, 2008, at 10:06 AM, Tim Panton wrote: On 5 Sep 2008, at 15:50, Steve Murphy wrote: Not in 1.4, but in trunk,(and 1.6.x) there is a the Bridge manager command you can call via the manager interface, which takes two required args, the names of the two channels to bridge, and an optional arg, that will send a tone to the second channel. see main/features.c Thats good to know. Will the xml-over http manager interface be able to do it too? (pretty please?) Yes. Also, in addition to being a manager action, it is also a dialplan application. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G722 and Asterisk 1.6
On Sep 4, 2008, at 6:25 AM, Steve Repo wrote: Specifically my questions are, [1] The quality of voice between g722 and say GSM or 729 I suppose that it's sort of subjective, but I think it sounds _awesome_. It's a huge difference in quality to me. You just need to try it out. :) [2] Interoperability between phones with g722 and other codecs Asterisk 1.4 does not support transcoding of G722. Asterisk 1.6 does support transcoding, so it can be used in combination with any other codec that Asterisk supports. [3] Asterisk support for G722 phones. Asterisk should work fine with any phone that supports that codec. Personally, I have only used it with Polycom phones. Also, again, Asterisk 1.4 only supports G722 passthrough, where as Asterisk 1.6 has full support. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stability problems in Asterisk 1.4.18 (and other 1.4.xx versions)
On Sep 4, 2008, at 4:12 AM, z_gringo wrote: I have several asterisk servers running a couple of different versions of 1.4. One of our severs in California is running 1.4.18 with the Dial Plan in Realtime mySQL. This server is storing voicemails in the database connecting via odbc. There are approximately 900 sip users registered at any given time. All of the SIP users are in the sip.conf file, which is extracted from the database. Any time this server gets to around 90 simutaneous calls (180 channels), the server is completely unstable. On some occasions, the asterisk process has continued to run, but is not processing any calls or registrations. On most occasions, the asterisk process crashes, restarts, crashes again, etc. During periods of lower traffic, the system appears to be stable. Going back to version 1.4.8 or 1.4.11 seems to be stable, but there is clearly a problem with 1.4.18 in this particular configuration. The OS is 64 bit debian. Has anyone seen something similar? We would be happy to help figure out what's going wrong on your system. However, the first step will have to be running the latest version. So, please give 1.4.22 a try. Then, please gather details and post them to http://bugs.digium.com/. If you'd like to discuss what you need to do to create the bug report, join #asterisk-bugs on the freenode IRC network. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G722 and Asterisk 1.6
On Sep 3, 2008, at 1:55 PM, Steve Repo wrote: I have a Grandstream GXP1200 and eager to try this codec. I've heard good things about the quality. Anyone tried it with asterisk? I can't until 1.6 is released. I have used G.722 with Asterisk many times. If you have more specific questions about it and Asterisk, I would be happy to try to answer them. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Versions of Asterisk, Asterisk-addons, Zaptel, and DAHDI
On Sep 3, 2008, at 8:32 PM, sean darcy wrote: Great. But I'm still a little confused. Does zaptel 1.4.12 work with asterisk-1.6.0-rc4? No. Asterisk 1.6.0 now _only_ supports DAHDI. It looks like we first upgrade to zaptel 1.4.12, and then to dahdi. We can go back to this release of zaptel if we have problems with dahdi. Or if we go back to zaptel, do we go back to 1.6.0-beta9 also? You can upgrade directly to DAHDI. However, if you have trouble with DAHDI and need to go back to Zaptel, then I would go back to Asterisk 1.4 instead of using an old beta of 1.6.0. Many things have been fixed since 1.6.0-beta9. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_jack and calling with pc only
Julien Claassen wrote: The question: Can I (mis)use my asterisk CLI interface to make and recieve calls coming in/going out via the ISDN-card, while using my soundcard I/Os under JACK as a phone? Yes, you can. You actually have two options for doing this. One is using app_jack and the other is using chan_console. chan_console uses libportaudio, which supports jack as an audio backend. To initiate a SIP call from the Asterisk CLI and connect it to app_jack, you would use a command similar to the following: *CLI originate SIP/[EMAIL PROTECTED] application Jack i(outputport)o(inputport) Using chan_console, you have various call control CLI commands available: *CLI console dial extension *CLI console hangup While using app_jack will give you more control over how you connect to Jack from the Asterisk CLI, using chan_console will give you more of a normal phone experience, as it will play tones to the audio interface that you normally hear when using a phone. Why I'm doing this and not use another app: 1. I'm blind, I LOVE my console/commandline 2. I tried linphone with SIP, didn't work. JACK crashed and the firewall is in the way. 3. The others don't have JACK and I need my JACK running (soundcard too big for the simple ALSA stuff and I'm a musician often in need of JACK's services. So asterisk seems to offer all I need. I know it's meant as a SERVER, but with all this horse-power: Is a simple client so far of the track? If you have any suggestions that would make app_jack or chan_console easier for you to use, then please let me know. Feel free to contact me directly. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM2400P Voice Quality Problem
Shariq Khan wrote: I m facing problem with TDM2400P pstn card. When someone dials, the voice quality is crappyInstead of hearing. Please contact Digium technical support for assistance with this problem. They are the experts when it comes to debugging these types of issues. This assistance is free when you are using Digium hardware. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VUC Friday: asterisk 1.6 fax, Drawing for Free Astricon Pass
randulo wrote: So this will be an option in selectmenu? (or menuselect or whatever it's called, it's been a long time since I've built asterisk) Yes, it is treated just like other Asterisk modules. It will be built by default if you have the proper dependencies installed. You can optionally use the menu to see if the dependencies have been met, or to disable it from being built and installed. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI working perfectly. Shouldn't it be broken??
Karl Fife wrote: Do I understand correctly that we are not talking about redundant MWI status traffic here, we're ONLY talking about the notion that asterisk ignores MWI subscription status and behaves as if it has 100% MWI subscription. That is unless subscribemwi=yes is in sip.conf. Is that an accurate summary? Yes. And theoretically, if I had thousands of endpoints and I needed 100% mwi subscription, there may be some theoretical efficiency to turning off all MWI subscriptions in all of the endpoints. Likewise if only 25% of my 'thousands' of endpoints needed any MWI, there would be some efficiency in setting subscribemwi=yes, and explicitly subscribing only those 25%. Is that right? Theoretically, I guess so. However, keep in mind that this is really about dealing with odd specifics of how certain phones behave more than anything else. Some phones won't subscribe but still expect MWI. Some phones may freak out if they receive MWI when they haven't subscribed to it. Thanks again for clarifying! I appreciate it! You are quite welcome. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] subscribemwi
Philipp Kempgen wrote: Is subscribemwi valid in peer context only or also in general? sip.conf.sample is not clear about that. Only within a peer definition. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI working perfectly. Shouldn't it be broken??
On Aug 24, 2008, at 11:29 AM, Karl Fife wrote: FYI, it's not an issue of the subscription not YET subscribing etc. If I were to restart the system and endpoints, all the subs slowly show up one by one, but the 962 never does -- even after days, weeks, and months. Yet the MWI always works perfectly from the get-go. Asterisk will send the NOTIFY for MWI even if the device doesn't subscribe, unless you tell it not to. This is necessary for some phones for MWI to work. If you _don't_ want Asterisk to do this, you can set the subscribemwi=yes option in sip.conf. This tells Asterisk to _only_ send MWI with an associated subscription. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pollmailboxes
Philipp Kempgen wrote: I'm pretty sure there are IMAP servers with custom hooks (Dovecot?). Not exactly easy but doable. That's true. That's another thing that I would like to get implemented one of these days ... BTW: Does pollmailboxes _disable_ the event based notifications? UPGRADE.txt is not clear about that. Some setups might want to use a mix. Event-based with custom triggers and polling-based with pollfreq=3600 as a safety net. No, polling does not disable the event mechanism. The polling loop simply generates events when it detects changes. The rest of Asterisk still expects MWI to be handled in an event based fashion. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pollmailboxes
On Aug 17, 2008, at 11:07 AM, Philipp Kempgen wrote: 1.6 UPGRADE.txt: * If you use any interface for modifying voicemail aside from the built in dialplan applications, then the option pollmailboxes *must* be set in voicemail.conf for message waiting indication (MWI) to work properly. This is because Voicemail notification is now event based instead of polling based. The channel drivers are no longer responsible for constantly manually checking mailboxes for changes so that they can send MWI information to users. Examples of situations that would require this option are web interfaces to voicemail or an email client in the case of using IMAP storage. I vote for a truly event based solution: - A web GUI modifies a mailbox (e.g. deletes a message) - The GUI triggers the polling of this specific mailbox Action: PollMailbox Mailbox: 1234 ... or Action: MailboxModifiedExternally Context: default Mailbox: 1234 Folder: Work File: msg0002 I agree that it would be a nice addition. I'm not so sure that it would remove the need to be able to enable periodic polling in Asterisk, though. For example, take IMAP storage of voicemail. There would not be any easy way to trigger the poll after someone has made changes using their IMAP client. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi and ztdummy
On Aug 15, 2008, at 8:54 PM, Jerry Geis wrote: Where is this file Zaptel-to-DAHDI.txt? I have searching svn asterisk, voip-info.org and dahdi linux complete and I did not see it. It will be included in the next releases of Asterisk that have the changes to support DAHDI (1.4.22 and 1.6.0). You can grab it directly from svn, as well. $ svn co http://svn.digium.com/svn/asterisk/branches/1.4 -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AstDB/Berkely DB - Hash function? Balanced-Tree? b-Tree? Linked List?
Karl Fife wrote: Does anyone know enough about the implementation of AstDB to know whether the data structure is a Hash function, a Balanced-Tree, a b-Tree, or a Linked List? I've never looked at the internals of db1. However, by simply looking at what code is included, it looks like it is based on a b-tree. You may have to add some debugging within db1 to see how nodes actually get laid out when you add your 160k entries. The code is in main/db1-ast/. If you're doing something this large, I would encourage you to consider just using a different database, and using func_odbc to access it as opposed to the astdb functions. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk vs c-client issues
Lee Lundrigan wrote: Hi everyone, Are there any incompatibility issues between asterisk and the c-client using SSL? When I enable SSL I get the error: *pbx.c:1832 pbx_extension_helper: No application 'VoiceMailMain' *whenever I am trying to access voicemail. But when SSL is disabled everything works great, just like its supposed to with imap. Any ideas? It looks like app_voicemail is failing to load when you build c-client with SSL support. Try running module load app_voicemail.so from the Asterisk CLI to see what the error message is. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ENUM lookup
On Aug 14, 2008, at 8:31 AM, Tilghman Lesher wrote: On Thursday 14 August 2008 07:33:11 Brian J. Murrell wrote: On Thu, 2008-08-14 at 14:15 +0200, Klaus Darilion wrote: Use the ENUMLOOKUP function, e.g.: And take note that it's very naive. See my previous posting for an enum AGI that is more intelligent. The only thing it does not do that I would like to add is give up on the DNS lookup much earlier than it does if a DNS server is unresponsive. If you'd like to give a suggestion on how to make the ENUMLOOKUP function more useful, I'm all ears. Sometimes the issue is that the people who are most qualified to make the dialplan functions more useful aren't in a position to do anything about it (either because they aren't C programmers or because they aren't ENUM users). I think the biggest issue with ENUMLOOKUP() is the inability to traverse multiple results after doing the lookup. To get access to a different result, you have to call the function again, which does another DNS lookup. However, this has already been improved in Asterisk 1.6. See the ENUMQUERY() and ENUMRESULT() functions. ENUMQUERY lets you do a query. ENUMRESULT lets you access the results of your query and perform whatever logic on them that you would like. (By the way, similar functions were also created for DUNDI - DUNDIQUERY and DUNDIRESULT) There is an unofficial backport of this functionality to Asterisk 1.4 available here: svn co http://svncommunity.digium.com/svn/russell/asterisk-1.4/ -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Accept anonymous connections from Unknown Peer
On Aug 14, 2008, at 3:25 AM, michel freiha wrote: I just need to permit to an extension created to accept anonymous connections from an UNKNOWN peer... currently I'm getting Declined response from Asterisk server when trying to dial this extension from OpenSer Server In the general section of sip.conf, set allowguest = yes. Also, set context = something, where something is a context that you would like unauthenticated calls to have access to. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] deadalocks in asterisk
Benny Amorsen wrote: D.J.Sateesh [EMAIL PROTECTED] writes: hi, i am recieving deadlocks frequently and its calls are getting hanged . Aug 11 13:13:53 WARNING[6367] channel.c: Avoided initial deadlock for '0xb6692258', 10 retries! That is actually more of a debug message, and is not necessarily an indication of a problem. We upgrade customers who hit that bug to 1.4... The locking is greatly improved. Which reminds me, is there an easy way to see what Asterisk uses memory for? Yeah. Compile Asterisk with the MALLOC_DEBUG option. With that, you will have the CLI commands memory show summary and memory show allocations, which show you all of the memory allocations that Asterisk has made by file, function, and line number. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP TLS error: ast_make_file_from_fd: FILE * open failed
Stefan Gofferje wrote: [Aug 8 23:30:13] SSL certificate ok [Aug 8 23:30:13] == Problem setting up ssl connection: error::lib(0):func(0):reason(0) [Aug 8 23:30:13] WARNING[23835]: tcptls.c:463 ast_make_file_from_fd: FILE * open failed! First, try the latest code in the Asterisk 1.6.0 branch. Also, make sure you're using a reasonably current version of OpenSSL. $ svn co http://svn.digium.com/svn/asterisk/branches/1.6.0 If you still have trouble, feel free to report it on http://bugs.digium.com/. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 encryption - LAN. no, INET: yes???
Stefan Gofferje wrote: I have configured all IAX clients with encryption. I use Zoiper as a softphone. When I make a call in the LAN from desktop-PC to *, the call is - according to wireshark not encrypted. Wireshark identifies the packets as normal G.711 mu-law packets. However, * reports the client as encrypted: k-tanco*CLI iax2 show peers Name/UsernameHost Mask Port Status sgofferj RFC-1918 IP(D) 255.255.255.255 4570 (E) OK (2 ms) Funnily, if my friend calls me from internet - also with Zoiper - Wireshark cannot identify the packets so I conclude, the call is encrypted. Does this make any sense? You'd have to provide a packet capture to see exactly what is happening. It sounds like on the call leg between your client and Asterisk, it isn't offering encryption as a capability, so it doesn't get used. However, when your friend calls you, and Asterisk makes a call out to your client, it offers encryption, and your client accepts it. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Getting Asterisk out of the RTP media path
On Aug 11, 2008, at 12:04 PM, SIP wrote: SIP wrote: When calling from our SIP proxy through Asterisk to the PSTN provider, we support reINVITES which tend to work seamlessly. However, when creating a call file which essentially connects a call from the SIP proxy to the SIP proxy, Asterisk wants to stay in the RTP media path. I understand that this is sort of the idea behind a bridged channel, but is there any way to avoid it? Is there any way to say Connect this number and this number and then get out of the way, or is this a design limitation? No ideas on this one? I've tried everything I can think of and then some and still can't get Asterisk out of the media path. I can do it if I don't originate the call with Asterisk, but only use Asterisk to connect one leg of the call, but if I use Asterisk to connect both legs, no luck. Going about this the wrong way? Asterisk will re-INVITE the media away from itself as long as it doesn't have a reason to need access to the media. For example, if you've enabled call recording, then clearly Asterisk needs access to the media. Other reasons include enabling features controlled via DTMF when the DTMF follows the media path. Nobody can help any further without seeing the details of your configuration. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The problem DIAL with option T,t
On Aug 11, 2008, at 2:03 AM, larry wrote: This is my setup of the features.conf but it had not any reaction after I pushed the *2 while calling was acting ! Could you tell me the reason ? Or give my the method of the setting. Thanks! LARRY [general] parkext = 700 parkpos = 701-702 context = parkedcalls [featuremap] atxfer = *2 The most likely cause of why it's not working is that you're not pressing the digits fast enough. The default timeout is 500 ms. So, if you don't press 2 within half a second of pressing *, it won't work. There is an option to extend this timeout - featuredigittimeout, I think. [applicationmap] set(DYNAMIC_FEATURES=tranf) tranf = *2,peer,waitexten(10|m) This is completely unnecessary for configuring call transfer. If you were to configure custom features, though, you would have the Set() command in the dialplan (extensions.conf). -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX t.38 on Asterisk 1.6?
On Aug 7, 2008, at 2:31 PM, Arturo Ochoa wrote: I got this scenario… FAX Machine - FXS (tdm800) -Asterisk - SIP - OPENSER - SIP - Asterisk - FXO(tdm400) - PSTN - FAX Machine Asterisk 1.6 currently has T.38 origination and termination support. It does not yet have fax gateway support. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-addons-1.6.0-beta4 compile error
On Aug 8, 2008, at 4:48 AM, Stefan Gofferje wrote: Hi, addons 1.6 don't compile here. Any ideas? It looks like you're trying to compiled Asterisk-addons 1.6 against Asterisk 1.4. You will need to install Asterisk 1.6 before you can compile and install Asterisk-addons 1.6. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX t.38 on Asterisk 1.6?
On Aug 8, 2008, at 7:39 AM, JR Richardson wrote: Asterisk 1.6 currently has T.38 origination and termination support. It does not yet have fax gateway support. -- Russell Bryant Russell, Can you please clarify what you mean. I think there is still a bit of confusion as to what termination and gateway and Asterisk 1.6 is all about, capability, functionality, call flow to what application, library requirements, spandsp versioning. And when do you think we can expect to see stable solutions for each. Origination and termination is exactly that. That means Asterisk is either originating the fax, or receiving the fax. Asterisk is the endpoint. This is supported for both regular fax and T.38 using the app_fax module in Asterisk 1.6, which requires spandsp 0.0.5. Gateway support is the ability to have a fax going through Asterisk as analog on one side and T.38 on the other. This is not yet support, but is obviously on the roadmap. I can't give you an estimation for when it will be done and officially supported right now, though. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-addons-1.6.0-beta4 compile error
Stefan Gofferje wrote: So, 1.6 must be _installed_ before compiling addons? It's not enough to have it readily compiled in the neighbour dir? That is correct, at least for the easy case. Alternatively, you can specify the Asterisk location as an argument to the configure script. -addons-1.6.0$ ./configure --with-asterisk=/path/to/asterisk-1.6.0 However, as Tzafrir noted in another reply, it is worth mentioning that regardless of which method you use, Asterisk-addons 1.6.0 modules _must_ be used with Asterisk 1.6.0. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [asterisk-dev] Locking, coding guidelines addition
On Jul 5, 2008, at 2:26 PM, Steve Totaro wrote: Again, your ability to miss the point is astounding. I never said I was a Freeswitch fan boy. I am just suggesting using a similar method of locking with FreeSwitch. Ideas, obviously Digium doesn't care enough to listen to it's users. I'm listening quite closely. However, saying FreeSWITCH is completely not helpful. If you want to talk about locking improvements, then I'm more than happy to talk about them. Naming an application is useless. Let's talk about the technical details that make one approach better than the other. You have yet to bring any useful discussion to the table. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [asterisk-dev] Locking, coding guidelines addition
On Jul 5, 2008, at 7:59 PM, Grey Man wrote: From what I can gather the suggestion from the FS approach is that each Asterisk channel should be handled after by it's own unique thread and save the need for any locking on the channel data structures in the first place. Having a thread per channel _absolutely does NOT_ remove the need for locking to synchronize access to channel data structures. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SayNumber while reading DTMF?
Douglas Garstang wrote: I'm using the SayNumber() app to read out a users balance for an IVR. Is there a way I can do that while waiting for DTMF input? Obviously, read() and Background() don't correctly say a number in number format. I do not know of a way to do that. It would be an extremely useful new feature to have, but as fair as I know, is not currently available. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 vs 1.4?
Steve Totaro wrote: I have consulted on so many systems with poor audio, the first thing I check is IAX or SIP. If IAX, I move over to SIP and the calls are prefect. I avoid IAX at all costs, use OpenVPN, open tons of ports on your firewall, whatever you can do to use SIP. The only time I will use IAX is if in some remote backwards part of the world, they have several NATs so it is impossible to control. Even www.iax.cc recommends using SIP. Overhead is of little concern with MPLS and big pipes. I would be interested to hear if you have had any of these problems with the latest 1.4 versions of Asterisk. A _lot_ of work has gone into IAX2 support in Asterisk 1.4, and specifically, the most recent 25% of the 1.4 series or so. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] features.conf not working
On Jun 6, 2008, at 4:33 PM, Manolet Gmail wrote: i have this on my features.conf: [applicationmap] testfeature = *9,callee,Playback,tt-monkeys extensions.conf: [globals] DYNAMIC_FEATURES=testfeature trunk_1 = Zap/g1 trunk_2 = Zap/g2 what else i have to add in order to make this works? im using 2 xlite, Just a hunch ... if you're using xltite, it's likely that you're not pressing the digits fast enough to satisfy the default timeout. The default featuredigittimeout is 500 ms. Change this option in features.conf and increase it to 2000 ms and try again. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.20 Released
Steve Davies wrote: Does this mean that the fixed IAX security fix for 1.2.28 (1.2.28.1?) will also be officially released now? If it helps, I have given 1.2 trunk some light testing and it seems reasonably sane. Thanks for the reminder. I will build that release right now. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Where does menuselect save your choices?
Sherwood McGowan wrote: Sherwood McGowan wrote: Just a quick question, wanted to see if anyone knew where the menuselect app stored your choices. I think it's menuselect.makeopts but I'm not sure...just thought someone might know. Sherwood McGowan P.S. I'll post here if I figure it out before there's a response :) Hrm...looks like I was right so far :) That is correct. You can also put some of those options in /etc/asterisk.makeopts or ~/.asterisk.makeopts if you want to set system-wide or per-user options for all Asterisk builds done on the system. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] are channel names unique
Benjamin Jacob wrote:
Are the channel names generated on 'Dial's supposed to be unique?
I see the channel names repeating on my asterisk box. I just wanted to
confirm this.
Can anyone point me to the lines of code where the channel name is
generated/calculated? I tried looking, but it looks like quite a big maze.
Channel names are not guaranteed to be unique at all. However, all channels
have a uniqueid associated with them. You can access it in the dialplan via
${UNIQUEID}.
--
Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc.
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Re: [asterisk-users] are channel names unique
Benjamin Jacob wrote: So I thought!! Thanks guys. But a query with regards to this : I need to send hangup commands based on these channel names only. So at any given point of time, for 'n' ongoing calls, will these 'n' channel names be different/ unique? If not, using AMI, how do we hangup a given channel? While channel names are not unique over time, at any given point in time, there should not be more than 1 channel with the same name. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G.722 for polycom
zhao_x_q wrote: I have test G.722 for many phones. I have try calls between sip G.722, sip G.722 to sip G.711, G.722 to RRI cards, PRIcards to G.722. I also test meetme conference. Other phones such as grandstream and fanwei have no problems. The sounds is good, grandstreams have little difference between G.711 and G.722. But Polycom's IP 550 have many problems. Polycom's G.722 to TE210E1 have problems the sound is choppy. Polycom's G.722 to conference also have problems, I even cannot heard the sounds. Has any friend knows the reasons for that? What version of Asterisk are you using? I have made a lot of G.722 related fixes over the last few months. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_queue New Function ToggleQueueMemberUse()
Benoit Plessis wrote: So i'm wondering if someone already as made a dialplan function that could toggle the 'Use' flag of an agent ? or if this kind of function would be integrated into the core if i build it ? This is a slightly different approach, but have you seen the state interface code that is in Asterisk 1.6? There is a backport of the code for 1.4 floating around somewhere, I think. It allows you to specify a different device for a queue member that app_queue will use to determine the state of an agent. So, you can still list a Local channel for dialing, but Asterisk will look at the state of SIP/myphone, for example, to know whether the agent is busy or not. Alternatively, if you would like to control the usability of an agent through the dialplan, then you could use the DEVICE_STATE() function to create a custom device state. Then, you could list your custom device as what app_queue should look at before attempting to call the agent. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk ZRTP?
Matthew Rubenstein wrote: What's the status of ZRTP supported by Asterisk? There was some discussion on the -dev list and -users list, but it was inconclusive. At about the same timeframe, a bug (#0010024) was opened and updated for several months, but has been suspended since late 2007. Does any version (1.4.x, 1.6.x) of Asterisk support ZRTP with clients (or with other servers)? Any successful testing with specific clients/peers to report? If not, are there any serious efforts underway? As you said, there have been various discussions about it. However, I have never seen any code, or demonstrations showing that the code may exist. So, it is not supported in any version of Asterisk, and I have absolutely no idea when and if it ever will be. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_queue New Function ToggleQueueMemberUse()
Benoit Plessis wrote: This is a slightly different approach, but have you seen the state interface code that is in Asterisk 1.6? There is a backport of the code for 1.4 floating around somewhere, I think. It allows you to specify a different device for a queue member that app_queue will use to determine the state of an agent. So, you can still list a Local channel for dialing, but Asterisk will look at the state of SIP/myphone, for example, to know whether the agent is busy or not. ok but since we are using IAX2 with ZoIPer for sendurl() handling this won't help I don't see why this wouldn't help. You just list the IAX2 peer as the device Asterisk uses to determine the state of the agent. Alternatively, if you would like to control the usability of an agent through the dialplan, then you could use the DEVICE_STATE() function to create a custom device state. Then, you could list your custom device as what app_queue should look at before attempting to call the agent This is more interesting :) Is it from 1.6 too ? Yes, this is also from 1.6, but an unsupported backport of DEVICE_STATE(), exists, as well. http://www.asterisk.org/node/48325 http://www.asterisk.org/node/48360 -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SLN File Format
Tzafrir Cohen wrote: The only downside is that you can simply concatenate two files using 'cat file1 file2 file1file2' with wav as you can with raw formats (provided that both originals are of the same format), because the header is not part of the stream. Correction for the archives ... you can _not_ simply concatenate two [wav] files. :) -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Restarting due to segfault
Sanjay Rajdev wrote: I am not a developer for Asterisk and even cannot make changes in the SVN as I do not know lot about the branches in it, but if someone from your side can take the effort to change this It would be great help for others. Please open a report on http://bugs.digium.com that describes what you have found. We will help you get the fix into svn. Thanks, -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SLN File Format
Philipp Kempgen wrote: Just out of curiosity: I can't remember when I last had to concatenate 2 sound files. So why does this always come up? IMHO it's one of those things you hardly ever need.(?) I can't remember the last time I have done that. :) Anytime I need to do something like that, I just use a nice tool like audacity ... -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] better enumlookup handler
Brian J. Murrell wrote: Does anyone have a better ENUM lookup handler than the built-in ENUMLOOKUP() function? The built-in function does not properly handle multiple return values such as: 8.9.9.3.2.8.8.6.6.8.1.e164.org has NAPTR record 200 10 u E2U+SIP !^\\+1866(.*)$!sip:[EMAIL PROTECTED] . 8.9.9.3.2.8.8.6.6.8.1.e164.org has NAPTR record 200 10 u E2U+SIP !^\\+1866(.*)$!sip:[EMAIL PROTECTED] . Have you taken a look at the ENUMQUERY() and ENUMRESULT() functions that are a part of Asterisk 1.6? The ENUMQUERY() function lets you do a single enum query for a number. Then, the ENUMRESULT() function lets you access and iterate through all of the records received from the query. Then, you can use dialplan logic to try each one without having to actually do the lookup over and over ... I have an unsupported 1.4 backport of these functions available. svn co http://svncommunity.digium.com/svn/russell/asterisk-1.4/ -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] better enumlookup handler
Brian J. Murrell wrote: I have not even entertained thinking of 1.6 yet. :-/ Fair enough. That's why I pointed out the feature. Dude! Where were you yesterday, before I spent a few hours last night writing my AGI? :-) Sorry. I have trouble keeping up with this list. :) Now that's what I want to hear. Back port. Woohoo! I wonder how hard a backport from your 1.4 to my 1.4.17 will be. I might just have to take a whack at it. :-) Thanx Russell! You're quite welcome. It should be as simple as dropping the func_ module into the funcs directory. The Asterisk build system should see it, compile it, and install it as usual. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Text for built-in recordings
Roderick A. Anderson wrote: Steve Prior's mention of using Allison's voice with Cepstral reminded me to ask: for a listing of the text for the built-in recordings. I found a web page but I'd prefer not having to scrape the info out of it. I didn't notice anything while wandering through the source code/files. I want to rebuild them for my system using Callie's voice. All of the sounds releases include a text file with the script for every sound file. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX issues with 1.4.19.1
Brian J. Murrell wrote: Right. Which to me at least, tightly couples it. IOW, the security fix, while yes, it fixes the security problem, is quite useless without this other fix as it makes iax2 unstable. I agree with you. I am in the process of working on the Asterisk 1.2.20 release, which will contain this IAX2 fix. However, you have convinced me that this fix should be released against the previous security release, as well. So, tomorrow, I will make an Asterisk 1.4.19.2 release, which includes this one change. As a part of making that release, it will come with a patch against 1.4.19.1, which will include the changes needed to make IAX2 usable again, if people needed the patch for a custom version. I will also update the security advisory to note the effects of the original changes to address the security issue. I will then publish announcements as usual that to hopefully notify everyone that doesn't closely monitor commits to Asterisk, or other high volume Asterisk mailing lists. Thanks for the feedback, -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DUNDi call impossible in one direction
Andrea Spadaccini wrote: I've set up DUNDi between two asterisk boxes, and sometimes happens that calls from machine A can't reach peers in machine B, but calls from B to A work correctly. The strange thing is that the CLI command 'dundi show peers' shows correctly the registered peer in both servers, and in this situation if I make a call from B to A, suddenly peers in server A are able to call peers in machine B. Try using the DUNDi query CLI command to see what results your server is getting when you try to make calls. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Basic modules of Asterisk
Sanjay Rajdev wrote: I just want to Run Asterisk with the basic required modules, What can I do to achieve so? My only requirement is to run SIP clients and the Dictate Module. 2 options: 1) Before compiling and installing Asterisk, run make menuselect to select only the modules that you want to use. That way, only those modules are compiled and installed. 2) After installing Asterisk, edit /etc/asterisk/modules.conf. By default, Asterisk will load all installed modules. You can turn off the autoload functionality, and explicitly list the modules that you need. You probably want pbx_config, chan_sip, app_dictate, app_dial, probably some others ... -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T38 Passthrough Verification
JR Richardson wrote: I have 1.4.9.1 setup, with the compiler flags enabled for T38, and have a Mediatrix 2102 and a Linksys SPA 8000-G1. I can pass faxes between devices but can't seem to invoke T38 pt UDPTL. It's enabled in sip.conf [general] and well as the [peer]. I get an error at the CLI: WARNING[3096]: chan_sip.c:14149 handle_request_invite: RTP re-invite after T38 session not handled yet ! sip show channels shows the call setup with ulaw. Try setting canreinvite=no for the peer doing T.38. It looks like the code in Asterisk 1.4 will not allow re-invites for an established T.38 passthrough call. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dundi network - redundancy / fault tolerance ?
Ex Vito wrote: Now, how to move on to acheive some kind of fault tolerance ? According to the docs we've studied, DUNDi does not like loops (which we assume one can limit with low enough TTLs). Which documentation are you referring to? You may have misunderstood something, or there may be some false information floating around the internet (*GASP*). The DUNDi protocol has built in handling for loops. It keeps track of which nodes have already been queried, so you don't have to worry about loops in your network. Every node can peer with every other node if you really wanted to. Of course, that's not necessarily the most efficient thing to do ... Our doubts are: - Should one use the order peer parameter to specify alternate lookup paths / peers ? Is that its purpose ? If not, what is it used for ? The order parameter is really a tool. There is not an exact situation that it is intended for. It depends on your network. Keep in mind that DUNDi caches results along the way. If you use the order option to have servers send queries through a primary server, you getter better caching performance. - Alternatively, should one create loops in the DUNDi topology and limit them via TTL ? As I said before, don't worry about loops. Set your TTL to handle a worst case path for a query in your DUNDi topology. - If both options are possible, which would be the trade-offs between them ? (Not clear at all to us!) I'm not sure what you mean. The best thing to do is to have multiple peers. Have every server have at least two peers. Setting a primary and secondary can be good for caching reasons. - Assuming any of the above is possible as a means to acheive redundancy, which of the following topologies would your prefer ? (hmmm, maybe I need to refresh my graph theory...) ;-) #1 - Peer each PBX with both hubs #2 - Duplicate both hubs and peer each PBX with its hub and its hub dup For better understanding, take a look at: #1 - http://www.2photosharing.com/images/qhpnzycd7j7kf26j2f.png #2 - http://www.2photosharing.com/images/npzbwvgnr4t079laou0.png Thanks in advance for review and feedback. I'm not necessarily up on my graph theory, either, but I would probably go with something like #1. A combination of having multiple peers and usage of the order option can give you good redundancy without hurting your performance. When you set primary, secondary, etc. peers, the server will attempt to contact them one at a time. If you have multiple peers, but do not set an order, they will all be contacted at once, which may (probably will) increase latency for call completion, will increase bandwidth consumption, among other things. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Restarting due to segfault
Sanjay Rajdev wrote: I have Asterisk 1.4.15 installed on a Fedora Core 8 machine. Asterisk is snip In the dialplan we have used MixMonitor() to record the calls. Can anyone help me on getting to the root of the problem or fixing it? We have fixed a _lot_ of issues in that area of the code since 1.4.15. I would suggest trying the latest version. If it still gives you trouble, please let us know on http://bugs.digium.com so that we can fix it up for you. Thanks, -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P dialout problem
Anthony Messina wrote: Using asterisk 1.4.18 and zaptel 1.4.9 on x86_64, I am having trouble dialing out to the pstn. The call is initiated at Zap/1-1 and should exit via Zap/3. I get the following: This should be fixed in Zaptel 1.4.9.2. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Request for testing: Distributed device state
Greetings, One of the things that I have been working with off and on is a new event API for Asterisk. Most of the current infrastructure was written almost a year ago. However, just recently, I extended this system to be able to allow device states to be shared within a cluster of Asterisk servers. The current system for distributed events is designed for a high speed LAN. However, I also have plans to extend this to DUNDi so that it can be used within a DUNDi network, as well. If you are interested in this work, I would appreciate any help with testing. I have documented how to set it up and how I used some custom states to verify the functionality. To download the code: $ svn co http://svn.digium.com/svn/asterisk/team/russell/events asterisk-events For information on setup and testing: http://svn.digium.com/view/asterisk/team/russell/events/doc/distributed_devstate.txt?view=markup -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Linux/Python 2.4.2] Forking Python doesn't work
Vincent wrote: On Wed, 13 Feb 2008 10:59:38 -0200, Diego Aguirre [EMAIL PROTECTED] wrote: try to use System() instead of AGI() Thanks, but no go. I get an error: [Feb 13 21:57:55] WARNING[2138]: app_system.c:107 system_exec_helper: Unable to execute '/tmp/netcid.py|2000|Joe' The arguments to System() are a bit different. Put it in just like you would type at the command line. System(/tmp/netcid.py 2000 Joe) -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrade 1.2 - 1.4 voice files
Adrian Marsh wrote: In the Make menuselect, I noticed theres no .SLN voicefile selection for the basic audiofiles - has SLN been depreciated? No, the sln format is still supported. We have just never distributed any files in that raw format. Previously, we only had gsm recordings. For Asterisk 1.4, we got all of the prompts re-recorded so that we could distribute them in a number of higher-quality codecs, as well as in 3 languages. The actually scripts of the files has not changed much, as far as I remember. The sounds.txt file in 1.2, and the 1.4 sounds packages should say exactly what they are. You can always compare them with diff. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] BLF and Asterisk 1.6.0b2
Thomas Kenyon wrote: Is anyone else having trouble with Asterisk 1.6.0b2 not sending busy hints to phones? I'm not reporting this a s a bug because (although I have it working with Asterisk 1.4.17, the hardware involved is different. What type of device are you subscribing to, is it another SIP phone? If so, what is the associated configuration in sip.conf? Do you have call-limit set to some value, or the combination of callcounter and busylevel? If so, what are they set to? (You must have these options set for it to work) -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk-Addons install success-Could not find ooh323.conf
[EMAIL PROTECTED] wrote: Am I doing something wrong? What I should do to get ooh323.conf cp asterisk-ooh323c/h323.conf.sample /etc/asterisk/ooh323.conf -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Discover Asterisk 1.4 :: Jitterbug, no, Jitterbuffers
Johansson Olle E wrote: In my series of articles about Asterisk 1.4, I've now arrived to the new jitter buffer that enhances voice quality for those of you using Asterisk as a PSTN gateway. Please read http://www.voip-forum.com/category/asterisk/asterisk14/ I wrote a patch that lets you use the jitterbuffer in Asterisk 1.4 for more than just PSTN gateway functionality. Originally, there was no way to use it when connecting to Asterisk applications that did not create outbound channels and bridge calls (basically only Dial and Queue). This is actually still the case, but what I did was add support for using the jitterbuffer when you are bridged to a Local channel. That way, you can use it when connected to Voicemail, Meetme, or whatever else you want. See this post for more information: http://www.russellbryant.net/blog/index.php/2007/10/09/asterisk-jitterbuffer-support-for-applications/ -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] realtime warning
Rilawich Ango wrote: Hi, The server log shows the following message. [Jan 29 04:59:02] WARNING[1896] config.c: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available Does it mean the server failed to file the mysql server? I have installed mysql and both asterisk and mysql are located in the same server. What do the message mean? It seems the message will cause the user failed to login. How can it be solved? Did you install res_config_mysql from asterisk-addons? -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialogic card
Steve Totaro wrote: I was under the impression that only ABE supports Dialogic boards. I thought I saw that in passing so I could be totally wrong. There was talk but ABE has never supported Dialogic cards. If anyone would be interested, I would recommend expressing it to [EMAIL PROTECTED] -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sipsock_read: BAD! BAD! BAD!
Douglas Garstang wrote: Does anyone know the cause of these BAD BAD BAD messages? I think I lost all my calls when it happened too. We have nagios running against IAX and nagios reports that IAX is down. It would seem that the entire application locks up when this happens and calls are dropped. Yes, that message generally indicates a deadlock in Asterisk. Connected to Asterisk 1.2.14 currently running on flexo (pid = 26846) There have been 380 fixes made to Asterisk 1.2 since 1.2.14, 61 of which were made to chan_sip. :) -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
