similar to: Asterisk Stops...where to look?

Howdy, Was there ever a fix for this? I have Trix 2.6 running asterisk 1.4 and have to set an extension with call-limit=1. However that user can no longer do attended transfers from Linkys 962 ip phone. Is there anyway around this? Cheers, Taff..

Ok Mark Michelson. Thank you very much! You answer tells me that I was in the wrong path trying to access information from SIP 183 message. I need to find a way to let the callee pass information/data to the caller, even before accepting the call. That is, send data during the ringing time. And in my case, there will be more than one callee ringing at same time. As ASTERISK will not forward each

The Asterisk.org development team has released Asterisk versions 1.2.30.3, 1.4.23-rc2, 1.6.0.2, 1.6.1-beta3, as well as Asterisk-Addons versions 1.6.0.1 and 1.6.1-rc2. These releases are available for immediate download from http://downloads.digium.com/. This update for Asterisk includes a fix for a regression introduced in Asterisk 1.2.30 and Asterisk 1.4.21.2 and has existed in the

Hi. The ASTERISK wiki has a page showing the function PJSIP_HEADER(). However, it doesn't explain if such function works only over SIP INVITE messages or if it can be use, for example, to read headers from others types of SIP messages too. So, can I use PJSIP_HEADER to read the SIP 183 message header? Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel

Went from 1.6.0.6 to 1.6.0.8 and resulted in segmentation fault. Reverted to 1.6.0.6 and back to normal. ------------------ Linux asterisk.hulber.com 2.6.18-128.1.1.el5 #1 SMP Mon Jan 26 13:58:24 EST 2009 x86_64 x86_64 x86_64 GNU/Linux Apr 3 11:49:56 asterisk kernel: asterisk[3780]: segfault at 00002ce1ac0537a8 rip 0000003e980715a8 rsp 00007fff5bf00c30 error 4 Apr 3 11:50:00 asterisk

Hi, In queues realtime, when the queue start and when it ends. I mean, for example to calculate service level, how many calls, etc. If I want to start the queue from with 0 calls, etc, how do I do this? And if I want to stop it, so I can start it again?? Thanks!! Regards, Sebastian -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I have changed directory ownership permissions recursively such that it is owned by username:groupname , where groupname is not the default group, i.e., username. However, when a user creates a new file the default permissions are again username:username. How can I give ownership permissions on a particular directory so that any files created in that directory will always have specifc

Hi I read on LLVM blog that meta-data has been implemented to coney debug information to run-time system. Can one use meta-data to convey developer specifc hints to run-time system (e.g. JIT compiler)? Keen to know your thoughts on this. Thanks Javed -- my homepage: http://www.javedabsar.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello all, I am still reading about my QoS rules and I need that one of my servers (that is into my LAN but has an routing ip address) did not get into the qos rules I have. So I want that all traffic coming or going to that specifc host did not get shapped by any traffic control and do not get even into a QoS class. How can I do this? Att, Nataniel Klug

Hi, Is it possible to pick the context of a call from chan_sip based on the domain of the To: header of the INVUTE? I've had a quick look throught he code and can't see anything, I want to use the voicemail virtual hosting with chan_sip. Can the sip domain be picked out with a global in extensions.conf? This woud also solve my problem. If not is there any specifc reason/restriction

I have an asterisk system where the asterisk daemon dies typically at least once per day. It is running in the wrapper safe_asterisk, which automatically starts the daemon back up. But we find this unacceptable because when the daemon dies, we usually have active calls drop, and sometimes we have to run asterisk -r -x "module reload" after the daemon starts back up before everything is

Hello list, I''m currently a bit planless so perhaps someone here could give me a point in the right direction. History: I wrote a shaper web tool (http://shaper.netshadow.at) and now got several feature requests if it would be possible to graph "what''s going on" (this mean per IP address, tcp/udp ports or protocols) in a specific chain. A chain represents a specific

Javed Absar wrote: > Hi > > I read on LLVM blog that meta-data has been implemented to coney debug > information to run-time system. > Can one use meta-data to convey developer specifc hints to run-time > system (e.g. JIT compiler)? > Keen to know your thoughts on this. I don't see why not. I've used LLVM metadata to record type-inference information and to

Has anyone come across the above in 2.0.7 in a Solaris/Sunos environment. The result is that certain users cann't map drives. Thought it might be something specifc to this version and the user being a secondary member of a large group (not netgroups)... Then I checked the source and the parse_net.c is the same version.... Anyone.... TIA Gordon

Hi all, I have upgraded my Asterisk box from 1.2.x to 1.4.x version: it seems that sometimes some phones become paused and cannot receive calls anymore. I tried to set autopause = no in every section of my queues.conf but nothing changes.... Anybody knows why a phone becomes paused? Is it an Asterisk 1.4 bug or there is a particular reason for this behaviour? Thank you. Giorgio.

Hi, Sometimes my asterisk server stops. (after a day or two) Last output from CLI is: -------------------------------- -- Registered SIP '000b82017eb7' at 213.237.12.125 port 11620 expires 120 -- Channel 0/26, span 1 got hangup -- Hungup 'Zap/26-1' voip1*CLI> Disconnected from Asterisk server Executing last minute cleanups Asterisk cleanly ending (0).

Hi All, I realise this is off topic, but its likely the best place to ask! I sent an email to snom support a few days ago but have yet to recieve a response.. Perhaps some one has found a solution to this problem already? I've searched the mailing lists and google and found nothing useful. I've also read Snom's mass deployment documentation but thats no real help in this case.

I typically start asterisk with the safe_asterisk script: 22865 pts/3 S 0:00 /bin/sh /usr/sbin/safe_asterisk 22867 pts/3 S 0:31 asterisk -vvvg -c 22871 pts/3 S 0:00 mpg123 -q -s --mono -r 8000 -b 2048 For-You.mp3 for.m 22873 pts/3 S 0:00 mpg123 -q -s --mono -r 8000 -b 2048 For-You.mp3 for.m But when I do a "stop now" from the CLI, the mpg123 causes a

Ok, this is my third help plea for the day, however it's something that has been bugging me for quite some time. To put it quite simply, "Stop now" doesn't. Neither does "Restart Now" Well, ok, if I start *, and then "stop now" it does. However, after a day's calling (2000+ calls from SIP->Zap pri or zap pri->sip), it doesn't. The only way

Hello, Long time subscriber/reader of this list - thank you for all the great ideas. Scenario: We currently provide a hosted ACD system using Mitel phones (speaking the Minet protocol) to an NCI based server solution. The logic behind this choice was the emulation of key system features etc... Many of our clients have asked for basic call queue functionality: - Agents having the ability to