search for: talk2ram

I try to configure that only registered sips can make calls. How can I do that? I was looking in sip.conf but I didn?t found wath opition configure this functionality. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071212/cfed2687/attachment.htm

Hello List. I just installed a2billing with asterisk 1.6 and got it working. The only problem is that I'm trying to setup something to manage who's using the most minutes in the house. I noticed a2billing only works for callin cards setups, or maybe I didn't configure it correctly for what I want. Can I use a2billing for "?VoIP residential services"? if yes, how? if no,

...everything is a > kludgy hack, and the UI is less than nice (if you are into UIs). > > I suggest you continue on your own custom development if you have the > time. Check out Aheeva for inspiration. > > Thanks, > Steve Totaro > > On Fri, Oct 17, 2008 at 1:31 AM, ram <talk2ram at gmail.com> wrote: > > look at Vicidial > > > > ram > > > > On Thu, Oct 16, 2008 at 4:46 PM, yavuz yildirim <yvzyldrm at gmail.com> > wrote: > >> > >> hi everybody > >> > >> This is Yavuz YILDIRIM > >> > &gt...

Hi iam not able to start asterisk give me following error any help STARTING ASTERISK /usr/sbin/safe_asterisk: line 42: 4633 Illegal instruction (core dumped) ${ASTSBINDIR}/asterisk ${CLIARGS} ${ASTARGS} >&/dev/${TTY} </dev/${TTY} Asterisk ended with exit status 132 Asterisk exited on signal 4. Automatically restarting Asterisk. /usr/sbin/safe_asterisk: line 42: 4637 Illegal

Hi does asterisk act as SIP proxy ?, like SER any documents if does, will be great help ram -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060516/9fce4aca/attachment.htm

...e Record are not store in Voicemessages Table ............................... Help Me.... ---------- Forwarded message ---------- From: raviprakash sunkara <sunkara.raviprakash.feb14@gmail.com> Date: Oct 14, 2006 10:57 AM Subject: Problem in Voice Message Storing............... To: ram <talk2ram@gmail.com> Hello Ram Good Morning, I need a small help from U on Regarding the Asterisk , Currently I'm Doing Voice Mail in Asterisk which is forwarded By OpenSER. I can Leave the Voice message to the Caller , But Stores in this Directory " /var/spool/asterisk/voicemail/ "...

Hey all, anyone know the status of disconnect supervision on POTS lines in India? Set up an asterisk box, TDM cards, in Mumbai, and doesn't seem to have disconnect supervision...... Thanks -- Chris Earle System Solutions Specialist

Hi i have 20 people calling agents calling when ever they calling i get this below error May 30 00:46:57 WARNING[2840]: channel.c:785 channel_find_locked: Avoided deadlock for '0x8b2f50', 10 retries! and the voice go choppy, and voice breakages iam using Latest SVN, any suggestion to come over this problem ram -------------- next part -------------- An HTML attachment was scrubbed...

Hi all what is the difference between show channels sip show channles i see the difference in both show channels show me 30 channels sip show channels shows me 221 channels any description on this ram -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070909/6dd07e17/attachment.htm

...recx, voip call recording and monitoring. www.orecx.com Thanks & Regards, Vidura Senadeera, Sri Lanka. Tel - +94114520001 Mobile - +94777766596 yahoo/skype Ids - vidurased > ------------------------------ > > Message: 17 > Date: Fri, 30 Nov 2007 08:58:41 +0530 > From: ram <talk2ram at gmail.com> > Subject: Re: [asterisk-users] Newb Question > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users at lists.digium.com> > Message-ID: > <b74751490711291928n692877c3w84c9d03f49b00bf8 at mail.gmail.com> &...

Hi guys, I want to integrate with two asterisk servers a kamailio sip server. Any of you know some good tutorial for this? Thanks in advance! Regards. -- jabber: triptik at 12jabber.com blog: http://impresionesdeunloco.wordpress.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi iam keep getting this log in my asterisk log is this harm anything, and how can stop this, any suggestions Jun 4 18:21:47 DEBUG[2093] chan_sip.c: Stopping retransmission on '45629314783bd11604363618632f07b9@201.x.x.x' of Request 102: Match Found Jun 4 18:21:48 DEBUG[2173] manager.c: Manager received command 'Command' Jun 4 18:21:48 DEBUG[2173] manager.c: Manager received

Hi all, I need to connect two asterisk server in same network and i'm using sip user as my clients...... plz anyone suggest me.... Regards, Thiru -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061227/aa4e409c/attachment.htm

Hi all I have installed AAH 2.2 in my P4 PC following AAH handbook PDF and http://mundy.org/blog/index.php?p=62#amp and made as per the guide says and downloaded SJ Phone, and registered user and when i try to dial the 19197543700 i get message that, all circuits are busy now, please try your call later and when i see in the console i get this mesage any help Called easycall/19197543700

Hi iam working with asterisk with mysql Realtime when i have confgured and run the asterisk iam getting the following error i dig all the places for help could not find the results could some one help me what is wrong iam using 1.2.5 on FC4 Mar 20 23:04:52 NOTICE[2054] cdr.c: CDR simple logging enabled. Mar 20 23:04:52 NOTICE[2054] indications.c: Removed default indication country

Hi all I have downloaded asterisk 1.2.8 try to make on RHEL AS 4 i get the following error any clue make[1]: Entering directory `/root/all/asterisk-1.2.8/res' make[1]: Nothing to be done for `all'. make[1]: Leaving directory `/root/all/asterisk-1.2.8/res' make[1]: Entering directory `/root/all/asterisk-1.2.8/channels' gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes

Hi all I have downloaded 1.2.13 installing on my FC5 when iam making, iam getting the following error could some one suggest me the what is the problem make[1]: Entering directory `/root/vici/asterisk-1.2.13/apps' gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=k8

hi all iam using ztdummy driver after my call end , when i look at debug mode in cli i get this errors --- (0 headers 0 lines) Nat keepalive --- -- Reloading module 'chan_agent.so' (Agent Proxy Channel) == Parsing '/etc/asterisk/agents.conf': Found -- Reloading module 'chan_local.so' (Local Proxy Channel) -- Reloading module 'chan_zap.so' (Zapata

Hi I have reading the voiip side i found some document says " The conference bridge runs Ulaw codec by default. If you let people connect with GSM or other codecs, Asterisk will use CPU power to convert audio between codecs " iam using vicidial and meetme for callcenter application. iam geting choppy voice, and voice breaks. iam using connecting VoIP SIP provider using g729 codec,

Hi does any one come acrosss the tool which convert the normal config files of sip.conf, extension.conf...etc will convert automatically to mysql. with with any problems if yes, kindly point me to that toolm which iam looking thanks ram -------------- next part -------------- An HTML attachment was scrubbed... URL: