That is not entirely true. We use MySQL without any kind of issues via
ODBC driver.
DanB
On 14.02.2014 10:18, sr-users-request(a)lists.sip-router.org wrote:
> Asterisk would be suitable solution, FREESwitch is also fine, but with
> FREESwitch latest version uses postgreSQL, and removed support for MySQL,
> if you are comfortable with postgreSQL for reporting stuffs like that then
> you can proceed with FREESwitch.
>
> Regards
>
>
>
> Thank you with regards,
> Gopalkrishnan N.
> Mob: +91 99404 91346
> VoIP call -sip:saigop@gtalk2voip.com
Hi All List Members,
I have worked on kamailio for one year. It is a great open source Skype
like solution.
Now i have to deploy a new solution with new requirements. Kindly
suggest that what will be
the best open source application will fulfill my requirements. I need
the following facilities:
(1) Call Recording (Outbound/Inbound)
(2) On Demand Recording
(3) Dictation Service
(4) Complex Inbound Routing
(5) IVR (Interactive Voice Response)
(6) Auto Attendant
(7) Client Recordings Mgmt & Playback Portal
(8) Multi-level Client hierarchy / Subscription Plans
(9) Voicemail
(10) Time based Routing (Message Box)
(11) Call Queuing & Conferencing
(12) Bulk Download & Emailing
(13) Call Whisper
(14) Smartphone apps (SIP clients)
Regards,
/
Hi,
Lately i am experiencing some issues with the sqlops module.
let's say, for example, that i use sqlops to query some information about
the subscriber when an INVITE arrives.
I use kamailio 3.3.2, MySQL 5.6 and the DB server is remote and not local.
Usually, the DB result is very fast and the call continues. But, when the
DB is unavailable, or slow, the result is delayed or lost.
In this case:
1. What is the query time out in kamailio?
2. If the database is available but slow, does it mean that no new INVITE
will be dealt by the current child process until the sqlops is gone?
3. does anyone has a good idea for managing sqlops in a call?
Thanks,
Uri
Hi,
I'm pretty ignorant about Kamailio, sorry for it and possible
terminology mistakes.
I'd like to allow any user to register. Ideally, I'd like to make some
script (Lua/whatever) function to check some data encoded in a
username and allow registering based on that username only.
I undefined WITH_AUTH in kamailio.cfg, and it used to work for me; but
sometimes I'm starting to get
"SIP Progress: 100 trying -- your call is important to us ()"
and the call never reaches the other party.
Here's a sample of my logs during the failed call:
https://gist.github.com/singalen/c343f09b67dcf7731fe0
What is the problem here?
I start and stop the clients all the time, so there might be some
garbage registrations accumulated. I assume I also need a user
registration to discard all his previous registrations?
Thank you!
Hi
I tried the binary operation using $var as per the documentation and its
seems not working. Can somebody help me to understand it.
http://kamailio.org/dokuwiki/doku.php/pseudovariables:3.1.x
$var(a) = 3 + (7&(~2));
if( [ $var(a) & 4 ] ) {
xlog("var a has third bit set\n");
}
The format with square bracket in the conditional statement is showing
parse error when i load the cfg. So it tried like below and that too not
working
$var(a) = 8;
if( $var(a) & 4 ) {
xlog("var a has third bit set\n");
}
Please let me know if something missing here.?
Thanks
Jijo
Hello, I have set up call forwarding in Kamailio using user_preference table. When I make a call from a local extension the callfwd funcition works. The call gets forwarded to an external did number. The problem happens when and external call ( from a DID) comes to this extension the call does not get forwarded. I had to comment out the following lines in the config file to make it work:
# only local users allowed to call
# if((from_uri!=myself)) {
# sl_send_reply("403", "Not Allowed");
# exit;
# }
My question is if it is a risky think to comment out these lines? If so, what are my options.
Thank you,
Arun
I am having problems with calls from webrtc to kamailio forwarded to Asterisk
These are snippet of the debug logs
Asterisk
CSeq: 4910 BYE
Reason: SIP ;cause=488; text="Not Acceptable Here"
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.0
Content-Length: 0
Jssip
Cause: Bad Media Description
Origin: remote
Searching on google I get some indication this is to do with ice config?
Please can some one suggest if this is so.
In my scenerio the webrt clients will only call to the asterisk server (and not to other user agent).
Considersing this I think maybe can do without ice.
Is it possbile to disable ice.
Kamailio is having a crash (Segfault at 18 IP - error 4 in auth_db.so) when
registering a cisco E20, via SIP
Here is a full dbg backtrack, and the core dump, but have no ideas about
how to proceed.
Im using Kamailio 4.1.1:
Got this Error at /var/log:
[14629359.586980] kamailio[982]: segfault at 18 ip 00007f034db8dcfd sp
00007fff34702950 error 4 in auth_db.so[7f034db85000+c000]
Kamailio Version:
----------
version: kamailio 4.1.1 (x86_64/linux) d6ffa1
flags: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS,
DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC,
DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE,
USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
id: d6ffa1
compiled on 16:28:48 Jan 29 2014 with gcc 4.4.5
-------------
Thanks in Advance.
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA256
The v1.9.0 release has been tagged (r10958) and made available for
download:
https://www.resiprocate.org/files/pub/reSIProcate/releases/
The release info page is here:
http://www.resiprocate.org/ReSIProcate_1.9_Release
and a direct link to the ChangeLog in SVN:
https://svn.resiprocate.org/rep/resiprocate/tags/resiprocate-1.9.0/ChangeLog
Debian, Fedora and Ubuntu packages have been updated, automated builds
are in progress and they should be available shortly.
Highlights of the v1.9.0 release:
* Session/registration accounting
* SIP over WebSocket support for WebRTC, with cookie authentication option
* Python scripting for repro routes
* repro dynamically-loaded routing plugins
* UAS PRACK
* Improvements to daemon processes for UNIX/Linux users
* Android support
* Multiple users in reTurn
* Many stability/bug fixes
SHA224 checksums for the files:
resiprocate-1.9.0.tar.gz
914b01c4633e0fc03f085fdfc77199577ac079c848c35e7f04c8d344
resiprocate-contrib-1.9.0.tar.gz
444b05ae9986a43d25347e7ea4908ab3d02a1bba4d70f0744de1da11
Contributors to v1.9.x include Scott Godin, Eric Rescorla, Adam Roach,
Byron Campen, Jason Fischl, Francis Joanis, Matthias Moetje, Catalin
Constantin Usurelu and Daniel Pocock. We are also very grateful to
all those who contributed feedback through the mailing list, testing
the beta packages in Debian and all past contributors to previous
releases of the reSIProcate project.
Any and all feedback is welcome
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