No subject
Mon Nov 25 18:50:14 CET 2013
Session Initiation Protocol (200)
Status-Line: SIP/2.0 200 OK
Status-Code: 200
[Resent Packet: False]
[Request Frame: 9]
[Response Time (ms): 4049]
Message Header
Via: SIP/2.0/UDP *.*.*.110:5060;received=3D*.*.*.110;branch=3Dz9hG4=
bK-d8754z-HSTATXOSEB0050004f58cb4f4b0f5-1---d8754z-;rport=3D5060
Transport: UDP
Sent-by Address: *.*.*.110
Sent-by port: 5060
Received: *.*.*.110
Branch: z9hG4bK-d8754z-HSTATXOSEB0050004f58cb4f4b0f5-1---d8754z=
-
RPort: 5060
Record-Route: <sip:*.*.*.200;lr>
Record-Route URI: sip:*.*.*.200;lr
Record-Route Host Part: *.*.*.200
Record-Route URI parameter: lr
From: "**** ****"<sip:********79@*.*.*.110:5060>;tag=3DHSTATXOSEB00=
50004f58cb4f4b0f6
SIP Display info: "**** ****"
SIP from address: sip:********79@*.*.*.110:5060
SIP from address User Part: ********79
SIP from address Host Part: *.*.*.110
SIP from address Host Port: 5060
SIP from tag: HSTATXOSEB0050004f58cb4f4b0f6
To: <sip:********33@*.*.*.200:5060>;tag=3Das4f58e1e1
SIP to address: sip:********33@*.*.*.200:5060
SIP to address User Part: ********33
SIP to address Host Part: *.*.*.200
SIP to address Host Port: 5060
SIP to tag: as4f58e1e1
Call-ID: 6654c342.c8fafa0a.5333822b.bf5
CSeq: 1 INVITE
Sequence Number: 1
Method: INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,=
INFO
Supported: replaces, timer
Session-Expires: 1800;refresher=3Duac
Contact: <sip:********33@*.*.*.102>
Contact URI: sip:********33@*.*.*.102
Contact URI User Part: ********33
Contact URI Host Part: *.*.*.102
Content-Type: application/sdp
Content-Length: 240
Message Body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): root 1240385050 1240385050 IN IP=
4 *.*.*.102
Owner Username: root
Session ID: 1240385050
Session Version: 1240385050
Owner Network Type: IN
Owner Address Type: IP4
Owner Address: *.*.*.102
Session Name (s): Asterisk PBX
Connection Information (c): IN IP4 *.*.*.102
Connection Network Type: IN
Connection Address Type: IP4
Connection Address: *.*.*.102
Time Description, active time (t): 0 0
Session Start Time: 0
Session Stop Time: 0
Media Description, name and address (m): audio 7610 RTP/AVP 0 1=
01
Media Type: audio
Media Port: 7610
Media Protocol: RTP/AVP
Media Format: ITU-T G.711 PCMU
Media Format: DynamicRTP-Type-101
Media Attribute (a): rtpmap:0 PCMU/8000
Media Attribute Fieldname: rtpmap
Media Format: 0
MIME Type: PCMU
Sample Rate: 8000
Media Attribute (a): rtpmap:101 telephone-event/8000
Media Attribute Fieldname: rtpmap
Media Format: 101
MIME Type: telephone-event
Sample Rate: 8000
Media Attribute (a): fmtp:101 0-16
Media Attribute Fieldname: fmtp
Media Format: 101 [telephone-event]
Media format specific parameters: 0-16
Media Attribute (a): ptime:20
Media Attribute Fieldname: ptime
Media Attribute Value: 20
Media Attribute (a): sendrecv
And the ACK message that goes back to the Asterisk server and not the proxy=
looks like this:
Session Initiation Protocol (ACK)
Request-Line: ACK sip:********33@*.*.*102 SIP/2.0
Method: ACK
Request-URI: sip:********33@*.*.*102
Request-URI User Part: ********33
Request-URI Host Part: *.*.*102
[Resent Packet: False]
Message Header
Via: SIP/2.0/UDP *.*.*110:5060;branch=3Dz9hG4bK-d8754z-HSTATXOSEB00=
50004f58cb533a2c8-1---d8754z-;rport
Transport: UDP
Sent-by Address: *.*.*110
Sent-by port: 5060
Branch: z9hG4bK-d8754z-HSTATXOSEB0050004f58cb533a2c8-1---d8754z=
-
RPort: rport
Max-Forwards: 70
To: <sip:********33@*.*.*200:5060>;tag=3Das4f58e1e1
SIP to address: sip:********33@*.*.*200:5060
SIP to address User Part: ********33
SIP to address Host Part: *.*.*200
SIP to address Host Port: 5060
SIP to tag: as4f58e1e1
From: "**** ****"<sip:********79@*.*.*110:5060>;tag=3DHSTATXOSEB005=
0004f58cb4f4b0f6
SIP Display info: "**** ****"
SIP from address: sip:********79@*.*.*110:5060
SIP from address User Part: ********79
SIP from address Host Part: *.*.*110
SIP from address Host Port: 5060
SIP from tag: HSTATXOSEB0050004f58cb4f4b0f6
Call-ID: 6654c342.c8fafa0a.5333822b.bf5
CSeq: 1 ACK
Sequence Number: 1
Method: ACK
Content-Length: 0
I am being told that the Contact header in the OK message should have the I=
P address of the proxy and not the Asterisk server. I'm looking at the RFC=
document, RFC3261, attempting to understand the "rules of the road" here, =
but am getting confused on the requirements of the Contact Header.
Is what I am being told correct? And, if so, what would be the cleanest wa=
y to go about correcting that particular header?
Cordially,
Peter Nayland Kust
Director of Technologies
BusinesSuites
24624 Interstate 45 North, Suite 200
Houston, TX 77386
peter.kust at businessuites.com
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<div class=3D"WordSection1">
<p class=3D"MsoNormal">I am currently testing an OpenSIPS/Asterisk combinat=
ion with a GenBand eSBC (Quantix QFlex).<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p> </o:p></p>
<p class=3D"MsoNormal">My basic architecture looks like this<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p> </o:p></p>
<p class=3D"MsoNormal">Phone (Cisco SPA525G2) <span style=3D"font-family:Wi=
ngdings">=E0</span> OpenSIPS proxy
<span style=3D"font-family:Wingdings">=E0</span> Asterisk Media Server<o:p>=
</o:p></p>
<p class=3D"MsoNormal">Asterisk Media Server <span style=3D"font-family:Win=
gdings">=E0</span> OpenSIPS proxy
<span style=3D"font-family:Wingdings">=E0</span> GenBand QFlex eSBC (<span =
style=3D"font-family:Wingdings">=E0</span>PSTN)<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p> </o:p></p>
<p class=3D"MsoNormal">The GenBand is handling both the SIP and RTP protoco=
ls, which means the Asterisk Media Server is sending the RTP stream direct =
to the GenBand.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p> </o:p></p>
<p class=3D"MsoNormal">A problem arises on inbound calls (from PSTN through=
GenBand to OpenSIPS/Asterisk). During the call setup the GenBand sen=
ds a SIP ACK message directly to my Asterisk server, which seems to be caus=
ing the Asterisk server to send the BYE
message at the end of the call directly to the GenBand instead of via the =
OpenSIPS proxy. The result is that the external call end point (i.e.,=
my cell phone), never gets a BYE message and that call leg stays open.<o:p=
></o:p></p>
<p class=3D"MsoNormal"><o:p> </o:p></p>
<p class=3D"MsoNormal">In the OK message from the proxy to the GenBand, the=
Contact header contains the IP address of my Asterisk server, and not the =
proxy. I am being told this is what prompts the GenBand to send to th=
e Asterisk server and not the proxy.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p> </o:p></p>
<p class=3D"MsoNormal">From a packet capture I have run on the offending ca=
ll scenario, the OK message in question looks like this:<o:p></o:p></p>
<p class=3D"MsoNormal">Session Initiation Protocol (200)<o:p></o:p></p>
<p class=3D"MsoNormal"> Status-Line: SIP/2.0 200 OK<o:p><=
/o:p></p>
<p class=3D"MsoNormal"> Status-Co=
de: 200<o:p></o:p></p>
<p class=3D"MsoNormal"> [Resent P=
acket: False]<o:p></o:p></p>
<p class=3D"MsoNormal"> [Request =
Frame: 9]<o:p></o:p></p>
<p class=3D"MsoNormal"> [Response=
Time (ms): 4049]<o:p></o:p></p>
<p class=3D"MsoNormal"> Message Header<o:p></o:p></p>
<p class=3D"MsoNormal"> Via: SIP/=
2.0/UDP *.*.*.110:5060;received=3D*.*.*.110;branch=3Dz9hG4bK-d8754z-HSTATXO=
SEB0050004f58cb4f4b0f5-1---d8754z-;rport=3D5060<o:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; Transport: UDP<o:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; Sent-by Address: *.*.*.110<o:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; Sent-by port: 5060<o:p></o:p></p>
<p class=3D"MsoNormal"> &nb=
sp; Received: *.*.*.110<o:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; Branch: z9hG4bK-d8754z-HSTATXOSEB0050004f58cb4f4b0f5-1---d87=
54z-<o:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; RPort: 5060<o:p></o:p></p>
<p class=3D"MsoNormal"> Record-Ro=
ute: <sip:*.*.*.200;lr><o:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; Record-Route URI: sip:*.*.*.200;lr<o:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; Record-Route Host Part: *.*.*.200<o:=
p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; Record-Route URI parameter: lr<o:p><=
/o:p></p>
<p class=3D"MsoNormal"> From: &qu=
ot;**** ****"<sip:********79@*.*.*.110:5060>;tag=3DHSTATXOSEB005=
0004f58cb4f4b0f6<o:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; SIP Display info: "**** ****"<o:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; SIP from address: sip:********79@*.*.*.110:5060<o:p></o:p></=
p>
<p class=3D"MsoNormal"> &nb=
sp; SIP from address User Part: ********=
79<o:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; SIP from address Host Part: *.*.*.11=
0<o:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; SIP from address Host Port: 5060<o:p=
></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; SIP from tag: HSTATXOSEB0050004f58cb4f4b0f6<o:p></o:p></p>
<p class=3D"MsoNormal"> To: <s=
ip:********33@*.*.*.200:5060>;tag=3Das4f58e1e1<o:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; SIP to address: sip:********33@*.*.*.200:5060<o:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; SIP to address User Part: ********33=
<o:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; SIP to address Host Part: *.*.*.200<=
o:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; SIP to address Host Port: 5060<o:p><=
/o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; SIP to tag: as4f58e1e1<o:p></o:p></p>
<p class=3D"MsoNormal"> Call-ID: =
6654c342.c8fafa0a.5333822b.bf5<o:p></o:p></p>
<p class=3D"MsoNormal"> CSeq: 1 I=
NVITE<o:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; Sequence Number: 1<o:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; Method: INVITE<o:p></o:p></p>
<p class=3D"MsoNormal"> Server: A=
sterisk PBX <o:p></o:p></p>
<p class=3D"MsoNormal"> Allo=
w: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<o:p></=
o:p></p>
<p class=3D"MsoNormal"> Supported=
: replaces, timer<o:p></o:p></p>
<p class=3D"MsoNormal"> Session-E=
xpires: 1800;refresher=3Duac<o:p></o:p></p>
<p class=3D"MsoNormal"> Contact: =
<sip:********33@*.*.*.102><o:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; Contact URI: sip:********33@*.*.*.102<o:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; Contact URI User Part: ********33<o:=
p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; Contact URI Host Part: *.*.*.102<o:p=
></o:p></p>
<p class=3D"MsoNormal"> Content-T=
ype: application/sdp<o:p></o:p></p>
<p class=3D"MsoNormal"> Content-L=
ength: 240<o:p></o:p></p>
<p class=3D"MsoNormal"> Message Body<o:p></o:p></p>
<p class=3D"MsoNormal"> Session D=
escription Protocol<o:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; Session Description Protocol Version (v): 0<o:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; Owner/Creator, Session Id (o): root 1240385050 1240385050 IN=
IP4 *.*.*.102<o:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; Owner Username: root<o:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; Session ID: 1240385050<o:p></o:p></p=
>
<p class=3D"MsoNormal"> &nbs=
p; Session Version: 1240385050<o:p></o:=
p></p>
<p class=3D"MsoNormal"> &nbs=
p; Owner Network Type: IN<o:p></o:p></p=
>
<p class=3D"MsoNormal"> &nbs=
p; Owner Address Type: IP4<o:p></o:p></=
p>
<p class=3D"MsoNormal"> &nbs=
p; Owner Address: *.*.*.102<o:p></o:p><=
/p>
<p class=3D"MsoNormal"> &nb=
sp; Session Name (s): Asterisk PBX <o:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; Connection Information (c): IN IP4 *.*.*.102<o:p></o:p>=
</p>
<p class=3D"MsoNormal"> &nbs=
p; Connection Network Type: IN<o:p></o:=
p></p>
<p class=3D"MsoNormal"> &nbs=
p; Connection Address Type: IP4<o:p></o=
:p></p>
<p class=3D"MsoNormal"> &nbs=
p; Connection Address: *.*.*.102<o:p></=
o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; Time Description, active time (t): 0 0<o:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; Session Start Time: 0<o:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; Session Stop Time: 0<o:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; Media Description, name and address (m): audio 7610 RTP/AVP =
0 101<o:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; Media Type: audio<o:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; Media Port: 7610<o:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; Media Protocol: RTP/AVP<o:p></o:p></=
p>
<p class=3D"MsoNormal"> &nbs=
p; Media Format: ITU-T G.711 PCMU<o:p><=
/o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; Media Format: DynamicRTP-Type-101<o:=
p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; Media Attribute (a): rtpmap:0 PCMU/8000<o:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; Media Attribute Fieldname: rtpmap<o:=
p></o:p></p>
<p class=3D"MsoNormal"> &nb=
sp; Media Format: 0<o:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; MIME Type: PCMU<o:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; Sample Rate: 8000<o:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; Media Attribute (a): rtpmap:101 telephone-event/8000<o:p></o=
:p></p>
<p class=3D"MsoNormal"> &nbs=
p; Media Attribute Fieldname: rtpmap<o:=
p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; Media Format: 101<o:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; MIME Type: telephone-event<o:p></o:p=
></p>
<p class=3D"MsoNormal"> &nbs=
p; Sample Rate: 8000<o:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; Media Attribute (a): fmtp:101 0-16<o:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; Media Attribute Fieldname: fmtp<o:p>=
</o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; Media Format: 101 [telephone-event]<=
o:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; Media format specific parameters: 0-=
16<o:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; Media Attribute (a): ptime:20<o:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; Media Attribute Fieldname: ptime<o:p=
></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; Media Attribute Value: 20<o:p></o:p>=
</p>
<p class=3D"MsoNormal"> &nbs=
p; Media Attribute (a): sendrecv<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p> </o:p></p>
<p class=3D"MsoNormal">And the ACK message that goes back to the Asterisk s=
erver and not the proxy looks like this:<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p> </o:p></p>
<p class=3D"MsoNormal">Session Initiation Protocol (ACK)<o:p></o:p></p>
<p class=3D"MsoNormal"> Request-Line: ACK sip:********33@=
*.*.*102 SIP/2.0<o:p></o:p></p>
<p class=3D"MsoNormal"> Method: A=
CK<o:p></o:p></p>
<p class=3D"MsoNormal"> Request-U=
RI: sip:********33@*.*.*102<o:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; Request-URI User Part: ********33<o:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; Request-URI Host Part: *.*.*102<o:p></o:p></p>
<p class=3D"MsoNormal"> [Resent P=
acket: False]<o:p></o:p></p>
<p class=3D"MsoNormal"> Message Header<o:p></o:p></p>
<p class=3D"MsoNormal"> Via: SIP/=
2.0/UDP *.*.*110:5060;branch=3Dz9hG4bK-d8754z-HSTATXOSEB0050004f58cb533a2c8=
-1---d8754z-;rport<o:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; Transport: UDP<o:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; Sent-by Address: *.*.*110<o:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; Sent-by port: 5060<o:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; Branch: z9hG4bK-d8754z-HSTATXOSEB0050004f58cb533a2c8-1---d87=
54z-<o:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; RPort: rport<o:p></o:p></p>
<p class=3D"MsoNormal"> Max-Forwa=
rds: 70<o:p></o:p></p>
<p class=3D"MsoNormal"> To: <s=
ip:********33@*.*.*200:5060>;tag=3Das4f58e1e1<o:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; SIP to address: sip:********33@*.*.*200:5060<o:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; SIP to address User Part: ********33=
<o:p></o:p></p>
<p class=3D"MsoNormal"> &nb=
sp; SIP to address Host Part: *.*.*200<o=
:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; SIP to address Host Port: 5060<o:p><=
/o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; SIP to tag: as4f58e1e1<o:p></o:p></p>
<p class=3D"MsoNormal"> From: &qu=
ot;**** ****"<sip:********79@*.*.*110:5060>;tag=3DHSTATXOSEB0050=
004f58cb4f4b0f6<o:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; SIP Display info: "**** ****"<o:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; SIP from address: sip:********79@*.*.*110:5060<o:p></o:p></p=
>
<p class=3D"MsoNormal"> &nbs=
p; SIP from address User Part: ********=
79<o:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; SIP from address Host Part: *.*.*110=
<o:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; SIP from address Host Port: 5060<o:p=
></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; SIP from tag: HSTATXOSEB0050004f58cb4f4b0f6<o:p></o:p></p>
<p class=3D"MsoNormal"> Call-ID: =
6654c342.c8fafa0a.5333822b.bf5<o:p></o:p></p>
<p class=3D"MsoNormal"> CSeq: 1 A=
CK<o:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; Sequence Number: 1<o:p></o:p></p>
<p class=3D"MsoNormal"> &nbs=
p; Method: ACK<o:p></o:p></p>
<p class=3D"MsoNormal"> Content-L=
ength: 0<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p> </o:p></p>
<p class=3D"MsoNormal">I am being told that the Contact header in the OK me=
ssage should have the IP address of the proxy and not the Asterisk server.&=
nbsp; I’m looking at the RFC document, RFC3261, attempting to underst=
and the “rules of the road” here, but am getting
confused on the requirements of the Contact Header.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p> </o:p></p>
<p class=3D"MsoNormal">Is what I am being told correct? And, if so, w=
hat would be the cleanest way to go about correcting that particular header=
?<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p> </o:p></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:"Ar=
ial","sans-serif"">Cordially,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:"Ar=
ial","sans-serif""><o:p> </o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:"Ar=
ial","sans-serif"">Peter Nayland Kust<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:"Ar=
ial","sans-serif";color:gray">Director of Technologies</span=
><span style=3D"font-size:12.0pt;font-family:"Times New Roman",&q=
uot;serif";color:gray"><o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:"Ar=
ial","sans-serif";color:gray">BusinesSuites<o:p></o:p></span=
></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:"Ar=
ial","sans-serif";color:gray">24624 Interstate 45 North, Sui=
te 200<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:"Ar=
ial","sans-serif";color:gray">Houston, TX 77386<o:p></o:p></=
span></p>
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:"=
;Arial","sans-serif"">peter.kust at businessuites.com </sp=
an></b><span style=3D"font-size:10.0pt;font-family:"Arial","=
sans-serif";color:gray"><o:p></o:p></span></p>
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