[asterisk-users] Could Asterisk be rewriting an incoming invite?

[Chris Garrigues]

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"
<sip:+19192282250 at 4.68.250.148<sip%3A%2B19192282250 at 4.68.250.148>
>;tag=VPSF506071629460.
To: <sip:+15129616808 at 4.79.212.229:5060>.
Call-ID: HOUMGC0520090316161653037223 at 209.244.63.35.
CSeq: 1 INVITE.
Contact: <sip:+19192282250 at 4.68.250.148:5060;transport=udp>.
Max-Forwards: 67.
Content-Type: application/sdp.
Content-Length: 177.
Remote-Party-ID: "BANDWIDTH COM"
<sip:+19192282250 at 4.68.250.148<sip%3A%2B19192282250 at 4.68.250.148>>;party=calling
;screen=no;privacy=off.
.
v=0.
o=- 1237220213 1237220214 IN IP4 209.244.187.176.
s=-.
c=IN IP4 209.244.187.176.
t=0 0.
m=audio 60458 RTP/AVP 0 18 101.
a=rtpmap:101 telephone-event/8000.

but asterisk is reporting it like this:

INVITE sip:+15129616808 at 216.82.224.202:5060;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"
<sip:+19192282250 at 4.68.250.148<sip%3A%2B19192282250 at 4.68.250.148>
>;tag=VPSF506071629460
To: <sip:+15129616808 at 4.79.212.229:5060>
Call-ID: HOUMGC0520090316161653037223 at 209.244.63.35
CSeq: 1 INVITE
Contact: <sip:+19192282250 at 4.68.250.148:5060;transport=udp>
Max-Forwards: 67
Content-Type: application/sdp
Content-Length: 175
Remote-Party-ID: "BANDWIDTH COM"
<sip:+19192282250 at 4.68.250.148<sip%3A%2B19192282250 at 4.68.250.148>
>;party=calling;screen=no;privacy=off

v=0
o=- 1237220213 1237220214 IN IP4 216.82.224.202
s=-
c=IN IP4 216.82.224.202
t=0 0
m=audio 60458 RTP/AVP 0 18 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

as a result, I don't get incoming audio for obvious reasons.  Is there any
possibility that it's my asterisk configuration?  I'm having a bear of a
time getting to someone useful at my ISP, so I'm hoping to find that it's my
problem.
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