Chris Garrigues
2009-Mar-16 17:08 UTC
[asterisk-users] 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" <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=VPSF506071629460To: <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=offv=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. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090316/5a11e4c5/attachment.htm
Chris Garrigues
2009-Mar-16 17:49 UTC
[asterisk-users] Could Asterisk be rewriting an incoming invite?
I've just determined that it IS happening on my box, but why?
I did a packet capture using tcpdump on this very same box and it shows the
correct invite while sip debug shows the wrong values. here's what I see in
wireshark:
No. Time Source Destination Protocol
Info
1 0.000000 216.82.224.202 67.198.16.18 SIP/SDP
Request: INVITE sip:+15129616808 at 67.198.16.18:5060;transport=udp, with
session description
Frame 1 (1043 bytes on wire, 1043 bytes captured)
Ethernet II, Src: EciTelec_00:a0:41 (00:02:0e:00:a0:41), Dst: Intel_92:3b:be
(00:0c:f1:92:3b:be)
Internet Protocol, Src: 216.82.224.202 (216.82.224.202), Dst: 67.198.16.18
(67.198.16.18)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
Request-Line: INVITE sip:+15129616808 at 67.198.16.18:5060;transport=udp
SIP/2.0
Message Header
Message Body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): - 1237225281 1237225282 IN IP4
209.244.187.171
Session Name (s): -
Connection Information (c): IN IP4 209.244.187.171
Time Description, active time (t): 0 0
Media Description, name and address (m): audio 60570 RTP/AVP 0
18 101
Media Attribute (a): rtpmap:101 telephone-event/8000
Media Attribute (a): fmtp:101 0-15
and here's what I see in sip debug:
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 216.82.224.202:5060;branch=z9hG4bK525.4ab0348.0
Via: SIP/2.0/UDP 4.79.212.229;branch=z9hG4bK525.3b6e7ab3.0
Via: SIP/2.0/UDP 4.68.250.148:5060;branch=z9hG4bK506071629460-1207517079314
From: "GARRIGUES,CHRIS"
<sip:+15124990483 at 4.68.250.148<sip%3A%2B15124990483 at 4.68.250.148>
;isup-oli=0>;tag=VPSF506071629460
To: <sip:+15129616808 at 4.79.212.229:5060>
Call-ID: HOUMGC0520090316174121064302 at 209.244.63.35
CSeq: 1 INVITE
Contact: <sip:+15124990483 at 216.82.224.202:5060;transport=udp>
Max-Forwards: 67
Content-Type: application/sdp
Content-Length: 175
Remote-Party-ID: "GARRIGUES,CHRIS"
<sip:+15124990483 at 4.68.250.148<sip%3A%2B15124990483 at
4.68.250.148>>;party=calling;screen=yes;privacy=off
v=0
o=- 1237225281 1237225282 IN IP4 216.82.224.202
s=-
c=IN IP4 216.82.224.202
t=0 0
m=audio 60570 RTP/AVP 0 18 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
What is rewriting my o= and c
??????
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20090316/ae9e9d2d/attachment.htm