[Serusers] SER Asterisk Voicemail - Message Playback

[Iqbal]

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 at 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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>