I use VoiceMailMain(@sip.domain.com) but you seem to want to send the mailboix number instead of asking the user to send, try
exten => 9999, 1, VoiceMailMain,s2092
Iqbal
n 7/28/2005, "Aisling" ashling.odriscoll@cit.ie wrote:
Hi Everyone,
Unfortunately when I was upgrading my version of SER to 0.9.0 I deleted my config file which contained code for forwarding to asterisk voicemail.Therefore I am attempting to set up all over again (Doh! I know!) :-( Anyhow SER seems to be correctly forwarding to Asterisk, the user can leave a message if the phone isn't picked up after 3 rings and then the message is sent to the users email account. I want the user to also be able to listen to their voicemails from the end user device. It is my understanding from what I have coded, that user should be able to dial 9999 and when SER receives this it sends it to Asterisk' VoiceMailMain where the user will be asked for an account name and password.
However there are problems when the user dials 9999. The user can hear nothing on the phone. Errors appear on both the SER screen and Asterisk screen so I am not sure what the problem is:
On SER it says:
Warning: sl_send_reply: I won't send a reply for ACK!! Forwarding to asterisk14(27355) contact_parser(): Empty body Parse_contact(): Error while parsing Get_contact_uri: Error while parsing Contact body
On Asterisk it says:
-- Executing VoiceMailMain("SIP/2092-dd89", "2092") in new stack -- Playing 'vm-login' (language 'en') WARNING: app_voicemail.c:3333 vm_execmain: Couldn't read username Spawn extension (test, 9999, 1) exited non-zero on 'SIP/2092-c660'
The relevant parts of my ser.cfg and asterisk config files are shown below.
Many thanks, Aisling.
ser.cfg
#Call Type Processing Section
if(uri==myself){
if(uri=~"^sip:9[0-9]*@serveraddress"){ log(1, "forwarding to asterisk"); rewritehostport("serveraddress:port"); append_branch(); t_relay_to_udp("serveraddress", "port"); break; }
if (method == "INVITE"){ t_on_failure("1"); route(3); break; }
failure_route[1]{ revert_uri(); rewritehostport("serveraddress:port"); append_branch(); t_relay_to_udp("serveraddress", "port"); break(); }
sip.conf
[general] context=test port=5062 bindaddr=0.0.0.0 srvlookup=yes
[2092] type=friend username=2092 canreinvite=no context=test mailbox=2092 host=dynamic nat=no dtmfmode=info disallow=all allow=alaw allow=ulaw
[314] type=friend username=314 canreinvite=no context=test mailbox=314 host=dynamic nat=no dtmfmode=info disallow=all allow=alaw allow=ulaw
extensions.conf
[test]
;leave voice messages exten => 2092, 1, Voicemail(u2092) exten => 2092, 2, Hangup
exten => 314, 1, Voicemail(u314) exten => 314, 2, Hangup
;play voice messages exten => 9999, 1, VoiceMailMain, 2092 exten => 9999, 2, Hangup
exten => 9999, 1, VoiceMailMain, 314 exten => 9999, 2, Hangup
voicemail.conf
[default]
2092 => 1234, 2092, emailaddress 314 => 1234, 314, emailaddress
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