Hello,
I have the following scenario: I receive an INVITE and do parallel forking,
having two branches. What I would like to do then is set a timer for the first
branch, and send a CANCEL on that branch in one of the following cases:
1. no final response is received on the first branch and the timer
expires. In this case it is important that the timer is per branch, so not
both branches expire, and not both are canceled.
2. a 181 Call is being forwarded reply is received on the second
branch.
Any suggestions are welcomed, because right now I have absolutely no idea
how I could solve this. I read the documentation for the TM module, all
kinds of timers, but they are all per transaction, not per branch, or at
least so I understood. (I am using OpenSER 1.2)
In conclusion, my questions are:
1. Can I set a per branch timer? (either from code or configuration script)
2. How can I send a CANCEL only on that branch when the timer expires?
3. How can I send a CANCEL on one branch when the other has received a non
final reply, 181 in this case. (the problem is the actual generation and
sending of the CANCEL message, not intercepting the 181 reply or finding out
on which branch it was received).
Thank you in advance,
Madalina
Hello,
I have few questions before i proceed to get the integration of openser with the b2bua;
I am confused that i need to use the mediaproxy module of the openser if i am taking up the media from the b2bua as i need the b2bua as openser is not performing topology hiding.
Cheers,
KChris
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Hi, I've started reading CDRTool documentation (very complete and long). Let
me two basic questions about it:
1) Does CDRTool allow limiting simultaneous calls from/to a user/domain?
I mean something as "fixed number of lines" per client.
2) I plan to use this scenario:
sip-users <-> OpenSer <-> Asterisk B2BUA <-> CDRTool <-> SIP_to_PSTN provider
So the sip-user will just authenticate against OpenSer, which will route call
to Asterisk, and Asterisk as B2BUA will authenticate against CDRTool, so
the "from_user" will be set by Asterisk but the "caller ID" will be the
sip-user's original caller ID.
Does CDRTool log and show the caller ID (not just the fromuser) in the web
report?
Or maybe it's not necessary to set the "from_user" in Asterisk? does CDRTool
allow digest auth different of the "from_user"? does CDRTool
require "check_from()"?
Thanks a lot for any help. Best regards.
--
Iñaki Baz Castillo
Everyone:
I've set up a mirror of the OpenSER mailing lists on Nabble. One of the
main reasons for this was to obtain search capabilities.
If you're interested in this, please take a look at:
http://www.nabble.com/OpenSER---the-Open-Source-SIP-Server-f26014.html
Their home page is http://www.nabble.com
Regards,
Norm
p.s I would like to thank the folks at Nabble for importing the
historical archives into their systems.
Hi Ram,
I had tried with many providers having Voice Master but have not been able
to work with any of them. People have tried opening username and password
account, only prefix based and only IP based authentication. But none have
worked, VM keep rejecting calls.
As far as I understand VM checks IP Address/username in contact field as
well, which I am unable to change. I can change username and or IP Address
in from field using uac_replace_from but could not find anything to change
the same in contact field of invite message.
Only Scenarios where I am able to send calls via VM are:
1.) I have both the endpoint and openser's IP Address to be opened at
VM. ( In case I have IP only authentication account at VM end)
2.) I send prefix from the endpoint end itself and not added at
openser's level. (In case I have prefix only authentication account at VM
end).
But in both scenario's openser becomes useless and can easily be bypassed by
client as VM will allow calls directly from ENDPOINT as well.
Regards,
Dan
_____
From: ram [mailto:talk2ram@gmail.com]
Sent: Monday, September 17, 2007 9:46 PM
To: Dan
Subject: Re: [OpenSER-Users] Openser and Voice master
how are u trying
should not be problem as for i know
ram
On 9/11/07, Dan <fiedler.dan(a)gmail.com> wrote:
Hi,
Has anybody able to successfully able to route calls to voicemaster from
openser.
I tried all method but none worked. From username/password to IP based
authentication to prefix based authentication.
Regards
Dan
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http://openser.org/cgi-bin/mailman/listinfo/users
Hi to all, I'm newbie using SER and need your help.
This is my
testbed:
SER running on 192.168.123.186
MediaProxy running on
192.168.123.186
Client_A X-lite running under WinowsXP on
192.168.123.168
Client_B X-lite running under WinowsXP on
192.168.123.10
I want to route RTP packets between 2 computers in the
same network through Mediaproxy. I know this may have
no sense, but
it's just a test configuration.
The call use_media_proxy() seems to
have no effect,
the RTP packets go directly from one computer to
another.
My guess is mediaproxy makes a test on the IPs (they
are
192.168.147.2 and 192.168.147.3) and refuses to
proxy the call.
Is
mediaproxy working just between different networks
or it's an error in
my configuration file?
But if i use this Enviroment:
SER running on
192.168.123.186
MediaProxy running on 192.168.123.186
Client_A X-lite
running under Linux Fedora on 192.168.123.168
Client_B X-lite running
under WinowsXP on 192.168.123.10
The call use_media_proxy() correctly
and the RTP packets
pass through the mediaproxy, infacty using tools
session.py I see the session active and traffic packet amount.
Why,
with client under Windows, Media proxy doesn't work correctly?
Any
suggestions, please?
Thanks in advance.
Orazio
Hi,
I have build my SER 2.0.0 with radius client to leverage existing
authentication application.
Can someone point me to any HOWTO use radius clent call authentication?
Or, can some thoughtful person share their own ser.cfg example?
Thanks ..mike..
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Hi,
I want to try SER 2.0.0 RC 1 with presence capabilities. Can any one tell me some clients with presence support which I can configure and test with SER presence module.
Thanking you in advance.
Best Regards,
Abdul Qadir
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Hi,
Here is an indication that my SER is running:
micadeyeye@asmicom:/usr/src/Research-Progress$ ser
Listening on
udp: 127.0.0.1 [127.0.0.1]:5060
udp: 196.24.224.76 [196.24.224.76]:5060
tcp: 127.0.0.1 [127.0.0.1]:5060
tcp: 196.24.224.76 [196.24.224.76]:5060
Aliases:
tcp: asmicom.local:5060
tcp: localhost:5060
udp: asmicom.local:5060
udp: localhost:5060
---------------------------------------------------------------------------------------------------
But each time I tried connecting to it from a USER AGENT (ZAP,
UCTIMSCLIENT), it doesn't interact with the client. I checked my
dump(via wireshack) but only REGISTER messages are sent with no reply
from SER.
Can anyone help me? I have both SER and client sitting on same PC (using
same port-5060).
Hi,
I did everything according to the tls module description
when I start SER I get the following error:
What do I have to do to make things work?
Thanks in advance.
Tomasz
Aug 24 18:42:39 sen ser[26840]: tls: _init_tls_h: compiled with
openssl version "OpenSSL 0.9.8c 05 Sep 2006" (0x0090803f), kerberos
support: off, compression: on
Aug 24 18:42:39 sen ser[26840]: tls: init_tls_h: installed openssl
library version "OpenSSL 0.9.8c 05 Sep 2006" (0x0090803f), kerberos
support: off, zlib compression: on compiler: gcc -fPIC -DOPENSSL_PIC
-DZLIB -DOPENSSL_THREADS -D_REENTRANT -DDSO_DLFCN -DHAVE_DLFCN_H -m64
-DL_ENDIAN -DTERMIO -O3 -Wa,--noexecstack -g -Wall -DMD32_REG_T=int
-DMD5_ASM
Aug 24 18:42:39 sen ser[26840]: ERROR: tls_init.c:366: Unable to set the
memory allocation functions
Aug 24 18:42:39 sen ser[26840]: could not initialize tls, exiting...