We are trying to set up the Asterisk to work as a voice mail server with the Metaswitch acting as the SIP proxy/registrar.
The initial extensions.conf file was configured with
;exten => 7204449999,1,Goto,vmail|s|1 ;when a call comes in for 7204449999, goto [vmail], extension s, first step.
Is this correct?
With this in the extensions.conf, I see that the MS is sending out 4 invite messages but the Asterisk does not respond. However, 10 seconds later, the asterisk seems to be sending out options messages to the Metaswitch
I also tried changing the extensions.conf file to
exten => 7204449999,1,Goto(vmail,s,1) ;when a call comes in for 7204449999, goto [vmail], extension s, first step.
I also turned sip debugging on and notice the following error
[Jun 20 14:25:59] WARNING[2655]: chan_sip.c:3778 retrans_pkt: Maximum retries exceeded on transmission C0C99200@metaswitch for seqno 610297763 (Critical Response) – See doc/sip-retransmit.txt.
[Jun 20 14:25:59] WARNING[2655]: chan_sip.c:3805 retrans_pkt: Hanging up call C0C99200@metaswitch - no reply to our critical packet (see doc/sip-retransmit.txt).
[Jun 20 14:25:59] WARNING[2918]: app_voicemail.c:9100 vm_authenticate: Unable to read password
Really destroying SIP dialog ‘C0C99200@metaswitch’ Method: INVITE
Any pointer/ suggestions ?
Thnaks in Advance