16 Mar
2005
16 Mar
'05
3:19 p.m.
--- serusers-request(a)lists.iptel.org wrote:
http://lists.iptel.org/pipermail/serusers/2004-March/006806.html
>
> But nothing has led me to a solution. I cannot just
> use
> lookup("location") and test the nat_flag since that
> won't always work
> on re-INVITEs. A mediaproxy function for something
> like
> is_existing_session() would be awesome to lookup the
> Call-ID in the
> existing mediaproxy sessions.
>
> What am I missing?
>
> Regards,
> Paul
>
>
>
> ------------------------------
>
> Message: 2
> Date: Tue, 15 Mar 2005 22:12:15 -0500
> From: Java Rockx <javarockx(a)gmail.com
> Subject: Re: [Serusers] using sipsak to send a BYE
> To: Iqbal <iqbal(a)gigo.co.uk
> Cc: serusers(a)lists.iptel.org
> Message-ID:
> <359a6582050315191229df7a1e(a)mail.gmail.com
> Content-Type: text/plain; charset=ISO-8859-1
>
> I have. During my debugging with a 3rd party PSTN GW
> provider I
> sometimes didn't send them a BYE message and they
> couldn't close their
> billing record.
>
> So I stored this in a file and used sipsak to close
> their billing record.
>
> BYE sip:5135551234@216.229.118.76:4060 SIP/2.0
> Route: <sip:216.229.127.60:5060;lr
> From: sip:5135551234@64.152.60.6;user=phone
> To: sip:9145550000@66.243.109.99:5060;user=phone
> Contact:
> sip:9145550000@66.243.109.99:5060;user=phone
> CSeq: 42759 BYE
> Call-ID: 9787963-3319555348-76751(a)66.243.109.99
> User-Agent: Grandstream BT100 1.0.5.11
> Max-Forwards: 16
> Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
> Content-Length: 0
>
> sipsak -f <filename> -s sip:<some IP address
>
> Regards,
> Paul
>
> On Wed, 16 Mar 2005 03:07:05 GMT, Iqbal
> <iqbal(a)gigo.co.uk> wrote:
> >
> > Hi
> >
> > Has anyone done this successfully to hangup a
> call.
> >
> > Iqbal
> >
> > _______________________________________________
> > Serusers mailing list
> > serusers(a)lists.iptel.org
> > http://lists.iptel.org/mailman/listinfo/serusers
> >
>
>
> ------------------------------
>
> Message: 3
> Date: Wed, 16 Mar 2005 14:18:46 +1100
> From: "Zeus Ng" <zeus.ng(a)isquare.com.au
> Subject: RE: [Serusers] LCR Backport Example For
> ser-0.9
> To: "'Java Rockx'" <javarockx(a)gmail.com > Cc: serusers(a)lists.iptel.org
> Message-ID:
> <013701c529d6$de175b60$6515a8c0@isdws101
> Content-Type: text/plain; charset="iso-8859-1"
>
> How about something like this.
>
> With the initial INVITE,
>
> ... Route processing ...
> if (caller is natted) {
>
subst('/From:(.*)<(.*)>(.*)/From:\1<\2>;nat=yes\3/');
> }
>
> call nathelper or mediaproxy
>
> t_on_reply(nat reply route);
> t_relay();
>
>
> In your reply route,
>
> if (callee is natted) {
>
subst('/From:(.*)<(.*)>(.*)/From:\1<\2>;nat=yes\3/');
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Today's Topics:
1. HELP: mediaproxy re-INVITE NAT detection (Java
Rockx)
2. Re: using sipsak to send a BYE (Java Rockx)
3. RE: LCR Backport Example For ser-0.9 (Zeus Ng)
4. Re: LCR Backport Example For ser-0.9 (Java
Rockx)
5. Re: LCR Backport Example For ser-0.9 (Java
Rockx)
6. SER consultant required (Sun Shung)
7. RE: LCR Backport Example For ser-0.9 (Zeus Ng)
8. RE: LCR Backport Example For ser-0.9 (Zeus Ng)
9. Re: LCR Backport Example For ser-0.9 (Java
Rockx)
10. Re: LCR Backport Example For ser-0.9 (Juha
Heinanen)
11. Users cannot connect (Barry Murphy)
12. All working cept budgetone (Barry Murphy)
13. get 302 contact in failure route (Zappasodi
Daniele)
14. what's the difference between the shm_malloc
and pkg_malloc (szj)
15. RE: All working cept budgetone (Zeus Ng)
16. Comments to the onsip's SIP getting Started
document
(Samuel Osorio Calvo)
17. What does that mean? (Olivier Taylor)
18. SER configuracion file (Llanos Serna
Garc?a-Conde)
19. Re: SER configuracion file (Joao Pereira)
----------------------------------------------------------------------
Message: 1
Date: Tue, 15 Mar 2005 22:08:34 -0500
From: Java Rockx <javarockx(a)gmail.com
Subject: [Serusers] HELP: mediaproxy re-INVITE NAT
detection
To: "serusers(a)lists.iptel.org" <serusers(a)lists.iptel.org Message-ID:
<359a658205031519087d640838(a)mail.gmail.com
Content-Type: text/plain; charset=ISO-8859-1
Hi All.
I'm using ser-0.9.1.
Is there a way to determine if mediaproxy is in use
for an existing
SIP call so that re-INVITE messages can avoid losing
audio when one or
the other SIP UAs are NATed?
We do not proxy RTP streams unless one or more SIP
UAs are NATed. But
on re-INVITE messages I cannot figure out how to
test the destination
of the re-INVITE for the UAs NAT status.
I've scoured the archives and found a few related
articles
http://lists.iptel.org/pipermail/serdev/2004-March/001515.html
call nathelper or mediaproxy
... Other processing ...
In your loose route,
if (loose_route()) {
if (method=="INVITE") {
=== message truncated ===
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