vicidial.org

by **ykhan** » Thu Jul 23, 2009 9:21 pm

When I log in, it shows that it plays the "Only person" audio file but I cannot hear it. A minute later the call disconnects. Then when I logout of the agent interface it disconnects from the conference. Following CLI output may help:

Code: Select all: -- Executing MeetMe("SIP/cc100-08225060", "8600051|F") in new stack == Parsing '/etc/asterisk/meetme.conf': Found -- Created MeetMe conference 1023 for conference '8600051' -- Playing 'conf-onlyperson' (language 'en') == Manager 'sendcron' logged off from 127.0.0.1 == Parsing '/etc/asterisk/manager.conf': Found == Manager 'sendcron' logged on from 127.0.0.1 == Parsing '/etc/asterisk/manager.conf': Found == Manager 'sendcron' logged on from 127.0.0.1 == Manager 'sendcron' logged off from 127.0.0.1 == Parsing '/etc/asterisk/manager.conf': Found == Manager 'sendcron' logged on from 127.0.0.1 == Manager 'sendcron' logged off from 127.0.0.1 == Manager 'sendcron' logged off from 127.0.0.1 Jul 23 22:14:45 NOTICE[5093]: chan_sip.c:11742 do_monitor: Disconnecting call 'SIP/cc100-08225060' for lack of RTP activity in 62 seconds -- Hungup 'Zap/pseudo-38882872' == Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/cc100-08225060' -- Executing DeadAGI("SIP/cc100-08225060", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0 == Parsing '/etc/asterisk/manager.conf': Found == Manager 'sendcron' logged on from 127.0.0.1 == Parsing '/etc/asterisk/manager.conf': Found == Manager 'sendcron' logged on from 127.0.0.1 == Manager 'sendcron' logged off from 127.0.0.1 == Parsing '/etc/asterisk/manager.conf': Found == Manager 'sendcron' logged on from 127.0.0.1 == Manager 'sendcron' logged off from 127.0.0.1 == Parsing '/etc/asterisk/manager.conf': Found == Manager 'sendcron' logged on from 127.0.0.1 == Parsing '/etc/asterisk/manager.conf': Found == Manager 'sendcron' logged on from 127.0.0.1 -- Executing MeetMeAdmin("Local/55558600051@default-86a1,2", "8600051|K") in new stack == Parsing '/etc/asterisk/meetme.conf': Found Jul 23 22:15:09 NOTICE[17780]: app_meetme.c:2210 admin_exec: Conference Number not found -- Executing Hangup("Local/55558600051@default-86a1,2", "") in new stack == Spawn extension (default, 55558600051, 2) exited non-zero on 'Local/55558600051@default-86a1,2' -- Executing DeadAGI("Local/55558600051@default-86a1,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0

ViciDial VERSION: 2.0.5-203 BUILD: 90323-1554

by **mflorell** » Fri Jul 24, 2009 6:38 am

Can you try using Zoiper with IAX instead and see if it works any better?

by **ykhan** » Sun Jul 26, 2009 8:38 pm

Zoiper does connect, but the sound in the conference is really choppy.

Code: Select all: -- Accepting AUTHENTICATED call from 69.172.124.11: > requested format = gsm, > requested prefs = (), > actual format = gsm, > host prefs = (gsm), > priority = mine -- Executing MeetMe("IAX2/cc400-4709", "8600051|F") in new stack == Parsing '/etc/asterisk/meetme.conf': Found -- Created MeetMe conference 1023 for conference '8600051' -- Playing 'conf-onlyperson' (language 'en') -- Hungup 'Zap/pseudo-1941546845' == Spawn extension (default, 8600051, 1) exited non-zero on 'IAX2/cc400-4709' -- Executing DeadAGI("IAX2/cc400-4709", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0

If I dial using Zioper directly, the call connects and sound is ok, but after a few seconds of dlay which diminishes over the first 10 seconds of the call.

by **mflorell** » Mon Jul 27, 2009 2:00 am

What zaptel timer are you using?

by **ykhan** » Mon Jul 27, 2009 10:05 am

I am using ztdummy. I have checked all of the network settings as well, but it does not seem that the issue is there. With SIP (Xlite) I only observed one difference between the internal connection and external connection (connecting from a separate ISP than the Vicibox server).

INTERNAL: Voice was choppy just like the Zoiper phone.
EXTERNAL: Call connects but there is no sound and the following CLI output

Code: Select all: -- Executing MeetMe("SIP/cc100-08225060", "8600051|F") in new stack == Parsing '/etc/asterisk/meetme.conf': Found -- Created MeetMe conference 1023 for conference '8600051' -- Playing 'conf-onlyperson' (language 'en') == Manager 'sendcron' logged off from 127.0.0.1 == Parsing '/etc/asterisk/manager.conf': Found == Manager 'sendcron' logged on from 127.0.0.1 == Parsing '/etc/asterisk/manager.conf': Found == Manager 'sendcron' logged on from 127.0.0.1 == Manager 'sendcron' logged off from 127.0.0.1 == Parsing '/etc/asterisk/manager.conf': Found == Manager 'sendcron' logged on from 127.0.0.1 == Manager 'sendcron' logged off from 127.0.0.1 == Manager 'sendcron' logged off from 127.0.0.1 Jul 23 22:14:45 NOTICE[5093]: chan_sip.c:11742 do_monitor: Disconnecting call 'SIP/cc100-08225060' for lack of RTP activity in 62 seconds -- Hungup 'Zap/pseudo-38882872' == Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/cc100-08225060' -- Executing DeadAGI("SIP/cc100-08225060", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0 == Parsing '/etc/asterisk/manager.conf': Found == Manager 'sendcron' logged on from 127.0.0.1 == Parsing '/etc/asterisk/manager.conf': Found == Manager 'sendcron' logged on from 127.0.0.1 == Manager 'sendcron' logged off from 127.0.0.1 == Parsing '/etc/asterisk/manager.conf': Found == Manager 'sendcron' logged on from 127.0.0.1 == Manager 'sendcron' logged off from 127.0.0.1 == Parsing '/etc/asterisk/manager.conf': Found == Manager 'sendcron' logged on from 127.0.0.1 == Parsing '/etc/asterisk/manager.conf': Found == Manager 'sendcron' logged on from 127.0.0.1 -- Executing MeetMeAdmin("Local/55558600051@default-86a1,2", "8600051|K") in new stack == Parsing '/etc/asterisk/meetme.conf': Found Jul 23 22:15:09 NOTICE[17780]: app_meetme.c:2210 admin_exec: Conference Number not found -- Executing Hangup("Local/55558600051@default-86a1,2", "") in new stack == Spawn extension (default, 55558600051, 2) exited non-zero on 'Local/55558600051@default-86a1,2' -- Executing DeadAGI("Local/55558600051@default-86a1,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0

by **mflorell** » Mon Jul 27, 2009 3:03 pm

might be a hardware issue possibly, what is the loadavg when you connect?

what kind of network switch are you using?

by **ykhan** » Mon Jul 27, 2009 3:13 pm

I am using a Belkin Wireless router, with built in 4 poort switch connected through Ethernet to the server. Loadavg is about 0.3 when I connect to 8500051 manually.

Not sure if it should be a HW issue cause the same problem is being described by other users in http://www.vicidial.org/VICIDIALforum/v ... php?t=8608 thread. Could this be issues with the new version or something?

by **mflorell** » Mon Jul 27, 2009 4:01 pm

We have not seen this issue at all on any of the new installs we've done in the last few days so I don't think it's a new version issue.

by **ykhan** » Mon Jul 27, 2009 6:21 pm

I had a previous install of VicidialNOW, but I did not have this issue in that install. It appears to be an issue of timing, cause it only affects the conference. Dialing directly from a softphone to a phone number gives good sound quality. Since I am not sure about the Vicibox installs, what should I do if I want to re-install Asterisk (including the required patches)? Are the patches downloaded during the initial install of ViciBox?

by **ykhan** » Mon Jul 27, 2009 6:51 pm

I reinstalled Asterisk but instead of 1.2.30.4, I installed 1.2.30.2. This still has not fixed the issue.

by **mflorell** » Mon Jul 27, 2009 11:36 pm

The patches are part of the install.

I would recommend trying Asterisk 1.4.21.2 along with zaptel 1.4 to see if that helps.

by **ykhan** » Thu Jul 30, 2009 8:24 am

I am not sure why, but with VicidialNow 1.2 CE install, zttest results are hovering around 99.97% and with Vicibox its -799%. This leads me to believe that its likely that there is a difference in the kernels that is handling Zaptel timing differently.

by **mflorell** » Thu Jul 30, 2009 11:57 am

-799% should not be possible as a zttest result.

by **ykhan** » Thu Jul 30, 2009 2:10 pm

That is odd indeed, I wonder what is causing such a result. I have ordered an X100P card and hope that can resolve these timing issues.

by **mflorell** » Thu Jul 30, 2009 4:26 pm

You should consider a Sangoma VoiceTime module as well, it's USB and much smaller than a PCI card(and no compatibility issues). We have installed dozens of them and they work great.

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