similar to: Asterisk 1.2.31.1, 1.4.22.2, 1.4.23.1, and 1.6.0.5 released

I installed version 1.6.0.3-rc1 and my AMI application stopped working. I reinstalled 1.6.0.1 and it worked again. I reinstalled 1.6.0.3-rc1 and it stopped. Looks like a problem in the software to me. Following the same steps using the same code for the AMI and conf files for * I get bad behavior in 1.6.0.3-rc1 and good behavior in 1.6.0.1. I have this action: Action: Originate Channel:

I just upgraded a very old Asterisk installation to the last 1.2.31 I can find in Asterisk.org site. Now for some reason my IAX clients cannot connect to the server. I can do a "iax2 show peer iaxmodem1" and I get this: * Name : iaxmodem1 Secret : <Set> Context : oficina Mailbox : Dynamic : Yes Callerid : "" <> Expire

Hi fellow devs, I was looking for converting LLVM IR to JavaBytecode by using lljvm project of https://github.com/davidar/lljvm They don't seem to compile, once i run *make* in the root directory it give me the error : cd include && make all make[1]: Entering directory `/home/rtiwari1/llvm_new/lljvm/include' make[1]: Nothing to be done for `all'. make[1]: Leaving

Dear All, i have following CLI error while try to run this command from Dialplan *TrySystem("DAHDI/45-1", "asterisk -rx "dialplan add extension 1234111,1,Goto(incomingdundi,s,1) into dundilookup"") in new stack WARNING[32626]: app_system.c:81 system_exec_helper: Unable to execute 'asterisk -rx "dialplan add extension 1234111,1,Goto(incomingdundi,s,1) into

Hi, all. I've been searching google, bug reports and forums and have looked in all the asterisk-users list archives back to 2003 but haven't seen an answer to this, so thought I'd post here. The problem seems to be that Asterisk 1.6.0.5 is sending backslashes (needed to escape commas and so forth in 1.4.21.2) as *literal* backslashes to Mysql, so that Mysql gives a syntax error

Hi, just upgraded my Asterisk from 1.4.22-rc5 to 1.4.23.1 keeping the same zaptel/libpri/mISDN/add-ons. It crashes when transferring a call. Anybody tried it with success? Thank you Giorgio

I know the call parking feature changed in 1.4.23.1 to fix some serious issues. I'm seeing a major change though which I find disturbing. A person parks a call by transferring it to the parking position (700). When the timeout value is reached, the call is NOT returned to that device, but rather the 's' extension of the phone's registered context (in this case [unrestricted]).

Hello List, I have been working on a little PHP software that uses AMI's UpdateConfig command in order to modify some of it's config files. I was working with 'Asterisk 1.4.22.1' and everything was working. After upgrading to 'Asterisk 1.4.23.1' I receive a lot of errors of the type: ERROR[11505]: utils.c:966 ast_carefulwrite: write() returned error: Broken pipe

Hello all, Not sure if this mail belongs to this users or dev list. Sorry about that. We have the following scenario: PhoneA OpenSER Asterisk PhoneB PhoneC | | | | | | | | | | | | | |

As a librarian and a father, I have both a professional and a parental interest in democracy and access to information. In addition to democracy, libraries and education are related to other concerns. For example, our physical and mental health is critical to our ability to learn and to process information fully. Suppose a worldwide referendum was held on a proposal for single-payer global

Hey, welcome! I'm curious about the unsequenced modification checker, is it something that I should have seen but missed for whatever reason? It might be useful, and I think I'm seeing why don't compiler warnings cover all cases, i.e. why the analyzer's path sensitivity would help here. But I can't answer until I see it :) -eg. on our Phabricator. We are currently having

The Asterisk Development Team has announced releases of Asterisk-Addons version 1.4.11, 1.6.0.5, 1.6.1.3, and 1.6.2.1. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ These releases resolve several issues reported by the community: * Fix reading samples from format_mp3 after ast_seekstream/ast_tellstream. (Closes issue #15224.

The Asterisk Development Team has announced releases of Asterisk-Addons version 1.4.11, 1.6.0.5, 1.6.1.3, and 1.6.2.1. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ These releases resolve several issues reported by the community: * Fix reading samples from format_mp3 after ast_seekstream/ast_tellstream. (Closes issue #15224.

hi all, I'm have a bit of a hard time with some segfaults on running 1.4.23.1 and mISDN 1.1.8. I already enabled " DONT_OPTIMIZE" and " DEBUG_THREADS" in asterisk so I can now generate a bt. I did that (following the instructions on voip-info) but I'm not sure how to "read' the output now. By looking at the bt below, can one see if the problem is caused by

That source file was removed from LLVM in r232397 on March 16, 2015. It looks like lljvm hasn't been updated in a long time. LLVM's C++ APIs are not stable, so there is no expectation that a project built against LLVM's C++ API in 2015 would build or reasonably function against LLVM trunk. The project probably works against LLVM 3.6.2 which was (I believe) the last LLVM release to

hi to all.. i having a BIG problem... i am the administrator of a "ISP"... the thing is that (besides the problem of behing a sysadm) i need to limit the bandwidth per user... i mean, every IP needs a diferent bandwith.. can i do that? how? i know is a very looong answear, but you can save my day! (and my job) if you could give me a hand with this.. PLEASE, im looking for this a long

Hi fellow devs, I was looking for a way to run C on the JVM and i found LLJVM which apparently does exactly that! Howevere,unfortunately it seems dead on the main david's git https://github.com/davidar/lljvm as well as the slightly newer one https://github.com/hyc/lljvm. They don't seem to compile, once i run make in the root directory it give me the error : " *In file included

Hi, Thanks for replying ! On 03/02/2018 10:58 PM, Artem Dergachev wrote: > Hey, welcome! > > I'm curious about the unsequenced modification checker, is it something > that I should have seen but missed for whatever reason? It might be > useful, and I think I'm seeing why don't compiler warnings cover all > cases, i.e. why the analyzer's path sensitivity

I am running Asterisk as non root and have set the required permissions for all directories including the moh dir specified in musiconhold.conf yet asterisk still complains it doesn't have access when starting? I get: WARNING[3600]: res_musiconhold.c:987 moh_scan_files: chdir() failed: Permission denied I ran asterisk with strace, and don't see it failing to access anything? Anyone know

Hi, I'm trying to run Asterisk 1.4.23.1 on a small ARM linux board (TS-7800). Everything compiles fine, but on startup Asterisk always crashes while loading chan_sip. If chan_sip is removed, it starts up fine, but I really need SIP to work. Any ideas? Thanks. -- James