In my office have setuped the Elastix machine and i have a static IP(external IP given by ISP), now the issue is that whenerve call from outside sip extensions which is register to the sip server , am not able hear audio from both side. both callee ..
i am trying to install asterisk newer version on the Elastix machine, but while installing the chan_sip,c module is not building while make. when i seein make menuselect options it showing XXX — extended , please let me know how to enable it and m..
i am using elastix 2.3 and created some dahdi extensions,now i dialing between the extensions i.e like 2000 to 2001 , but there is delay of 3 to 4second before it ring the destination. so cany anyone know how fix it so that after dialing the digits ..
can anyone help me how to setup a simple gateway for voip phones on elastix system. I dnt no really how it should be connected in reality…? and how to test it .Regard..
just going through the code i found that thisCONFIG_VOICEBUS_ECREFERENCEundef in the voicebus then how it will defined to run the echo cancell on the respectivedrievers wctdm24xxp ?? explain how thisCONFIG_VOICEBUS_ECREFERENCEenabled and where it..
Can any one tell me on which linux kernel version i can compile and run the DAHDI-2.0 release and test it .*Regards..
i am trying to add my own sound file in the Asterisk dial plan extension for playback option , i dont no where to put the file and how to give the path in extension file and all so is need that the sound file should be convert in asterisk as .wav file???rega..
i am trying to install the Asterisk test suite on my ubuntu system , i have followed all the installlation steps as mentioned in the link (http://blogs.asterisk.org/2010/04/29/installing-the-asterisk-test-suite/) , but when i am trying to run the scr..
Can anyone tell me how to do the load test for the FXS and FXO cards and find how much the asterisk machine can loadfor different processors configuration .Regard..
I am trying to write dial plan for sip to auto answer (auto attend) the incoming call to the sip phone.- If i call from sip1 to sip2 then sip2 should automatically answer the call and play some sound file. I am trying to do this but as new to the aster..