this is an asterisk problem not a ser one, if you debug the sip channel
in asterisk CLI, and then press the keys are the dtmf signals being
sent/picked up

Iqbal

Yan Yu Lim wrote:

>Hi guys,
>
>I currently have a sip proxy server (sip express router) which
>registers the sip phones. I need to add voice mail capability, i.e.
>sip express router will forward all incoming calls to Asterisk if the
>user does not pick up the call in 15 seconds.
>
>The voicemail recording stops when the user hangs up. However, the
>recording does not end if the user presses the # key, i.e. it is
>ignoring the user input.
>
>Similarly, when the user dials 2102 to access his voice mail, Asterisk
>plays the prompt, but it seems to ignore all the user input keys.
>
>Please kindly advise.
>
>Regards,
>YY
>
>*****************************************************
>Config files
>------------------------------
>1) Ser
>
>---------------------
>ser.cfg (SER)
>---------------------
>
># -- tm params --
># set time for which ser will be waiting for a final response;
># fr_inv_timer sets value for INVITE transactions,
># fr_timer for all others
>modparam("tm","fr_inv_timer",15)
>modparam("tm","fr_timer",10)
>
>       if (uri==myself) {
>
>               if (method=="REGISTER") {
>
>               # attempt handoff to PSTN
>               if (uri=~"^sip:9[0-9]*@magnum.test.net") {      ##  This assumes
>that the caller
>                       log(1, "Forwarding to PSTN\n");
>##  is registered in our realm
>                       forward(10.10.10.3, 5060);
>##  Our Cisco router
>                       break;
>               };
>
>               # retrieve voicemail
>               #
>               if (uri=~"^sip:2[0-9]*@magnum.test.net") {
>                       log(1, "Retrieving voicemail\n");
>
>                       # redirect now!
>                       rewritehostport("202.125.25.102:5061");
>                       append_branch();
>                       t_relay_to_udp("202.125.25.106","5061");
>                       break;
>               };
>
>               # native SIP destinations are handled using our USRLOC DB
>               if (!lookup("location")) {
>                       sl_send_reply("404", "Not Found");
>                       break;
>               };
>
>                timeout occurred ... now to forward to Asterisk's
>voicemail service
>               if(method == "INVITE")
>               {
>                       t_on_failure("1");
>               };
>       };
>       t_relay();
>
># leave voicemail
>#
>failure_route[1] {
>               log(1,"Activating voicemail!!\n");
>               revert_uri();
>
>               # redirect now to Asterisk (on the same machine) !
>               rewritehostport("202.125.25.102:5061");
>               append_branch();
>               t_relay_to_udp("202.125.25.106","5061");
>       }
>
>--------------------
>
>2) Asterisk
>
>------------
>sip.conf
>------------
>
>[general]
>context=test
>port=5061                       ; UDP Port to bind to (SIP standard
>port is 5060)
>bindaddr=0.0.0.0                ; IP address to bind to (0.0.0.0 binds to all)
>srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
>
>; ip phone 1012, registered with SER
>[1012]
>type=friend
>username=1012
>canreinvite=no
>context=test
>mailbox=1012
>host= 203.125.25.106
>nat=no
>dtmfmode=info
>disallow=all
>allow=alaw
>allow=ulaw
>
>-----------------------
>extensions.conf
>-------------------------
>
>[test]
>;leave voice messages
>exten => 1012,1,Voicemail(u1012)
>exten => 1012,2,Hangup
>
>;play voice messages
>exten => 2012,1,VoiceMailMain,1012
>exten => 2012,2,Hangup
>
>-------------------------
>voicemail.conf
>------------------------
>
>[default]
>1012 => 1234, YY, ylim@test.net
>
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>serusers@lists.iptel.org
>http://lists.iptel.org/mailman/listinfo/serusers
>
>.
>
>
>

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