Hi all,
Can't receive a 500 (Server Internal Error) response from openser register server when sending a REGISTER with the equal CSeq header field value to the last value of CSeq or lower than the number and “*” in a Contact header field. The details can be as follows:
1 Test topology:
NUT(REG && PX) UA11 UA12 DNS
| | | |
IP : 3ffe:501:ffff:50::50 3ffe:501:ffff:1::1 3ffe:501:ffff:2::2 3ffe:501:ffff:4::1
aor-uri : ss.under.test.com UA11(a)under.test.com UA12(a)under.test.com
contact-uri : UA11(a)node.under.test.com UA12(a)node11.under.test.com
2 tcpdump file can be seen in "12 SIP Registrar- Record-Route header field in REGISTER request" attachment
3 openser configuration file can be seen in "openser.cfg" attachment
How can I modify the configuration file if I want to receive the 500?
--------------------------------------
Regards!
Hu xiaoqiang
I'm trying to install Siremis 1.0.0 to work with Kamailio 3.0 and I'm
having some problems which hopefully somebody can help with.
Kamailio, the MySQL server and Siremis are running on the same server.
The server is running Debian 5.0.4
I have followed the Siremis installation instructions at the following
link: http://siremis.asipto.com/install/
I reach a point were I can get to the Siremis web interface login page
but when I enter the user name and password nothing happens.
/var/log/apache2/access.log :
=============================
192.168.1.101 - - [19/Jun/2010:15:53:22 +0200] "GET /siremis/ HTTP/1.1"
302 271 "-" "Mozilla/5.0 (X11; U; Linux i686; en-US; rv:1.9.2.3)
Gecko/20100423 Ubuntu/10.04 (lucid) Firefox/3.6.3"
192.168.1.101 - - [19/Jun/2010:15:53:22 +0200] "GET /siremis/bin/
HTTP/1.1" 200 1083 "-" "Mozilla/5.0 (X11; U; Linux i686; en-US;
rv:1.9.2.3) Gecko/20100423 Ubuntu/10.04 (lucid) Firefox/3.6.3"
192.168.1.101 - - [19/Jun/2010:15:53:22 +0200] "GET /siremis/js/sound.js
HTTP/1.1" 404 270 "http://192.168.1.14/siremis/bin/" "Mozilla/5.0 (X11;
U; Linux i686; en-US; rv:1.9.2.3) Gecko/20100423 Ubuntu/10.04 (lucid)
Firefox/3.6.3"
192.168.1.101 - - [19/Jun/2010:15:53:23 +0200] "GET /siremis/js/sound.js
HTTP/1.1" 404 270 "http://192.168.1.14/siremis/bin/" "Mozilla/5.0 (X11;
U; Linux i686; en-US; rv:1.9.2.3) Gecko/20100423 Ubuntu/10.04 (lucid)
Firefox/3.6.3"
192.168.1.101 - - [19/Jun/2010:15:53:38 +0200]
"POST /siremis/bin/controller.php HTTP/1.1" 200 20
"http://192.168.1.14/siremis/bin/" "Mozilla/5.0 (X11; U; Linux i686;
en-US; rv:1.9.2.3) Gecko/20100423 Ubuntu/10.04 (lucid) Firefox/3.6.3"
/var/log/apache2/error.log :
============================
[Sat Jun 19 15:53:22 2010] [error] [client 192.168.1.101] File does not
exist: /var/www/siremis-1.0.0/siremis-web/js/sound.js, referer:
http://192.168.1.14/siremis/bin/
[Sat Jun 19 15:53:23 2010] [error] [client 192.168.1.101] File does not
exist: /var/www/siremis-1.0.0/siremis-web/js/sound.js, referer:
http://192.168.1.14/siremis/bin/
I am not 100% sure I fully understand my issue, but I think I'm on the right
track. I have a situation where Asterisk will drop calls a few seconds
after they are set up. What I believe is happening:
a=asterisk
k=kamailio
k --> INVITE --> a
a --> 100 TRYING --> k
a --> 183 RINGING --> k
a --> 200 OK --> k
k --> ACK -->a (this packet is never received by asterisk)
a --> 200 OK (retransmit) -->k
a --> 200 OK (retransmit) -->k
a --> 200 OK (retransmit) -->k
a --> no response to critical packet - terminating call
It doesn't appear that Kamailio is retransmitting the ACK. I would think
that would be part of the TM module, but perhaps I am not using it
properly. Is this possible or is there something else going on?
Thanks.
Hi people,
I am trying to install kamailio in a server with OpenSolaris (SunOS sun-test 5.11 snv_83 i86pc i386 i86pc), to make some tests, performance, capacity, etc.
I tried to compile on Solaris but, it dont works perfectly. :)
On the Internet and in the Manuals, I saw that there is a port from kamailio to solaris, but I dont find where to get this.
Docs: INSTALL
Supported architectures:
- Linux/i386, Linux/amd64, Linux/armv4l,
- FreeBSD/i386, OpenBSD/i386, NetBSD/sparc64
- Solaris/sparc64, Solaris/i386
So I have some questions:
Is there a port to Solaris?
Can I to compile Kamailio on Solaris (OpenSolaris)?
Are there any manuals to install kamailio on Sun Machines?
By the way, Can anybody tell me if the performance and capacity of kamailio on the solaris is better than linux?
Thanks...
Hello,
I am using Kamailio 1.5.4.
I read RFC3261 section 17 and the TM doc, I had a few questions :
Does t_relay() manage both the client and server transactions for the
packet being forwarded ? Meaning it would absorb retransmissions it
receives and would retransmit the relayed message when needed ?
Do I need to call t_release() when I call t_relay () ?
Where would I use t_checktrans(), t_release() and t_newtran() ? My
understanding is that if I want to receive a packet and process it
locally without forwarding it, I should use t_newtran(). This would be
good for things such as PUBLISH, PRESENCE and REGISTER ( even though
it's not a good idea ). So when do I release the transaction ? In the
reply route ? In the same run as I call t_newtran() ?
Does t_checktrans() absorb retransmissions that were started with
t_relay and t_newtran?
If you have any material that I could read that would help me better
understand the TM module, please say so.
Thanks,
David
Greetings,
I have a challenge which I can't imagine is unique to my situation. I have
agreements with multiple ITSP/SIP providers for inbound and outbound use
cases. I am trying to figure out a way to track and monitor - on a per
trunk basis - statistics about open dialogs. In the end, I would like to
have nice graphs that show my usage throughout the day... and most
importantly, my max open dialogs so I can determine if I am coming close to
my max sessions with any particular carrier.
Does anyone know of a tool which could help me track this? It looks like
Kamailio has some SNMP modules, perhaps SNMP + Cacti might be the common
solution?
Any input is greatly appreciated.
Thanks.
I'm attempting to run an existing SER config file under a recent build of sip-router. I get a syntax error on the t_on_failure("noroute") call that exists in my config file. The specific error is "bad expression: type mismatch (str instead of int)". I presume this means sip-router does not support named failure route labels. Is that correct?
Thanks,Steve
---
ISC Networking & Telecommunications
3401 Walnut Street, Suite 221A
Philadelphia, PA 19104
215-573-8396
215-898-9348 (fax)
Hey Guys,
Is there any module/function to parse and assign to AVPs parameters from
RPID? (in kamailio 1.5.3)
I'm using $(hdr(Remote-Party-ID){tobody.params}) to get the parameters but
maybe there's something already implemented to get privacy, screen, etc.
I'll be using this with P-Asserted-Identity as well.
Thanks in advance!
Uriel
