similar to: Without ZapTel inferface or Card install , is Conference working or Not

Hello Users, Good Morning, I'm doing on Conference Bridge with Asterisk + OpenSER with CBMySql modules. And I'm not Using the Zapptel Cards. 9001 ----------> dial 19001(conference Users)-------openSER ---------> Asterisk ------------------------------------------------------------------------ *In Extension.conf * [from-sip] exten =>

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)

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'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

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

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 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 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 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

/Hello Users, Good Morning, In Conferemcing How to Disconnect the phone while in between the Conference ..... When *I press the ' # ' key for Disconnecting the Conference.......... Below the Following to shows some Warning, ( in Red Color ) from-sip en *CLI> -- Executing Playback("SIP/9002-08f9feb8", "conf-hasentered") in new stack

/Hello Users, Good Morning, In Conferemcing How to Disconnect the phone while in between the Conference ..... When *I press the ' # ' key for Disconnecting the Conference.......... Below the Following to shows some Warning, ( in Red Color ) from-sip en *CLI> -- Executing Playback("SIP/9002-08f9feb8", "conf-hasentered") in new stack

Hello Users , I Posted to mailing list, No one is replying My issues, My Issue is No Audio when Openser and Asterik integrated in Same LAN , When UAC are Behind the NAT, With out the Asterisk integration Behind the NAT is working Fine. SIP port and RTP ports are forwarded into router to OpenSER System only. openser.cfg listen=192.168.2.11 alias=sip.hyperion.com # Invite Section if ( method==

Hello Users openSER is sip proxy and registrar , Asterisk is as PBX, Conference and Voicemail servers, openSER and Asterisk are in the Same N/w Where As the UAC are in Behind the NAT, When Astetrisk is not integrated , UAC are in Behind the NAT is working, openSER is 192.168.2.5 Asterisk is 192.168.2.6 I'm just use rewritehost to asterisk server, UAC ----> openSER - - - ->

Hello Users, Good AfterNoon to all I'm Mainly focused on OpenSER and Asterisk Integration. I didn't Find any solution of My Question ? Till now I'm doing only communicating OpenSER and Asterisk through SIP Channel only. User in Asterisk can Call to OpenSER and also vice-versa . But My Question ? I have one VoIP Service line from Voyage ( SIP change ), I want, if one of the

Hello Users I have one VoIP service from Packet8 ( SIP protocal ) Packet8-----> Astetisk server ---------> My SIP agents My Sip Agents are in Asterisk Server , I configured.. If any one user in My Asterisk has to Call the Packet8 service providers , How can I configure it. Till now I'm Doing on OpenSER and ASterisk (Voicemail and Confereing ..) But My Asterisk has to connect

I would us Meetme for conferance SIP-->SIP fist. my Meetme.conf: [rooms] conf => 9999 my extensions.conf: exten => 9999,1,MeetMe(9999) But : == Parsing '/etc/asterisk/meetme.conf': Found Jun 16 10:33:22 WARNING[12100]: chan_zap.c:916 zt_open: Unable to open '/dev/zap/pseudo': No such file or directory Jun 16 10:33:22 ERROR[12100]: chan_zap.c:6969 chandup: Unable

Hello, I'm trying to set up a conference room. When I dial it's extension, I get an audible error saying "Not a valid conference room, please try again" followed by a disconnect. I've got debug sip peer 1001 (my X-Lite client) and I see this in the logs: (I'm pretty sure it has something to do with ztdummy, but I dunno... I have the port installed, but I

Upon setting up and configuring the my extension.conf, meetme.conf and following the instruction outlined at this web page: http://www.voip-info.org/wiki-Asterisk+timer+ztdummy I get the following errors when calling the meetme number. Executing Wait("SIP/216.53.118.2-f41196e0", "1") in new stack -- Executing MeetMe("SIP/216.53.118.2-f41196e0",

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