Phone Doesnt Hangup After Logout

[prince.fer]

Hi,

I am using vicidialnow-1.2 which is having astguiclient-2.0.5. When i m login to agent it is giving me incoming ring and when i accept it is not saying ur the online person in conference.

Because whenever a LIVE CALLS are coming i cant hear customer voice. And when i logout my soft phone (eyebeam) is not getting hangup. It is showing call established.

What could be the problem.

thanks

[mflorell]

Asterisk CLI output?

[prince.fer]

below is the output of asterisk cli..


== Manager 'sendcron' logged off from 127.0.0.1
-- Registered SIP 'cc100' at 122.169.88.243 port 24445 expires 3600
-- Saved useragent "eyeBeam release 1013t stamp 43069" for peer cc100
-- Unregistered SIP 'cc100'
-- Registered SIP 'cc100' at 122.169.88.243 port 24445 expires 3600
-- Saved useragent "eyeBeam release 1013t stamp 43069" for peer cc100
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
> Channel SIP/cc100-084a4d48 was answered.
-- Executing MeetMe("SIP/cc100-084a4d48", "8600052|F") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1022 for conference '8600052'
-- 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
-- Executing AGI("Local/441517331419@default-fb93,2", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("Local/441517331419@default-fb93,2", "SIP/00441517331419@netindia|100|tTo") in new stack
-- Called 00441517331419@netindia
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing AGI("Local/441517340757@default-4350,2", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("Local/441517340757@default-4350,2", "SIP/00441517340757@netindia|100|tTo") in new stack
-- Called 00441517340757@netindia
-- SIP/netindia-084acea8 is making progress passing it to Local/441517331419@default-fb93,2
-- SIP/netindia-0851c7c8 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Hangup("Local/441517340757@default-4350,2", "") in new stack
== Spawn extension (default, 441517340757, 3) exited non-zero on 'Local/441517340757@default-4350,2'
-- Executing DeadAGI("Local/441517340757@default-4350,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----1-----CONGESTION----------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
== Manager 'sendcron' logged off from 127.0.0.1
-- SIP/netindia-084acea8 answered Local/441517331419@default-fb93,2
> Channel Local/441517331419@default-fb93,1 was answered.
-- Executing Answer("Local/441517331419@default-fb93,1", "") in new stack
-- Executing AGI("Local/441517331419@default-fb93,1", "agi://127.0.0.1:4577/call_log") in new stack
== Manager 'sendcron' logged off from 127.0.0.1
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing AGI("Local/441517331419@default-fb93,1", "agi-VDAD_ALL_outbound.agi|NORMAL-----LB") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
== Spawn extension (default, 441517331419, 2) exited non-zero on 'Local/441517331419@default-fb93,2'
-- Executing DeadAGI("Local/441517331419@default-fb93,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----6-----0") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---6-----0 completed, returning 0
-- AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
-- Executing AGI("SIP/netindia-084acea8", "agi-VDAD_ALL_outbound.agi|NORMAL-----LB") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
-- AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
-- Executing Goto("SIP/netindia-084acea8", "default|8600052|1") in new stack
-- Goto (default,8600052,1)
-- Executing MeetMe("SIP/netindia-084acea8", "8600052|F") in new stack
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing MeetMe("Local/58600052@default-ed7a,2", "8600052|Fmq") in new stack
> Channel Local/58600052@default-ed7a,1 was answered.
-- Executing Answer("Local/58600052@default-ed7a,1", "") in new stack
-- Executing Monitor("Local/58600052@default-ed7a,1", "wav|20091118-183759_1517331419") in new stack
-- Executing Wait("Local/58600052@default-ed7a,1", "3600") in new stack
== 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
== 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
vici*CLI>


When i reboot the server it works for 3-4 calls and than again silence. means no voice of customer is hear..Even i m not able to hear "u r the only person in conference"

[prince.fer]

My NTP server is working and time is sync..I have set GMT as 5.50 for indian time as my server is in india..

Thanks

[mflorell]

Try using an IAX softphone like Zoiper or KIAX

[prince.fer]

I tried zopier but facing same problem. i m using sip protocol not iax so i cant use iax softphone

what could be the problem.

[mflorell]

The problem could be SIP, try IAX instead.

[mr_mehul_shah]

Try upgrading your asterisk version.....

[williamconley]

Open the port for IAX and use KIAX.

Try without vicidial (make a few calls to your mom, and some high school buddies or something). Make several calls with a fairly long duration and the same server load as you had when using vicidial. You could be experiencing some issues with your bandwidth unrelated to Vicidial.

Also, "3-4 calls then again silence" ... you mean it dials a customer and puts the customer into the conference with you but you cannot hear the customer? (Please verify this, as i've had language issues before and found a description such as this actually indicated that the calls stopped coming in because there were no calls being made ... and you have a congestion notice in your cli ...)

Please also post your entire configuration, and put it into your signature for easy reference. (Including vicidial build from your php screen footers and whether any other software is included in the box and your basic hardware, and in THIS case: bandwidth at the server location and bandwidth at your location ... server is in india, where are you?)

[prince.fer]

Thanks my prob. is solve by upgrading the asterisk..