similar to: Polycom 501 alternate

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

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>

Hi, I am looking at http://www.theschmandts.org/blog/?p=28 to setup missed call notification but i am having issue. following is my dialplan [macro-stdexten] exten => s,1,Dial(${ARG2}) exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER) exten => s-NOANSWER,1,Voicemail(${ARG1},u) ; If

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, Whenever i calling any extension i am getting following WARNING messages do you have any idea they coming from where? -Satish shirley*CLI> == Using SIP RTP CoS mark 5 -- Executing [7623 at from-sip:1] Macro("SIP/7527-00000008", "stdexten,7623,sip/7623&sip/7624") in new stack -- Executing [s at macro-stdexten:1] Dial("SIP/7527-00000008",

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>

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...

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.

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

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:

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

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! 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

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

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

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>

Hi, I've been meaning to fix my non-working paging feature here for a while, and I've just spent the last 5 hours looking at many, many web pages that all say the same thing. I am using Asterisk 1.6.2.18 and Polycom phones, both older (501 with "latest" legacy 3.1.7 firmware) and newer (335 and 650 with latest 3.3.1f). I have changed the correct values in sip.cfg like

please send me the ways to change asterisk port from 5060 to 5061 i need to configure it because we are already using 5060 port in router then we cant use it again we have to configure other sip server so please suggest me a way.......................... On 4/10/11, asterisk-users-request at lists.digium.com <asterisk-users-request at lists.digium.com> wrote: > Send asterisk-users mailing

Hi All, I have the following section in my smb.conf file: [Satish' Hindi Alltime Hits - Gems] path = /mnt/d-drive/Music/Satish' Hindi Alltime Hits Gems browseable = yes public = yes readonly = yes guest ok = yes I am able to browse this share through win2k but not from winNT. But If I change the share

Hey Support, I am planing to implement new page system with asterisk 1.8 we have 200 SIP calls and page() will overkill my system if announce in one shot. so i am planing to record and play page over 50...50...50 chunk.. I am planing to do with .call file for auto calling after record message but i don't know how to call multiple extension ? and how to use page() with .call file for