<br>
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:<br>
<br>
dial-peer voice 10 voip<br>
application session<br>
destination-pattern .T<br>
progress_ind setup enable 3<br>
rtp payload-type nte 98<br>
voice-class codec 1<br>
session protocol sipv2<br>
session target sip-server<br>
dtmf-relay rtp-nte<br>
ip qos dscp cs5 media<br>
! <br>
<br><br><div><span class="gmail_quote">On 7/12/05, <b class="gmail_sendername">Iqbal</b> <<a href="mailto:iqbal@gigo.co.uk">iqbal@gigo.co.uk</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
this is an asterisk problem not a ser one, if you debug the sip channel<br>in asterisk CLI, and then press the keys are the dtmf signals being<br>sent/picked up<br><br>Iqbal<br><br>Yan Yu Lim wrote:<br><br>>Hi guys,<br>
><br>>I currently have a sip proxy server (sip express router) which<br>>registers the sip phones. I need to add voice mail capability, i.e.<br>>sip express router will forward all incoming calls to Asterisk if the
<br>>user does not pick up the call in 15 seconds.<br>><br>>The voicemail recording stops when the user hangs up. However, the<br>>recording does not end if the user presses the # key, i.e. it is<br>>ignoring the user input.
<br>><br>>Similarly, when the user dials 2102 to access his voice mail, Asterisk<br>>plays the prompt, but it seems to ignore all the user input keys.<br>><br>>Please kindly advise.<br>><br>>Regards,<br>
>YY<br>><br>>*****************************************************<br>>Config files<br>>------------------------------<br>>1) Ser<br>><br>>---------------------<br>>ser.cfg (SER)<br>>---------------------
<br>><br>># -- tm params --<br>># set time for which ser will be waiting for a final response;<br>># fr_inv_timer sets value for INVITE transactions,<br>># fr_timer for all others<br>>modparam("tm","fr_inv_timer",15)
<br>>modparam("tm","fr_timer",10)<br>><br>> if (uri==myself) {<br>><br>> if (method=="REGISTER") {<br>><br>> # attempt handoff to PSTN<br>
>
if (uri=~"^sip:9[0-9]*@<a href="http://magnum.test.net">magnum.test.net</a>")
{ ## This assumes<br>>that the caller<br>>
log(1, "Forwarding to PSTN\n");<br>>## is registered in our realm<br>>
forward(<a href="http://10.10.10.3">10.10.10.3</a>, 5060);<br>>## Our Cisco router<br>>
break;<br>> };<br>><br>> # retrieve voicemail<br>> #<br>>
if (uri=~"^sip:2[0-9]*@<a href="http://magnum.test.net">magnum.test.net</a>") {<br>>
log(1, "Retrieving voicemail\n");<br>><br>>
# redirect now!<br>>
rewritehostport("<a href="http://202.125.25.102:5061">202.125.25.102:5061</a>");<br>>
append_branch();<br>>
t_relay_to_udp("<a href="http://202.125.25.106">202.125.25.106</a>","5061");<br>>
break;<br>> };<br>><br>>
# native SIP destinations are handled using our USRLOC DB<br>> if (!lookup("location")) {<br>>
sl_send_reply("404", "Not Found");<br>>
break;<br>> };<br>><br>> timeout
occurred ... now to forward to Asterisk's<br>>voicemail service<br>> if(method == "INVITE")<br>> {<br>>
t_on_failure("1");<br>> };<br>> };<br>> t_relay();<br>><br>># leave voicemail<br>>#<br>>failure_route[1] {<br>> log(1,"Activating voicemail!!\n");
<br>> revert_uri();<br>><br>>
# redirect now to Asterisk (on the same machine) !<br>> rewritehostport("<a href="http://202.125.25.102:5061">202.125.25.102:5061</a>");<br>> append_branch();<br>>
t_relay_to_udp("<a href="http://202.125.25.106">202.125.25.106</a>","5061");<br>> }<br>><br>>--------------------<br>><br>>2) Asterisk<br>><br>>------------<br>>sip.conf<br>
>------------<br>><br>>[general]<br>>context=test<br>>port=5061
; UDP Port to bind to (SIP standard<br>>port is 5060)<br>>bindaddr=<a href="http://0.0.0.0">0.0.0.0</a> ;
IP address to bind to (<a href="http://0.0.0.0">0.0.0.0</a> binds to all)<br>>srvlookup=yes
; Enable DNS SRV lookups on outbound calls<br>><br>>; ip phone 1012, registered with SER<br>>[1012]<br>>type=friend<br>>username=1012<br>>canreinvite=no<br>>context=test<br>>mailbox=1012<br>>host=
<a href="http://203.125.25.106">203.125.25.106</a><br>>nat=no<br>>dtmfmode=info<br>>disallow=all<br>>allow=alaw<br>>allow=ulaw<br>><br>>-----------------------<br>>extensions.conf<br>>-------------------------
<br>><br>>[test]<br>>;leave voice messages<br>>exten => 1012,1,Voicemail(u1012)<br>>exten => 1012,2,Hangup<br>><br>>;play voice messages<br>>exten => 2012,1,VoiceMailMain,1012<br>>exten => 2012,2,Hangup
<br>><br>>-------------------------<br>>voicemail.conf<br>>------------------------<br>><br>>[default]<br>>1012 => 1234, YY, <a href="mailto:ylim@test.net">ylim@test.net</a><br>><br>>_______________________________________________
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