search for: vpsf506071629460

Displaying 4 results from an estimated 4 matches for "vpsf506071629460".

Could Asterisk be rewriting an incoming invite?
2009 Mar 16
1
Could Asterisk be rewriting an incoming invite?
I'm not getting inbound audio from bandwidth.com. Their engineer said the invite that they're sending me looks like this: INVITE sip:+15129616808 at 67.198.16.18:5060;transport=udp SIP/2.0. Record-Route: <sip:216.82.224.202;lr;ftag=VPSF506071629460>. Record-Route: <sip:4.79.212.229;lr;ftag=VPSF506071629460>. Via: SIP/2.0/UDP 216.82.224.202;branch=z9hG4bK6314.4f7b5d05.0. Via: SIP/2.0/UDP 4.79.212.229;branch=z9hG4bK6314.15486fb6.0. Via: SIP/2.0/UDP 4.68.250.148:5060;branch=z9hG4bK506071629460-1207516720501. From: "BANDWIDTH COM&qu...
Variable Name needed
2009 Dec 02
2
Variable Name needed
...tripped out the IPs <--- Transmitting (no NAT) to:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP;branch=z9hG4bK6631.98b11517.0;received= Via: SIP/2.0/UDP;branch=z9hG4bK6631.f18e9a97.0 Via: SIP/2.0/UDP :5060;branch=z9hG4bK506071629460-1256675031807 Record-Route: <sip:;lr;ftag=VPSF506071629460> Record-Route: <sip:;lr;ftag=VPSF506071629460> From: "BEAUMONT TX" <sip:+14096798092@;isup-oli=0>;tag=VPSF506071629460 To: <sip:+14098383113@>;tag=as4b59d217 Call-ID: DALMGC0520091202194656056692@ CSeq: 1 INVITE User-Agent: Asterisk PBX 1.6.0.6 Allow:...
call drops after a few seconds
2009 Aug 12
2
call drops after a few seconds
...9 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:+19194393536 at 216.82.224.202:5060 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:+18592192438 at 4.68.250.148;isup-oli=0>;tag=VPSF506071629460 To:<sip:+19194393536 at 4.79.212.229:5060> Call-ID: ATLM...
FW: Variable Name needed
2009 Dec 02
0
FW: Variable Name needed
...tripped out the IPs <--- Transmitting (no NAT) to:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP;branch=z9hG4bK6631.98b11517.0;received= Via: SIP/2.0/UDP;branch=z9hG4bK6631.f18e9a97.0 Via: SIP/2.0/UDP :5060;branch=z9hG4bK506071629460-1256675031807 Record-Route: <sip:;lr;ftag=VPSF506071629460> Record-Route: <sip:;lr;ftag=VPSF506071629460> From: "BEAUMONT TX" <sip:+14096798092@;isup-oli=0>;tag=VPSF506071629460 To: <sip:+14098383113@>;tag=as4b59d217 Call-ID: DALMGC0520091202194656056692@ CSeq: 1 INVITE User-Agent: Asterisk PBX 1.6.0.6 Allow:...