Page 128 - Asterisk™: The Future of Telephony
P. 128

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
                   Supported: replaces
                   Content-Type: application/sdp
                   Content-Length: 265
               With the fromuser:

                   Audio is at 66.135.99.122 port 11700
                   Adding codec 0x2 (gsm) to SDP
                   Adding codec 0x4 (ulaw) to SDP
                   Adding non-codec 0x1 (telephone-event) to SDP
                   Reliably Transmitting (no NAT) to 10.251.55.100:5060:
                   INVITE sip:15195915119@10.251.55.100 SIP/2.0
                   Via: SIP/2.0/UDP 66.135.99.122:5060;branch=z9hG4bK635b0b1b;rport
                   From: "asterisk" <sip:my_unique_id@66.135.99.122>;tag=as3186c1ba
                   To: <sip:15195915119@10.251.55.100>
                   Contact: <sip:my_unique_id@66.135.99.122>
                   Call-ID: 0c7ad6156f92e70b1fecde903550a12f@66.135.99.122
                   CSeq: 102 INVITE
                   User-Agent: Asterisk PBX
                   Max-Forwards: 70
                   Date: Fri, 20 Apr 2007 15:00:30 GMT
                   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
                   Supported: replaces
                   Content-Type: application/sdp
                   Content-Length: 265
               The deny and permit statements are used to deny all incoming calls to this peer except
               the IP address defined by the permit parameter. This is simply a security measure used
               to make sure nothing else matches on this peer except traffic coming from the IP address
               we expect.
               At the end is insecure=invite, which may be required for your provider. This is because
               the source of the INVITE may originate from its backend platform, but could be di-
               rected through its SIP proxy server. Basically what this means is that the IP address that
               the peer is coming from, and which you are matching on, may not be the IP address
               that is in the Contact line: field of the INVITE message when you are accepting a call
               from your provider. This tells Asterisk to ignore this inconsistency and to accept the
               INVITE anyway.


                           You may need to set invite=invite,port if the port address is also in-
                           consistent with what Asterisk is expecting.




               Now we need one additional parameter set in the [general] section of our sip.conf file:
               register. register is going to tell the service provider where to send calls when it has
               a call to deliver to us. This is Asterisk’s way of saying to the service provider, “Hey! If
               you’ve got a call for me, send it to me at IP address 10.251.55.100.” The register
               parameter takes the following form:



               100 | Chapter 4: Initial Configuration of Asterisk
   123   124   125   126   127   128   129   130   131   132   133