Hi Guys
At times when I want to get the current user list using
/usr/local/sbin/openserctl ul show I get this error
ERROR: Error opening OpenSER's FIFO /tmp/openser_fifo
ERROR: Make sure you have the line 'modparam("mi_fifo",
"fifo_name", "/tmp/openser_fifo")' in your config
ERROR: and also have loaded the mi_fifo module.
I dont think this is a configuration error, as killing and restarting
openser does fix this.
With Regards
Ali Jawad
System Administrator
Splendor Telecom (www.splendor.net <outbind://147/www.splendor.net> )
Beirut, Lebanon
Phone: +961 1 373725
Fax: + 961 1 375554
Hi everybody,
I want to announce the launch of SIREMIS Project, aiming to build web
management interface for Kamailio (OpenSER). Later will follow also the
sip-router.org project.
The project was developed so far by Asipto - http://www.asipto.com -
using OpenBiz PHP framework - the target was to make it easier to extend
for non-web developers. Soon, the project sources will be hosted on
sourceforge.net SVN (project name: siremis).
Homepage of the project is:
- http://siremis.asipto.com
The first beta release is available for download and includes the features:
- subscriber, database aliases and speed dial management
- location table view
- dispatcher, prefix-domain translation and least cost routing (lcr)
management
- access control lists (user groups) and permissions management
- accounting records, missed calls and call data records view
- hash table, dial plan table and user preferences table management
- off-line message storage view
- it was tested with MySQL but should support other sql-like DBs
Internal design:
* developed on top of a framework: PHP/Web2.0/Ajax
* XML Metadata driven extensions
* Model-View-Controller (MVC) architecture
You can see some screenshots at:
- http://siremis.asipto.com/photos/siremis-screenshots/
For few days, a demo site can be accessed at:
- http://siremis.asipto.com/demo/
- username: admin and password: admin
Download and installation details:
- http://siremis.asipto.com/install/
I hope you will enjoy it and help to extend it. The plan is to release
first stable version more or less at the same time with Kamailio
(OpenSER) 1.5.0.
Your feedback is welcome!
Best regards,
Ramona
It worked finally,
As per hint I did change
endif
DEFS+=$(CONFUSEDEFS)
LIBS=$(CONFUSELIBS)
To
endif
CONFUSEDEFS=-I$(LOCALBASE)/include -I/usr/local/include -I/opt/include
-I/usr/sfw/include
CONFUSELIBS=-L$(LOCALBASE)/lib -L/usr/local/lib -L/usr/sfw/lib
-L/opt/lib -lconfuse
DEFS+=$(CONFUSEDEFS)
LIBS=$(CONFUSELIBS)
Now I got an error that
Jan 21 14:36:24 [24863] INFO:carrierroute:bind_data_loader: use file as
configuration sourceJan 21 14:36:24 [24863]
ERROR:carrierroute:bind_data_loader: can't stat config file
But this is a totally different story :), let me hit the carrierroute
documentation.
Thanks A million
-----Original Message-----
From: Henning Westerholt [mailto:henning.westerholt@1und1.de]
Sent: 2009-01-21 15:32
To: Ali Jawad
Subject: Re: [Kamailio-Users] Problem With Carrierroute on Solaris
On Wednesday 21 January 2009, you wrote:
> At current it is
>
> /opt/lib -lconfuse <-- note the L
>
> should I make that
>
> /opt/lib -confuse -lgcc
>
> or
>
> /opt/lib -lconfuse -lgcc <- double L
The last one, with two 'l' is correct. The '-l' tells the compiler that
this is a linker configuration, after the l comes the library name. '-L'
is a library location.
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and JAVA
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- Experience with tools for Network
analysis like Ethereal
- VoIP knowledge with experience on
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and groups at all levels
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Hello,
Version 1.1.0 of SEMS, the SIP Express Media Server, has been released.
SEMS is a free, high performance, extensible media and application
server for SIP (RFC3261) based VoIP services.
New in version 1.1 is:
* DSM state machine scripting (it's cool!)
* an (experimental) ISDN gateway module
* binrpc: MT (SER->) and connection pool (->SER)
* MT xmlrpc server
* controlled server shutdown
* improved logging
* g722 in 8khz compat mode
* out of dialog request handling for modules & dialogs without
sessions
* audio file autorewind, AmAudio mixing
* SIP and media IP separately configurable
* UID/DID support for voicemail/-box/annrecorder
* and quite some bugs and mem leaks fixed, documentation, etc.
You can get the sources from
ftp://ftp.iptel.org/pub/sems/sems-1.1.0.tar.gz
and find documentation and packages at
ftp://ftp.iptel.org/pub/sems/1.1/1.1.0/
and a debian repository (etch/lenny) at
deb http://ftp.iptel.org/pub/sems/debian etch free
deb-src http://ftp.iptel.org/pub/sems/debian etch free
Further documentation, links to lists, tracker etc is available at the
project's homepage: http://iptel.org/sems/
I would like to thank for all contributions and patches, this time from
Stefan Sayer, Raphael Coeffic, Grzegorz Stanislawski,
Bogdan Pintea, Greger Teigre, Rui Jin Zheng,
Alfred E Heggestad, Juha Heinanen, Peter Lemenkov,
Peter Loeppky, Robert Szokovacs, Jeremy A, Alex Gradinar
Apologies if you receive multiple copies of this message.
Best Regards
Stefan Sayer
--
Stefan Sayer
VoIP services
stefan.sayer(a)iptego.com
www.iptego.com
IPTEGO GmbH
Am Borsigturm 40
13507 Berlin
Germany
Amtsgericht Charlottenburg, HRB 101010
Geschaeftsfuehrer: Alexander Hoffmann
Hi,
How can I extract the URIs in "P-Asserted-Identity" when there are multiple "P-Asserted-Identity" headers.
Variable $ai return the URI in the first "P-Asserted-Identity" header.
Thanks,
Alex
_________________________________________________________________
Windows Live™: Keep your life in sync.
http://windowslive.com/howitworks?ocid=TXT_TAGLM_WL_t1_allup_howitworks_012…
Hi All
I did remove carrierroute from the excluded modules and I did recompile,
all went smooth however when I try to run the server with the module
loaded I get this error :
fatal: relocation error: file carrierroute.so: symbol cfg_init:
referenced symbol not found
echo $LD_LIBRARY_PATH
/opt/csw/mysql5/lib/mysql/:/opt/csw/lib/:/opt/csw/include/:/opt/csw/incl
ude/openssl/:/usr/local/lib/
find /usr/ -name "*confuse*"
/usr/local/include/confuse.h
/usr/local/lib/libconfuse.a
/usr/local/lib/libconfuse.la
/usr/local/lib/libconfuse.so
/usr/local/lib/libconfuse.so.0
/usr/local/lib/libconfuse.so.0.0.0
Please advice.
Hi list,
Has anyone a suggestion to solve this error or where it might come from?
I'm using carrierroute module to route calls through to different domains,
but I can't make it work properly.
[root@localhost kamailio]# Jan 20 12:05:52 localhost
/sbin/kamailio[24686]: INFO:carrierroute:cr_do_route: uri 123456 was
rewritten to sip:123456@200.100.200.100
Jan 20 12:05:52 localhost /sbin/kamailio[24686]:
ERROR:tm:t_forward_nonack: no branch for forwarding
Jan 20 12:05:52 localhost /sbin/kamailio[24686]: ERROR:tm:w_t_relay:
t_forward_nonack failed
Jan 20 12:05:52 localhost /sbin/kamailio[24680]:
CRITICAL:tm:t_should_relay_response: pick_branch failed (lowest==-1) for
code 484
Thanks in advance.
Juan.-
Hello,
following discussions on the mailing lists and tracker related to string
and URI/AoR comparison, textops and siputils introduce new dedicated
functions:
- cmp_str(str1, str2) - compare two strings case sensitive
- cmp_istr(str1, str2) - compare two strings case insensitive
- cmp_uri(str1, str2) - compare two URIs as per RFC3261(up to parameters
comparison rules)
- cmp_aor(str1, str2) - compare two address of records
http://kamailio.org/docs/modules/devel/textops.htmlhttp://kamailio.org/docs/modules/devel/siputils.html
Just to refresh, the string comparison using config operators (==, <=,
...) are case insensitive (== is equivalent to cmp_istr(...)).
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://www.asipto.com
Hello,
I hope this is the right forum to post a request for consulting services. We
have a requirement for setting up a HA OpenSIPS 1.4.2 installation with a
primary (active) and secondary (failover) server running CentOS 5.x. SIP
messages from the active server must come from a virtual IP address and
carrier gateways receiving our call traffic must remain oblivious to any
server failure. Failover must occur in < 5 seconds.
We would like to see a simple solution based on open source components like
vrrp or Linux-HA, including the SIP heartbeat mechanism and failover per se,
as well as the recovery process once the primary server recovers from the
failure causes. The work will be done in conjunction with our Linux
Administrator and IP Telephony Engineer and must be documented, at least in
a basic fashion. Failover testing under load will be performed prior to
project acceptance.
If you are an expert in OpenSER HA implementations and are interested to
help us, please respond via this forum or directly to my e-mail address,
serge A T elasticall D O T com. Thanks in advance.
Best Regards,
Serge
--
View this message in context: http://www.nabble.com/Vrrp-Linux-HA%2C-paid-consulting-opportunity-tp215460…
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