[Serusers] Problem integrating ser + asterisk voicemail

[Daniel Poulsen]

Also, if you are coming through a gateway make sure the gateway is equipped 
to handle dtmf. On cisco you dial peer should look something like this:

dial-peer voice 10 voip
application session
destination-pattern .T
progress_ind setup enable 3
rtp payload-type nte 98
voice-class codec 1
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
ip qos dscp cs5 media
! 


On 7/12/05, Iqbal <iqbal at gigo.co.uk> wrote:
> 
> 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 <http://magnum.test.net>") { ## 
> This assumes
> >that the caller
> > log(1, "Forwarding to PSTN\n");
> >## is registered in our realm
> > forward(10.10.10.3 <http://10.10.10.3>, 5060);
> >## Our Cisco router
> > break;
> > };
> >
> > # retrieve voicemail
> > #
> > if (uri=~"^sip:2[0-9]*@magnum.test.net <http://magnum.test.net>") {
> > log(1, "Retrieving voicemail\n");
> >
> > # redirect now!
> > rewritehostport("202.125.25.102:5061 <http://202.125.25.102:5061>");
> > append_branch();
> > t_relay_to_udp("202.125.25.106 <http://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 <http://202.125.25.102:5061>");
> > append_branch();
> > t_relay_to_udp("202.125.25.106 <http://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 <http://0.0.0.0> ; IP address to bind to (0.0.0.0<http://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 <http://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 at test.net
> >
> >_______________________________________________
> >Serusers mailing list
> >serusers at lists.iptel.org
> >http://lists.iptel.org/mailman/listinfo/serusers
> >
> >.
> >
> >
> >
> 
> _______________________________________________
> Serusers mailing list
> serusers at lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
>
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