Dear list,
I have situation where SIP UA is behind NAT, that maps everything to
ports above number 61000. I use nathelper, rtpproxy and SER 0.8.12.
Everything works fine.
Until I have UA that does not send "Contact" field in BYE (or OK for
BYE) during call termination. In this case SER is forwarding responses
not to NAT-ed port, but to port 5060.
QUESTION: What can be done to correct this situation?
Is this improved in new SER releases?
Sequence is as follows:
IP_of_A:61001 -> IP_of_proxy:5060
BYE sip:B@IP_of_proxy
loose_route
...
IP_of_proxy:5060 -> IP_of_B:5060
BYE sip:B@IP_of_B
...
IP_of_B:5060 -> IP_of_proxy:5060
SIP/2.0 200 OK
...
IP_of_proxy:5060 -> IP_of_A:5060 <WRONG! Should be 61001>
SIP/2.0 200 OK
...
It does not happen when terminating UA sends "Contact" field in first BYE.
--
Regards,
Arek Bekiersz
Hi!
Recently on the serdev mailinglists there were considerations about the
stability of the TLS code:
http://mail.iptel.org/pipermail/serdev/2006-January/006608.html
I know there was a fix to disable openssl compression. Was this the only
known problem? Is TLS now running stable?
Has someone ever used the TLS stack successfully under load?
regards
klaus
HI,
set fifo_mode = 666 in your ser.cfg
Karel
Guty napsal(a):
>
>
> Hello!
>
>
>
> I’m having a trouble with my serweb.
>
>
>
> I can access to the user interface login screen but when I log on I
> receive the next error:
>
>
>
> *Warning*: fopen(/tmp/ser_fifo): failed to open stream: Permission
> denied in */home/gustavo/ser/serweb-0.9.4/html/functions.php* on line *340*
>
> *Warning*: Cannot modify header information - headers already sent by
> (output started at
> /home/gustavo/ser/serweb-0.9.4/html/functions.php:340) in
> */home/gustavo/ser/serweb-0.9.4/html/page.php* on line *35*
>
> * *
>
> Thanks!!!!!!
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> Serusers mailing list
> serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
Hello!
I'm having a trouble with my serweb.
I can access to the user interface login screen but when I log on I receive
the next error:
Warning: fopen(/tmp/ser_fifo): failed to open stream: Permission denied in
/home/gustavo/ser/serweb-0.9.4/html/functions.php on line 340
Warning: Cannot modify header information - headers already sent by (output
started at /home/gustavo/ser/serweb-0.9.4/html/functions.php:340) in
/home/gustavo/ser/serweb-0.9.4/html/page.php on line 35
Thanks!!!!!!
In the situation below, a call made from PSTN to openser.
PSTN -----> openser ---------> UA1
How can I remove the caller such that UA1 can't show the caller number
in her display?
Hello everybody,
a new package building of CVS branch rel_1_0_0 (release v1.0.0) will
take place soon. This comes as an update to release 1.0.0 which will
include the fixings occurred since 1.0.0 was released.
We like to collect all issues of v1.0.0 you are aware of, to be fixed
before package generation. Next is a list of the issues that should be
fixed:
- TLS - when TLS connections is refused, immediate reply instead of
waiting for timeout
- TCP - when the call is initiated over TCP but the contact advertised
is UDP, the BYE from callee is not routed properly (it is tried TCP due
to double RR)
- record_route_preset() parameter size
- ruri matching of canceled transactions (see:
http://openser.org/pipermail/users/2006-February/002951.html)
Any feedback is appreciated.
Cheers,
Daniel
Could a solution be using multiple domains per interface of the machina so i
can run diferent instantes of the media proxy ??
_____
De: users-bounces(a)openser.org [mailto:users-bounces@openser.org] En nombre
de Fernando Rodriguez
Enviado el: Lunes, 20 de Febrero de 2006 11:32 a.m.
Para: users(a)openser.org
Asunto: [Users] Multiple Interfaces rtpproxy
Hello,
I have a running openser on a machine with multiple interfaces im using
mediaproxy but I was only able to set it up using 2 interfaces ..
How can I set it up to have more working interfaces
1 lan
1 dmz
1 wan
Right now it only mediaproxies the wan and dmz interfaces
But I want to be able to mediaproxy betrwen the lan and wan and dmz to lan
Does any one has a set up like this and will be able to guide me on seting
this up ..
Thanks a lot for your help
Fernando Rodriguez V.
frod(a)aitelecom.net
Hey guys,
What is the best way to modify the SDP message to always include (if not
replace) the a= tags in a SDP message?
My problem is specifically with g729, according to the RFC, if the
parameter: "a=fmtp:18 annexb=yes" is present or missing then it is assumed
that g729b (g729 with VAD) and if "a=fmtp:18 annexb=no" is present then the
devices will use g729a (g729 with VAD Disabled). My problem is that some
cisco gateways and ATAs behave differently (of course Cisco always doing it
their way), if "a=fmtp:18 annexb=no" is missing then it assumes that g729a
will be used, instead of following the RFC where it should negotiate g729b.
Also, asterisk doesn't support g729b so I want to avoid using g729b
everywhere in my network.
So what is the best way or is there a way, to either:
a) insert "a=fmtp:18 annexb=no" if is not present
b) replace "a=fmtp:18 annexb=yes" for "a=fmtp:18 annexb=no" if it is present
Thanks in advance!!
Lenir
p.s. Please see the below excerpt from RFC 3555 regarding how to specify
G729.
RFC 3555 MIME Type Registration of RTP Payload Formats July 2003
4.1.9. Registration of MIME media type audio/G729
MIME media type name: audio
MIME subtype name: G729
Required parameters: None
Optional parameters:
ptime, maxptime
annexb: indicates that Annex B, voice activity detection, is used or
preferred. Permissible values are "yes" and "no" (without the quotes); "yes"
is implied if this parameter is omitted
I have 2 questions about the port using in openser with NAT enabled.
Q1. Can I using another port number instead of 5060?
I have tried to change the setting in configuration file. Below is a
part of configuration file with NAT enabled.
...
port=6060
...
if (method=="INVITE" && client_nat_test("3")) {
# INSERT YOUR IP ADDRESS HERE
record_route_preset("203.193.26.234:6060;nat=yes");
} else if (method!="REGISTER") {
record_route();
};
Sip phone (UA1) can register in openser successfully. But in the
location table, I have the contact and received field still using port
5060.
sip:871966806561@10.0.0.46:5060 | sip:210.184.23.31:5060
As a result, UA1 failed to make a call as the port is not the one we want.
Q2. How can I set multiple port for an openser if another port can be
used instead of 5060?
Say I want openser listen to 3 ports, ie 5060, 6060, 7060.
