Thanks for the tip, i tried to fix the Voicemail position but realised it was actually going into NOANSWER. So i did a new test that looks something like this:
I still get the same result though, first call i get farther into the message than following ones. (that's after restarting asterisk completely from the init script) [inbound] exten => 5145551212,1,Dial(SIP/spaa&SIP/spab&SIP/spac&SIP/spad&SIP/n9,20,wt) exten => 5145551212,n,Goto(in5145551212-${DIALSTATUS},1) exten => 5145551212,n,Hangup exten => in5145551212-BUSY,1,Voicemail(2102,b) exten => in5145551212-BUSY,n,Hangup(17) exten => in5145551212-CONGESTION,1,Voicemail(2102) exten => in5145551212-CONGESTION,n,Hangup(3) exten => in5145551212-CHANUNAVAIL,1,Voicemail(2102,u) exten => in5145551212-CHANUNAVAIL,n,Hangup() exten => in5145551212-NOANSWER,1,Voicemail(2102,u) exten => in5145551212-NOANSWER,n,Hangup(16) exten => _in5145551212-.,1,Hangup(16) (with level 10 verbosity) -------------------- START FIRST TRY / CALL, AFTER RESTART OF ASTERISK ----------------------- > Saved useragent "Telepathy-SofiaSIP/0.5.18.1 sofia-sip/1.12.10devel" for peer n9 -- Accepting UNAUTHENTICATED call from 206.191.37.138: > requested format = ulaw, > requested prefs = (ulaw|gsm), > actual format = gsm, > host prefs = (), > priority = caller -- Executing [5145551...@unlimitel-inbound:1] Dial("IAX2/206.191.37.138:4569-13", "SIP/spaa&SIP/spab&SIP/spac&SIP/spad&SIP/n9,20,wt") in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 == Using SIP VRTP TOS bits 136 == Using SIP VRTP CoS mark 4 -- Called spaa == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 == Using SIP VRTP TOS bits 136 == Using SIP VRTP CoS mark 4 -- Called spab == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 == Using SIP VRTP TOS bits 136 == Using SIP VRTP CoS mark 4 -- Called spac == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 == Using SIP VRTP TOS bits 136 == Using SIP VRTP CoS mark 4 -- Called spad == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 == Using SIP VRTP TOS bits 136 == Using SIP VRTP CoS mark 4 [Mar 31 10:58:36] WARNING[6383]: app_dial.c:1745 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 == Using SIP VRTP TOS bits 136 == Using SIP VRTP CoS mark 4 -- Called n9 -- SIP/spaa-00000000 is ringing -- SIP/spac-00000002 is ringing -- SIP/spab-00000001 is ringing -- SIP/spad-00000003 is ringing -- SIP/n9-00000004 is ringing -- Got SIP response 480 "Terminated" back from 192.168.1.129 -- SIP/n9-00000004 is circuit-busy -- Nobody picked up in 2000 ms -- Executing [5145551...@unlimitel-inbound:2] Goto("IAX2/206.191.37.138:4569-13", "in5145551212-NOANSWER,1") in new stack -- Goto (unlimitel-inbound,in5145551212-NOANSWER,1) -- Executing [in5145551212-noans...@unlimitel-inbound:1] VoiceMail("IAX2/206.191.37.138:4569-13", "2102,u") in new stack -- <IAX2/206.191.37.138:4569-13> Playing '/var/spool/asterisk/voicemail/default/2102/unavail.gsm' (language 'fr') -- <IAX2/206.191.37.138:4569-13> Playing 'vm-intro.gsm' (language 'fr') == Spawn extension (unlimitel-inbound, in5145551212-NOANSWER, 1) exited non-zero on 'IAX2/206.191.37.138:4569-13' -- Hungup 'IAX2/206.191.37.138:4569-13' -------------------- END FIRST TRY / CALL, AFTER RESTART OF ASTERISK ----------------------- The second call looks the same but cuts after "Bonjour" (welcome) instead of farther into the voicemail message. In both case I didn't get any beep... cheers! -- To UNSUBSCRIBE, email to debian-bugs-dist-requ...@lists.debian.org with a subject of "unsubscribe". Trouble? Contact listmas...@lists.debian.org