Conference Call MeetMe fails and hangs up

Debian lenny/sid
FreePBX version: (2.5.1)
Asterisk (Ver. 1.4.22): Summary

Every time I try and join a Meet Me conference call the meetme module fails and the meeting drops

any help with this would be appreciated.

Verbosity is at least 18
– Remote UNIX connection
– Executing [[email protected]:1] Macro(“SIP/100-00a21b30”, “user-callerid|”) in new stack
– Executing [[email protected]:1] Set(“SIP/100-00a21b30”, “AMPUSER=100”) in new stack
– Executing [[email protected]:2] GotoIf(“SIP/100-00a21b30”, “0?report”) in new stack
– Executing [[email protected]:3] ExecIf(“SIP/100-00a21b30”, “1|Set|REALCALLERIDNUM=100”) in new stack
– Executing [[email protected]:4] Set(“SIP/100-00a21b30”, “AMPUSER=100”) in new stack
– Executing [[email protected]:5] Set(“SIP/100-00a21b30”, “AMPUSERCIDNAME=Lance”) in new stack
– Executing [[email protected]:6] GotoIf(“SIP/100-00a21b30”, “0?report”) in new stack
– Executing [[email protected]:7] Set(“SIP/100-00a21b30”, “AMPUSERCID=100”) in new stack
– Executing [[email protected]:8] Set(“SIP/100-00a21b30”, “CALLERID(all)=“Lance” <100>”) in new stack
– Executing [[email protected]:9] Set(“SIP/100-00a21b30”, “REALCALLERIDNUM=100”) in new stack
– Executing [[email protected]:10] ExecIf(“SIP/100-00a21b30”, “0|Set|CHANNEL(language)=”) in new stack
– Executing [[email protected]:11] GotoIf(“SIP/100-00a21b30”, “0?continue”) in new stack
– Executing [[email protected]:12] Set(“SIP/100-00a21b30”, “__TTL=64”) in new stack
– Executing [[email protected]:13] GotoIf(“SIP/100-00a21b30”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,20)
– Executing [[email protected]:20] NoOp(“SIP/100-00a21b30”, “Using CallerID “Lance” <100>”) in new stack
– Executing [[email protected]:2] Set(“SIP/100-00a21b30”, “MEETME_ROOMNUM=400”) in new stack
– Executing [[email protected]:3] GotoIf(“SIP/100-00a21b30”, “0?USER”) in new stack
– Executing [[email protected]:4] Answer(“SIP/100-00a21b30”, “”) in new stack
– Executing [[email protected]:5] Wait(“SIP/100-00a21b30”, “1”) in new stack
– Executing [[email protected]:6] Set(“SIP/100-00a21b30”, “MEETME_OPTS=I”) in new stack
– Executing [[email protected]:7] Goto(“SIP/100-00a21b30”, “STARTMEETME|1”) in new stack
– Goto (from-internal,STARTMEETME,1)
== Spawn extension (from-internal, STARTMEETME, 1) exited non-zero on ‘SIP/100-00a21b30’
– Executing [[email protected]:1] Macro(“SIP/100-00a21b30”, “hangupcall”) in new stack
– Executing [[email protected]:1] ResetCDR(“SIP/100-00a21b30”, “vw”) in new stack
– Executing [[email protected]:2] NoCDR(“SIP/100-00a21b30”, “”) in new stack
– Executing [[email protected]:3] GotoIf(“SIP/100-00a21b30”, “1?skiprg”) in new stack
– Goto (macro-hangupcall,s,6)
– Executing [[email protected]:6] GotoIf(“SIP/100-00a21b30”, “1?skipblkvm”) in new stack
– Goto (macro-hangupcall,s,9)
– Executing [[email protected]:9] GotoIf(“SIP/100-00a21b30”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,11)
– Executing [[email protected]:11] Hangup(“SIP/100-00a21b30”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on ‘SIP/100-00a21b30’ in macro ‘hangupcall’
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on ‘SIP/100-00a21b30’

This will happen if you have compiled asterisk before zaptel.
When compiling asterisk it will check to see if there is a timing source (zaptel), if that is not detected it will not compile meetme application.

Look in /usr/lib/asterisk/modules for a file called app_meetme.so
If that file is missing you will get the above error.

If that file is present then you have no timing source. If your setup is for SIP then check your /etc/sysconfig/zaptel that there is a line like this:

MODULES="$MODULES ztdummy"      # UHCI USB Zaptel Timing Only Interface

If that line is preceded with a ‘#’ remove that character then restart zaptel.

I my issue was that I didn’t have the Zaptel drivers installed.

I guess you need them even if your doing a voip only solution for the conferencing.