Hi Mike, It looks correct.
I've found that asterisk doesn't tell you much when a call fails to go through -- what does ethereal/tcpdump see if you sniff port 5060 traffic? On 1/18/06, Mike C. Fletcher <[EMAIL PROTECTED]> wrote: > I'm looking at implementing a feature that will allow a user to click on > a telephone number displayed on a web page and have a call set up such > that we call the user first, and when/if they pick up, we connect them > to the number on which they have clicked. At the moment I'm writing the > code on an asterisk server (behind a NAT) registered with a SER server. > > I've set up the sip.conf with the following: > > [testout] > type=peer > secret=ThePasswordForTheAccount > username=20037 > fromuser=20037 > fromdomain=aci.on.ca > host=aci.on.ca > callerid=Asterisk <[EMAIL PROTECTED]> > nat=yes > insecure=invite > > and am producing call files in spool/asterisk/outgoing, that look like this: > > Channel: SIP/[EMAIL PROTECTED] > MaxRetries: 0 > RetryTime: 60 > WaitTime: 30 > > Context: outgoing > Extension: s > Priority: 1 > > This causes asterisk to send out SIP INVITE messages, but they aren't > being acknowledged by the MTA registered as 20007 on the SER server. > So, I'm wondering if I've really got the right format for the file > (particularly the channel specification, (for which I didn't really find > a good SIP example for the outgoing operation)). We won't be using SIP > channels for this in the final deployment, but I would like to know how > to make these channels work for outgoing calls. > > It's generating invites like this: > > INVITE sip:[EMAIL PROTECTED] SIP/2.0 > Via: SIP/2.0/UDP 72.137.18.234:5060;branch=z9hG4bK5df94db8;rport > From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as392fd208 > To: <sip:[EMAIL PROTECTED]> > Contact: <sip:[EMAIL PROTECTED]> > Call-ID: [EMAIL PROTECTED] > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Date: Tue, 17 Jan 2006 22:38:50 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Content-Type: application/sdp > Content-Length: 265 > > v=0 > o=root 20269 20269 IN IP4 72.137.18.234 > s=session > c=IN IP4 72.137.18.234 > t=0 0 > m=audio 18736 RTP/AVP 3 0 8 101 > a=rtpmap:3 GSM/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > (NAT) to 205.207.148.251:5060 > > > Anyway, if anyone's done a SIP-channel outgoing call and can spot the > obvious error it would be appreciated. > > Have fun, > Mike > > -- > ________________________________________________ > Mike C. Fletcher > Designer, VR Plumber, Coder > http://www.vrplumber.com > http://blog.vrplumber.com > > > > --------------------------------------------------------------------- > To unsubscribe, e-mail: [EMAIL PROTECTED] > For additional commands, e-mail: [EMAIL PROTECTED] > > -- | First they ignore you, then they laugh at you, then they fight you, | then you win. -- Mohandas Gandhi | | The Toronto Asterisk Users Group | Join the discussion group by visiting http://taug.ca | or by sending email to [EMAIL PROTECTED]
