vicidial.org

I am having a problem with meetme because it doesn't have any sounds. I mean you should at least hear "you are only the person in this conference" when you dial a conference number but I don't hear anything. Its seems call dialed from webclient but no sound at all.
without webclient manual dialing work fine.


Here is the CLI output

-- Registered SIP '802' at 209.170.108.26 port 9466 expires 3600
-- Saved useragent "X-Lite release 1006e stamp 34025" for peer 802
-- Got SIP response 400 "Bad Request" back from 209.170.108.26
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Got SIP response 400 "Bad Request" back from 209.170.108.26
> Channel SIP/802-08e443c0 was answered.
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing MeetMe("SIP/802-08e443c0", "8600051") in new stack
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing MeetMe("Local/8600051@default-1550,2", "8600051") in new stack
> Channel Local/8600051@default-1550,1 was answered.
-- Executing AGI("Local/8600051@default-1550,1", "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/8600051@default-1550,1", "sip/18888198099@SIPtrunk|55|o") in new stack
-- Called 18888198099@SIPtrunk
-- SIP/SIPtrunk-08e49900 is ringing
-- SIP/SIPtrunk-08e49900 is making progress passing it to Local/8600051@default-1550,1
== Manager 'sendcron' logged off from 127.0.0.1
-- SIP/SIPtrunk-08e49900 is making progress passing it to Local/8600051@default-1550,1
-- SIP/SIPtrunk-08e49900 is making progress passing it to Local/8600051@default-1550,1
-- SIP/SIPtrunk-08e49900 answered Local/8600051@default-1550,1
May 7 12:42:51 NOTICE[10785]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 69.1.229.45
== 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
== Spawn extension (default, 18888198099, 2) exited non-zero on 'Local/8600051@default-1550,1'
-- Executing DeadAGI("Local/8600051@default-1550,1", "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 DeadAGI("Local/8600051@default-1550,1", "agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----27-----18)") in new stack
-- AGI Script agi://127.0.0.1:4577/VD_hangup--HVcause ... -27-----18) completed, returning 0
== Manager 'sendcron' logged off from 127.0.0.1

Hi,

what is your timing device? ztdummy?zaptel card?

I am using zaptel. I have X100P.


kkj

when you dial a conference number what happened?

whats your hardware?
OS?

Thanks its solved. I just compiled zaptel again and reboot the system.

kkj