LONG -> Day troubleshooting vicidialnow
Posted: Sun Jun 21, 2009 10:29 am
HI, ive installed vicidialnow ce 1.2 with astguilient 2.0.5 released.
i Have intel xeon dual core processor and 4 GB ram with 250 SATA drive on my vicidialnow server hardware.
having setup around 6 client PC to be used for dialing using softphones eyebeam.And using a leased line connection 1Mbps.
My network was setup to separate voice and data traffic having another DSL connection mainly intended for browsing for campaign specific inquiry while my softphones routed to my vicidialnow/asterisk dialer and data routed to my modem/router.
vicidialnow/asterisk server have 2 set of NIC assigned for wan connection and for private LAN, firewall is set off.( note: all the cable termination are done on my core switch-which are cables coming from vicidialnow/asterisk server, Client PC, DSL modem/router)
testing 3 SIP VOIP provider ( using IP based and one with username & password).
Using codec supported by voip provider alaw, ulaw, g729. But mainly for my local SIP softphones client setting i used ulaw & alaw.
Problem:
1.Softphones register with equivalent phone ID and users to my vicidial server. but when login in to the agent interface softphones recieves a ring and answer but cant hear a welcome conference voice but what happen is the conference voice out is delayed it will appear on during the calling process, out of 6 agent 2 callers can only recieve welcome conf voice.but on the asterisk CLI log i can see that phone ID logins to the meetme conf. and agents able to execute calls.
2.During the calling process tested with predictive and manual dialing agents cant hear voice or any activity on the call, sometimes can heaar rings and but mostly cant, no voice form client even when on live call-tested with numbers of livecall-all have coice issues. Asterisk logs display no signs of error- just the calling process.test with 3 voip provider
2. Among the 3 voip tested- dialing through manual using eyebeam registered to my vicidialnow/asterisk server, i dial 91 + number of the client.
Result: callers able to hear the client and also client able to hear the caller person.clear voice and line.
3. Turn off comfort noise log prompt always appear on the asterisk CLI log
SUMMARY: manual dialing softphones voice on both parties are fine, but using the agents inetrface like predicitve dialing and manual dial clicking dial next number. voice issues cant hear anything no audio ouput.
Tested with reloaded vicidianow for about 3 times. same vocie issue problem with vicidial. 4 days on torubleshooting no signed of progress.
any suggestion deeply appriciated
i Have intel xeon dual core processor and 4 GB ram with 250 SATA drive on my vicidialnow server hardware.
having setup around 6 client PC to be used for dialing using softphones eyebeam.And using a leased line connection 1Mbps.
My network was setup to separate voice and data traffic having another DSL connection mainly intended for browsing for campaign specific inquiry while my softphones routed to my vicidialnow/asterisk dialer and data routed to my modem/router.
vicidialnow/asterisk server have 2 set of NIC assigned for wan connection and for private LAN, firewall is set off.( note: all the cable termination are done on my core switch-which are cables coming from vicidialnow/asterisk server, Client PC, DSL modem/router)
testing 3 SIP VOIP provider ( using IP based and one with username & password).
Using codec supported by voip provider alaw, ulaw, g729. But mainly for my local SIP softphones client setting i used ulaw & alaw.
Problem:
1.Softphones register with equivalent phone ID and users to my vicidial server. but when login in to the agent interface softphones recieves a ring and answer but cant hear a welcome conference voice but what happen is the conference voice out is delayed it will appear on during the calling process, out of 6 agent 2 callers can only recieve welcome conf voice.but on the asterisk CLI log i can see that phone ID logins to the meetme conf. and agents able to execute calls.
2.During the calling process tested with predictive and manual dialing agents cant hear voice or any activity on the call, sometimes can heaar rings and but mostly cant, no voice form client even when on live call-tested with numbers of livecall-all have coice issues. Asterisk logs display no signs of error- just the calling process.test with 3 voip provider
2. Among the 3 voip tested- dialing through manual using eyebeam registered to my vicidialnow/asterisk server, i dial 91 + number of the client.
Result: callers able to hear the client and also client able to hear the caller person.clear voice and line.
3. Turn off comfort noise log prompt always appear on the asterisk CLI log
SUMMARY: manual dialing softphones voice on both parties are fine, but using the agents inetrface like predicitve dialing and manual dial clicking dial next number. voice issues cant hear anything no audio ouput.
Tested with reloaded vicidianow for about 3 times. same vocie issue problem with vicidial. 4 days on torubleshooting no signed of progress.
any suggestion deeply appriciated