Conference Hangs Up - DAHDI 2.6 and Asterisk 1.6.22

Hi All!

I’m really scratching my head on this one. My current setup is:

Asterisk 1.6.22
DHADI 2.6 Complete
LibPRI 1.4.12
FreePBX 2.8.1.4

I’ve setup a meetme room in FreePBX, but no matter what I do (followed steps 1 - 6 of this document: http://www.freepbx.org/support/documentation/faq/common-problems/invalid-conference ) the conference always hangs up! I used this tutorial to recompile both Asterisk and Dahdi (http://astbook.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/installing_how_to_install_it.html)

I’ve recompiled both Asterisk and Dahdi to the current versions of 1.6, I’ve reformatted the server numerous times and I’m still stuck.

This is what happens when I dial into the conference from an extension:

== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Executing [[email protected]:1] Macro(“SIP/2610-00000002”, “user-callerid,”) in new stack
– Executing [[email protected]:1] Set(“SIP/2610-00000002”, “AMPUSER=2610”) in new stack
– Executing [[email protected]:2] GotoIf(“SIP/2610-00000002”, “0?report”) in new stack
– Executing [[email protected]:3] ExecIf(“SIP/2610-00000002”, “1?Set(REALCALLERIDNUM=2610)”) in new stack
– Executing [[email protected]:4] Set(“SIP/2610-00000002”, “AMPUSER=2610”) in new stack
– Executing [[email protected]:5] Set(“SIP/2610-00000002”, “AMPUSERCIDNAME=Christina G”) in new stack
– Executing [[email protected]:6] GotoIf(“SIP/2610-00000002”, “0?report”) in new stack
– Executing [[email protected]:7] Set(“SIP/2610-00000002”, “AMPUSERCID=2610”) in new stack
– Executing [[email protected]:8] Set(“SIP/2610-00000002”, “CALLERID(all)=“Christina G” <2610>”) in new stack
– Executing [[email protected]:9] GotoIf(“SIP/2610-00000002”, “0?continue”) in new stack
– Executing [[email protected]:10] Set(“SIP/2610-00000002”, “__TTL=64”) in new stack
– Executing [[email protected]:11] GotoIf(“SIP/2610-00000002”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,18)
– Executing [[email protected]:18] Set(“SIP/2610-00000002”, “CALLERID(number)=2610”) in new stack
– Executing [[email protected]:19] Set(“SIP/2610-00000002”, “CALLERID(name)=Christina G”) in new stack
– Executing [[email protected]:20] NoOp(“SIP/2610-00000002”, “Using CallerID “Christina G” <2610>”) in new stack
– Executing [[email protected]:2] Set(“SIP/2610-00000002”, “MEETME_ROOMNUM=2965”) in new stack
– Executing [[email protected]:3] Set(“SIP/2610-00000002”, “MAX_PARTICIPANTS=0”) in new stack
– Executing [[email protected]:4] Set(“SIP/2610-00000002”, “MEETME_MUSIC=none”) in new stack
– Executing [[email protected]:5] GotoIf(“SIP/2610-00000002”, “0?USER”) in new stack
– Executing [[email protected]:6] Answer(“SIP/2610-00000002”, “”) in new stack
– Executing [[email protected]:7] Wait(“SIP/2610-00000002”, “1”) in new stack
– Executing [[email protected]:8] Set(“SIP/2610-00000002”, “MEETME_OPTS=oqMs”) in new stack
– Executing [[email protected]:9] Playback(“SIP/2610-00000002”, “custom/join-conference-2012”) in new stack
– <SIP/2610-00000002> Playing ‘custom/join-conference-2012.slin’ (language ‘en’)
– Executing [[email protected]:10] Goto(“SIP/2610-00000002”, “STARTMEETME,1”) in new stack
– Goto (from-internal,STARTMEETME,1)
– Executing [[email protected]:1] ExecIf(“SIP/2610-00000002”, “1?Set(CHANNEL(musicclass)=none)”) in new stack
– Executing [[email protected]:2] Set(“SIP/2610-00000002”, “GROUP(meetme)=2965”) in new stack
– Executing [[email protected]:3] GotoIf(“SIP/2610-00000002”, “0?MEETMEFULL,1”) in new stack
– Executing [[email protected]:4] MeetMe(“SIP/2610-00000002”, “2965,oqMs,”) in new stack
== Parsing ‘/etc/asterisk/meetme.conf’: == Found
== Spawn extension (from-internal, STARTMEETME, 4) exited non-zero on ‘SIP/2610-00000002’
– Executing [[email protected]:1] Macro(“SIP/2610-00000002”, “hangupcall”) in new stack
– Executing [[email protected]:1] GotoIf(“SIP/2610-00000002”, “1?skiprg”) in new stack
– Goto (macro-hangupcall,s,4)
– Executing [[email protected]:4] GotoIf(“SIP/2610-00000002”, “1?skipblkvm”) in new stack
– Goto (macro-hangupcall,s,7)
– Executing [[email protected]:7] GotoIf(“SIP/2610-00000002”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,9)
– Executing [[email protected]:9] Hangup(“SIP/2610-00000002”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on ‘SIP/2610-00000002’ in macro ‘hangupcall’
== Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/2610-00000002’

I am so lost because I have recompiled and reinstalled both Dahdi and Asterisk, and I’ve also made sure that Dahdi is running, as well as Asterisk. Both are running fine!

Please, any help would be much appreciated. I’ve been working on this for a week and still nothing :frowning:

I checked the help files in Asterisk to find out why this was happening. I then issued dahdi restart… voila! conference works!