similar to: Bring India together

On 8/7/07, raviprakash sunkara <sunkara.raviprakash.feb14 at gmail.com> wrote: > > Hello Russell, > Nice To meet U and Good Morning. I got u r mail-Id from > http://www.asterisk.org/node/48325 > Recently i started the SLA configuration. But i didn't understand the > Flow of its Functionality > One of the My Client Ask to have do deploySLA feature > He Using

Hello Users, I Installed the Asterisk-1.2.11, For My Real time Use I'm Use MySql For Asterisk Database, By Using the Asterisk-addons -1.2.4 in My Linux. For My Voice messages Storage , I want To Use the MySql. In Googled it shows me the ODBC integration.. Is it need for that ODBC integration with MySql for my Voice Message storing in MySql. When I'm trying to integrate with ODBC +

Hello Users, I googled Call Snooping, its shows only the it means, But i didn't find How to dialplan the Call Snooping, I seen that " What is Trixbox " in Asterisk I Use only some Feature in Asterisk (20), Is it need Asterisk to install the TrixBox in that same System where i installed the Asterisk Server Help me please :P -- Thanks and Regards Ravi Prakash Sunkara

Hello Users....... I'm Strucked in Call parking... I'm Using the Asterisk-1.1.11 version in My FC5 box, In That there is feature.conf I'm Using SIP channel By using Asterisk + OpenSER [general] parkext => 9006 ; What extension to dial to park parkpos => 9007-9009 ; What extensions to park calls on. These needs to be

Hi Users, I'm Setting UP the Voicemails by integrating with Asterisk and OpenSER, After 32 sec or 6 ring, it has to go the Voicemail server of Asterisk, In openser.cfg ........... is not hiiting the Asterisk server ............. ... any one help me ........ .... .... modparam("tm","fr_timer",6) modparam("tm","fr_inv_timer",24)

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/ " context. But For My Real Time and User interface developing , I want to Store in Database, As per My Knowledge and Googled ,

Hello Users, Is Without Zaptel interface Installed, conference Bridge is worked or not. Why it need, For SIP conferences through OpenSER.... Please Help me For me its Giving Some Errors and warnings. *== Parsing '/etc/asterisk/meetme.conf': Found Oct 25 18:16:13 WARNING[12281]: chan_zap.c:913 zt_open: Unable to open '/dev/zap/pseudo': No such file or directory Oct 25

I have a fairly simply laid out network that could easily use static routing. However, I plan to expand quickly and don''t wish to spend the better part of my life maintaining static routes. INTERNET | --------- Gateway | - LAN 172.16.1.x | - LINK 192.168.1.1 --------- |<----wifi --------- router 2 | -LAN 172.16.2.x | -Link 192.168.1.2 ---------

Hello Users, I doing on Voicemail in Asterisk For my RealTime, By using the ODBC connectivity For Voicemessages. in Made the Change in res_odbc.conf, odbc.ini, odbcinst.ini and voicemail.conf When I start My Asterisk server it give me Some Warning, When I googled , a proper Docummentation is not found, it found in some there languages, the First Warning is. Warning [30188]

Hello Users I'm Doing working on Both OpenSER and Asterisk ....... 9001 and 9003 are registered in OpenSER in extension.conf [from-sip] exten=>115,1,Park() exten =>115,2.Hungup() in Feature.conf ( default park no 701) in sip.conf [9001] ... .. [9002] [9003] When 9003 dial the 115 ( Parking itself) , Asterisk Server says " U parked on 701 extension " After When 9001 dial

Hello Users......... I'm Now doing on Asterisk Manager for My knowledge Growth, Can anybody explan me on Asterisk Manager settings....... in manager.conf [general] enabled =yes port = 5038 bindaddr = 192.168.2.75 displayconnects = yes [hyperion] secret = hyperion permit=192.168.2.76/255.255.255.0 deny=0.0.0.0/0.0.0.0 read = system,call,log,verbose,command,agent,user write =

Hello Users I'm planning to do Call Barging and Call snooping , I saw this Feature in asterisk.org. This Barging and Snooping are test for " is Agents are replying the Answer or not " that I'm guessing Can anybody help me... this Feature ...... How to do Call Barging and snooping in SIP Channels , I;m not using any Zaptel Card -- Thanks and Regards Ravi Prakash Sunkara

hi all i am submitting form and i take valu from two text fields. my controller code is like that def saleinsert @verkaufsliste = Verkaufsliste.create(params[:bestandsliste]) @listes = Verkaufsliste.create(params[:verkaufsliste.to_s]) @verkaufsliste=Verkaufsliste.create(@listes.vkdatum.to_s) if @verkaufsliste.save redirect_to :action => ''list'' else

I am trying to configure Asterisk for BSNL, india network. I have successfully configured it for outgoing calls. When any outside number make any call to trunk then it receives the call properly but when the call is disconnected by inside extension then outside phone does not get a busy tone. Asterisk incoming call log: -- Executing [s at incoming:2] Dial("Zap/4-1", "Zap/1")

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Hello The National Institute for Public Finance and Policy is looking to hire someone for the month of March 2009. The project is in time series econometrics (threshold cointegration with package tsDyn) and involve computer intensive methods, so external help is needed for overall code optimisation, involving: - parallel work (set-up of a small cluster), -rewritting pieces of code from R to

Good morning, I've been having some problems getting the SPA-3102 working properly in India. Specific problem is that calls from the Asterisk server out the FXS port is failing. When trying to make calls, I'm getting this message: [Jan 26 23:00:31] NOTICE[2136]: chan_sip.c:13774 handle_request_invite: Call from '' to extension '66200' rejected because extension not found.

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