search for: satish_lx

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

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, 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, 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 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! 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 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 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 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, 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

...email (vip killa) > 9. Re: asterisk login to voicemail (satish patel) > 10. Re: IAX2/0.0.29.199 (Paul Belanger) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Fri, 8 Apr 2011 15:55:39 +0000 > From: satish patel <satish_lx at hotmail.com> > Subject: Re: [asterisk-users] IAX2/0.0.29.199 > To: asterisk-users <asterisk-users at lists.digium.com> > Message-ID: <BLU159-w50125FF32BB6ED7ACBCBC390A70 at phx.gbl> > Content-Type: text/plain; charset="iso-8859-1" > > > @Paul - many...

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",

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 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>

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

What is this error ? is this harmful ? *CLI>*CLI> dahdi restart [Mar 23 14:01:06] WARNING[4314]: chan_dahdi.c:17422 process_dahdi: Ignoring any changes to 'userbase' (on reload) at line 23. [Mar 23 14:01:06] WARNING[4314]: chan_dahdi.c:17422 process_dahdi: Ignoring any changes to 'vmsecret' (on reload) at line 31. [Mar 23 14:01:06] WARNING[4314]: chan_dahdi.c:17422

We're looking to purchase new phones for Asterisk. There are a limited number of new Polycom 501's on the market, mostly refurbished available. Can you recommend a replacement phone? What ever model replaced the 501? -Satish -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

Hi, I have PRI configured and up but when i am dialing outside i am not getting any ringback tone but my call is connected. following is my example SIP----------------->PRI ------------> mobile I have set progress=yes in chan_dahdi.conf but still not working if i call inbound from my mobile to internal extension ringing working please help me -S -------------- next

I am reading at http://www.asteriskguru.com/tutorials/queues.html They are using member in both static and dynamic method. member => <technology>/XXXX -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110519/ec8d19e1/attachment.htm>