similar to: pstack debug asterisk

Hi, How to install pstack in CentOS 6.5? Which repository or package it is part? Thanks a lot, Cl?vis -- Clovis Tristao - UNICAMP/Faculdade de Engenharia Agricola Administrador de Redes - Secao de Informatica (SINFO) E-mail: clovis at feagri.unicamp.br http://www.feagri.unicamp.br MSN: clovis_tristao33 at hotmail.com Fone: 55(19) 35211021-35211031-35211038-91173116

Hi ALL, After upgrade 1.8 my MWI wasn't working I do have setting in voicemail.conf. Do i need to do anything else to fix my MWI on polycom 501 ? It was working with 1.2 asterisk. pollmailboxes=yes -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110608/e1f59a92/attachment.htm>

Hi Guys! This is strange issue with 1.8 I have restarted my asterisk and it destroy all registered SIP peers now only solution is i manually reboot all phones to get them register back. I have never seen issue like this before. Any idea what would be the issue ? Thanks S -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hey Guys! We had migrate asterisk 1.2 to 1.8 now big issue is queue system. Before we had 3 queues and we were using AgentCallbackLogin but now its quite difficult to use AddQueueMember. Is there any easy way to logged into multiple queue using AddQueueMember ? and restrict agent for specific queue ? -S -------------- next part -------------- An HTML attachment was scrubbed...

Hey Guys! We have two Asterisk with A102D Sangoma cards now i want to connect them back-to-back over PRI line via Cross-cable so what would be the configuration specially timing source and all? anybody did it before like this ? I want to make sure everything before putting in production.. (saving my downtime) -S -------------- next part -------------- An HTML attachment was

Hey Guys! I have two asterisk 1.8.3.2 same version on both machine but why one asterisk has "reload" command but other doesn't ? satish-desktop*CLI> core show version Asterisk 1.8.3.2 built by root @ satish-desktop on a x86_64 running Linux on 2011-03-25 16:10:39 UTC satish-desktop*CLI> re <tab><tab> realtime reload shirley*CLI> core show version Asterisk

Hey Guys, I knew this is stupid question but i just want to know what is the difference between Queue member logged out vs Pause ? -Satish -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110321/7646a66b/attachment.htm>

I need to change the sip port from 5060 to 5061 actually we already used 5060 for proxy to sip any idea to change 5060 to 5061 so all can acces the sip using this port please help........................ On 4/8/11, asterisk-users-request at lists.digium.com <asterisk-users-request at lists.digium.com> wrote: > Send asterisk-users mailing list submissions to > asterisk-users at

Following is my scenario to connect back to back PRI of two asterisk server. PRI cards are Sangoma A102D [Asterisk1]------------[PRI]-Cross Cable---------[Asterisk2] Asterisk1 ; Span 1 (MASTER) switchtype = national ; commonly referred to as NI2 context = from-pstn group = 0,24 echocancel = yes signalling = pri_net channel => 1-23 Asterisk2 ; Span 1 switchtype = national ; commonly

Hey Guys! I am getting this wired error when i am calling IAX trunk. Everything works! but i want to get rid on these RED WARNING messages.. what is wrong here ? I have func_aes.so module loaded. also i remove and test but still same error. -Satish == Using SIP RTP CoS mark 5 -- Executing [7623 at from-sip:1] Macro("SIP/7527-0000000d", "orasebcamdial,7623") in

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!!!!!!!!!!! -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110608/e99afa31/attachment.htm>

Hey Users, I am using record application to record MeetMe conf. but look like its creating individual files for every channel. What applucation is best to record MeetMe conf ? ~ # ls -l /var/spool/asterisk/monitor/ total 489220 -rw-r--r-- 1 asterisk asterisk 44 Feb 16 08:42 8881-conf-20110216-084224.wav -rw-r--r-- 1 asterisk asterisk 1858284 Feb 16 13:05 8881-conf-20110216-130321.wav

I have deployed several 1.8.3.2 systems as upgrades of customers systems and now I am seeing random crashes. For some reason the builds lock up and stop taking sip connections. Existing calls stay on but when the user hangs up no new calls or reg attempts work. In most cases a "core restart now" cleans things up. Some times I have to kill the asterisk process. The stability of 1.8.2

hi: our current connection is below: sip phone<--->asterisk<---->alcatel PBX<---->PSTN asterisk and alcatel PBX is connected via E1 isdn-pri. when I use sip phone to dial outside PSTN world: 1. with 1.4 it is fine. 2. with 1.6.2, I need to set prematuremedia=no is sip.conf. or sip phone can not hear the ring and the beginning of the PSTN voice. 3.

Hi All, I have just latest branch of asterisk 1.8 and i didn't found dahdi command in CLI everything seem fine. am i missing something ? campbx2*CLI> dahdi <tab tab> No such command 'dahdi' (type 'core show help dahdi' for other possible commands) campbx2*CLI> root at campbx1:/etc/wanpipe# wanrouter hwprobe ------------------------------- | Wanpipe Hardware

Hey guys, Currently we have non HWEC sangoma pri card but now we are planing to replace card with HWEC support card for echo cancellation. So in this case do I need to re-install everything? Like zaptel, asterisk etc.. Or just replace the card? -- Sent from my iPhone

Hi All, Anyone out there successfully tested Asterisk 1.8 with Web-Meetme 4.0v in my case my asterisk got crashed when i dialing conf room number. Best, S -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20101215/52082964/attachment.htm>

Hi Guys! I'm middle of testing my asterisk-1.8.3.2 just make sure how much call it could handle in production so following is my senario. [sipp_client]---------------[Asterisk]----------------[sipp_server] sipp_client ./sipp -sf uac_pcap.xml -d 100000 -i 172.30.254.211 -s 2000 172.30.1.47 -l 1000 -r 250 -rp 5000 -m 1000 sipp_server ./sipp -sn uas -i 172.30.245.208 In above if i set -r

Package: xcp-storage-managers Version: 0.1.1-3 Severity: normal This was already reported in the archived bug 691805 but I sent an e-mail and I received back permanent delivery failure because this bugs has been archieved. Still the path of "pidof" in the file /usr/lib/xcp/sm/iscsilib.py is wrong on Wheezy. It should be /bin/pidof instead of /sbin/pidof root at xcpcompute2:~# cat

How to get rid on following.. why its Invalid ? holler*CLI> queue show queue1 queue1 has 0 calls (max unlimited) in 'rrmemory' strategy (0s holdtime, 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s Members: Agent/7201 (Invalid) has taken no calls yet Agent/7202 (Invalid) has taken no calls yet No Callers -------------- next part -------------- An HTML