I have setup my asterisk box using freepbx. I can call extension and make
outbound calls. the outbound calls drop between 10-30sec. we are using
bandwidth.com and they have logged our call. below is your bad followed by what
they say is a good call. I can't figure out where the problem is on your end. I
know we are missing some stuff at the bottom but I don't know where to start.
**************BAD CALL************************
Wed Aug 5 18:22:28 2009 64.191.130.78:5060 ---> 216.82.224.202:5060
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 64.191.130.78:5060;branch=z9hG4bK20dc2d74;rport
From:"Justin's Face"<sip:[email protected]>;tag=as5d2a3b2a
To:<sip:[email protected]>
Contact:<sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 05 Aug 2009 18:22:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 230
***********GOOD CALL***************************
INVITE sip:[email protected]:5060 SIP/2.0
Record-Route:<sip:4.79.212.229;lr;ftag=VPSF506071629460>
Via: SIP/2.0/UDP 4.79.212.229;branch=z9hG4bK8ec1.70782da5.0
Via: SIP/2.0/UDP 4.68.250.148:5060;branch=z9hG4bK506071629460-1246361886000
From:"HIX INC"<sip:[email protected];isup-oli=0>;tag=VPSF506071629460
To:<sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact:<sip:[email protected]:5060;transport=udp>
Max-Forwards: 68
Content-Type: application/sdp
Content-Length: 173
v=0
o=- 1249498358 1249498359 IN IP4 63.215.29.149
s=-
c=IN IP4 63.215.29.149
t=0 0
m=audio 61030 RTP/AVP 0 18 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
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