Hi,
Due to our presentation at CeBIT 2015 at the Open-Source Business
Alliance’s Booth, we’ve received a limited number of free tickets for
CeBIT 2015!
If you want a free ticket, please answer the following question:
- What is the key benefit of deploying VoLTE in a mobile network?
Send your answer to info(a)ng-voice.com (hint: we’ve got some benefits
posted on our website: www.ng-voice.com/volte), I will send the ticket
by email to the 5 best answers!
Don’t miss my presentation on Kamailio & VoLTE at the booth of the
Open-Source Business Alliance (www.osb-alliance.de), Tuesday, March
17th, ~2 PM, Hall 6, Stand H16, (410)!
Thanks,
Carsten
--
Carsten Bock
CEO (Geschäftsführer)
ng-voice GmbH
Schomburgstr. 80
D-22767 Hamburg / Germany
http://www.ng-voice.com
mailto:carsten@ng-voice.com
Office +49 40 5247593-0
Fax +49 40 5247593-99
Sitz der Gesellschaft: Hamburg
Registergericht: Amtsgericht Hamburg, HRB 120189
Geschäftsführer: Carsten Bock
Ust-ID: DE279344284
Hier finden Sie unsere handelsrechtlichen Pflichtangaben:
http://www.ng-voice.com/imprint/
Hi,
I'm continuously seeing the following WARNING in my logs:
WARNING: db_mongodb [mongodb_dbase.c:454]: db_mongodb_get_columns():
unhandled data type column (instance) type id (10), use DB1_STRING as
default
It looks like the WARNING appears when data is fetched from the location
collection. The instance field is always null in my case.
Thanks,
Mickael
Hello,
We are exploring kamailio source for use in our VOIP solution. This is the
first time we are looking at open source. We have few doubts in using
kamailio for providing our VOIP service. Under GPL any customization
(customization mainly with respect to interacting with our proprietary
applications e.g. Billing server) we do to kamailio also has to be provided
to the end user. Our doubt is whether we can procure commercial rights to
our customised code?
Kindly clarify on this.
Best Regards,
Shankar
Hi,
The dialog module documentation remains unclear about the order of
operations with regard to when to call dlg_manage() or set the
transaction flag.
My impression is that dlg_manage() only registered TM callbacks, so it
doesn't matter when you call it, as long as it's before t_relay().
However, the documentation neither confirms nor denies this.
So, this raises the questions:
1) Is this okay?
set_dlg_profile("caller", "$fU");
dlg_manage();
...
t_relay();
Or do I need to do this?
dlg_manage();
set_dlg_profile("caller", "$fU");
...
t_relay();
2) What about setting dialog-persistent variables? Is this okay?
$dlg_var(account_id) = 49555;
dlg_manage();
If so, where does the variable go if I never call dlg_manage() because
the call is aborted beforehand, e.g.
$dlg_var(account_id) = 49555;
sl_send_reply("403", "Forbidden");
exit;
dlg_manage();
...
t_relay();
3) Any other gotchas or caveats in relation to the order of operations?
I suppose my preference would be to set the dialog profiles in various
places throughout call processing and call dlg_manage() at the very end,
right before t_relay(). Is this acceptable?
Thanks,
-- Alex
--
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
United States
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
I have an issue that from what I can tell is related to forking. My
configuration works just fine, and makes a request out to execute curl to
hit an external API:
$var(my_query) = "curl -m 2 -s http://x.x.x.x/service/method?fromAddr=" +
$var(fu) + "\&toAddr=" + $var(tu);
Normally this gives us back a nice JSON response, no problems. All I do is
enable snmpstats module, and it fails.
Mar 5 16:51:30 server-name /usr/sbin/kamailio[1051]: ERROR: exec
[exec.c:291]: exec_avp(): cmd curl -m 2 -s
http://x.x.x.x/service/method?fromAddr=me@mydomain.com\&toAddr=someone@mydo…
failed. exit_status=-1, errno=10: No child processes
I have edited this for security reasons of course. If I disable SNMP, the
problem goes away.
Thoughts?
kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour
I found this guide for an older version of Kamailio, but there seems to
be some issues when trying to use it on the latest stable version.
When I try to start Kamailio i get this :
[....] 0(17153) INFO: tls [tls_init.c:399]: init_tls_compression(): tls:
init_tls: disabling compression... 0(17153) ERROR: <core> [cfg.y:3286]:
yyparse(): cfg. parser: failed to find command force_rtp_proxy (params
0) 0(17153) : <core> [cfg.y:3426]: yyerror_at(): parse error in config
file /etc/kamailio/kamailio.cfg, line 729, column 19: unknown command,
missing loadmodule? 0(17153) ERROR: <core> [cfg.y:3286]: yyparse(): cfg.
parser: failed to find command force_rtp_proxy (params 0) 0(17153) :
<core> [cfg.y:3426]: yyerror_at(): parse error in config
fil[FAILc/kamailio/kamailio.cfg, line 806, column 19: unknown command,
missing loadmodule? ERROR: bad config file (2 errors) ...
It seems it cannot load the rtpproxy module or that force_rtp_proxy
command does not exist. For Debian the modules are located in
/usr/lib/x86_64-linux-gnu/kamailio/modules/ and it was found there,
seems to be loaded in the config file (line 5 and 235). According to the
rtpproxy doc the module has this command.
Does anyone have an idea what could be the issue ?
I am planning to update this guide for the current version once I fixed
all the issues.
The most easy issues are the db/usernames/password and the path to the
modules.
--
Med venlig hilsen
Tom Braarup Cuykens
plusTEL.dk
Hi there,
I'm using kamailio version 4.2.2 to a short time, and I'm using lua scripts
to build some avps with information needed to make calls(lua functions uses
redis as source data to build avp data.), what i have noticed today making
some tests is that when i make a call my lua function:
if(!lua_run("get_prefs","$fU","$fd"))
{
xdbg("SCRIPT: failed to execute lua function!\n");
}
kamailio gets the avp's after send the invite to the destine,(the invite
will fail because kamailio needs avp information).
it seems that kamailio calls the lua_run in asynchronous way, i never
noticed that before in old version. is that a normal behavior or it is an
issue in this version?
Best Regards
--
Cumprimentos
José Seabra
Hello Everyone,
I set up tcp and udp for right now to test all config and dialog is not established correctly. Call between extensions not working. Call from extesnsion to extension going right a way to voicemail. Any help thank you.
Here one call debug.
http://fpaste.org/195572/14258553/
Slava.
From: "Slava Bendersky" <volga629(a)networklab.ca>
To: miconda(a)gmail.com
Cc: "sr-users" <sr-users(a)lists.sip-router.org>
Sent: Tuesday, March 3, 2015 9:08:19 AM
Subject: Re: [SR-Users] kamailio asterisk
Hello Daniel,
Is notify should follow subscribe routes ?
Here SUBSCRIBE completely look normal to me.
<--- SIP read from UDP:10.18.130.46:5060 --->
SUBSCRIBE sip:10102@networklab.ca SIP/2.0
Record-Route: <sip:kamailio_pub_ip:5081;transport=tls;lr=on;ftag=08c99307d3;nat=yes>
Accept: application/simple-message-summary
Via: SIP/2.0/UDP 10.18.130.46;branch=z9hG4bKdee2.b83c2028dc57c641d2b8cad968347d6a.0;i=2
Via: SIP/2.0/TLS 192.168.0.16:5063;rport=5063;received=client_pub_ip;branch=z9hG4bK0c54032bd8905d262
Max-Forwards: 69
From: "10102" <sips:10102@networklab.ca>;tag=08c99307d3
To: <sips:10102@networklab.ca>
Call-ID: cea703eb8c21f709
CSeq: 1096440084 SUBSCRIBE
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, REFER, SUBSCRIBE, UPDATE
Allow-Events: message-summary
Authorization: Digest username="10102",realm="networklab.ca",nonce="29ac01f1",uri="sips:10102@networklab.ca",response="0298c2a93bdd4969a3cd0b757d671dbb",algorithm=MD5
Contact: <sips:10102@client_pub_ip:5063>;audio
Event: message-summary
Expires: 3600
Supported: eventlist, replaces, timer
User-Agent: Media5-fone/4.1.6.3283 Android/4.1.2
Content-Length: 0
Path: <sip:outbound@10.18.130.46;lr;received=sip:client_pub_ip:5063%3Btransport%3Dtls>
P-hint: outbound
Slava
From: "Daniel-Constantin Mierla" <miconda(a)gmail.com>
To: "Slava Bendersky" <volga629(a)networklab.ca>
Cc: "sr-users" <sr-users(a)lists.sip-router.org>
Sent: Tuesday, March 3, 2015 1:50:03 AM
Subject: Re: [SR-Users] kamailio asterisk
Hello,
I pushed a fix for it -- the issue was in a LM_DBG() trying to print te address for local socket, but it was not set yet, it was affecting only when running in debug mode with debug=3. I pushed the patch to branch 4.2, so if you want the fix, then you have to install from git branch 4.2.
For REGISTER, you need to add the Path header (see path module), otherwise the OPTIONS and INVITE sent out to the phone are not going via Kamailio. You have to check the version of your Asterisk to be one that supports Path.
Cheers,
Daniel
On 27/02/15 14:40, Slava Bendersky wrote:
Hello Daniel,
Here paste for gdb
http://fpaste.org/191338/25043949/
I got REGISTER and SUBSCRIBE start working correctly I see on asterisk correct record routes and sip traffic flow, but when asterisk or client ( soft phone) send OPTIONS or NOTIFY can't get properly relay it.
This is SUBSCRIBE route.
<--- Transmitting (NAT) to 10.18.130.46:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.18.130.46;branch=z9hG4bKf852.e0223f39c2bbad8366fdf1b7cb22b336.0;i=8;received=10.18.130.46;rport=5060
Via: SIP/2.0/TLS 192.168.88.252:5062;received=Client public ip;branch=z9hG4bK0bbe1f7d27257bba9;rport=5062
Record-Route: <sip:10.18.130.46;r2=on;lr=on;ftag=a185d974ec;nat=yes>
Record-Route: <sip:PUBLIC_KAMAILIO_IP:5081;transport=tls;r2=on;lr=on;ftag=a185d974ec;nat=yes>
From: "Slava Bendersky" <sips:10101@networklab.ca> ;tag=a185d974ec
To: <sips:10101@networklab.ca> ;tag=as00757d3e
Call-ID: b08adb1ad1804a83
CSeq: 236711034 SUBSCRIBE
Server: FPBX-2.11.0(11.15.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 3600
Contact: <sip:10101@10.18.130.50:5060> ;expires=3600
Content-Length: 0
Slava.
From: "Daniel-Constantin Mierla" <miconda(a)gmail.com>
To: "Slava Bendersky" <volga629(a)networklab.ca>
Cc: "sr-users" <sr-users(a)lists.sip-router.org>
Sent: Friday, February 27, 2015 6:42:33 AM
Subject: Re: [SR-Users] kamailio asterisk
Hello,
I asked for the wrong command, as the bt full was already sent before -- I wanted the output from gdb for:
p *tcpconn
Daniel
On 27/02/15 04:10, Slava Bendersky wrote:
BQ_BEGIN
Hello Daniel,
Here bt full from back trace.
http://fpaste.org/191207/50064491/
Slava.
From: "volga629" <volga629(a)skillsearch.ca>
To: miconda(a)gmail.com , "Slava Bendersky" <volga629(a)networklab.ca>
Cc: "sr-users" <sr-users(a)lists.sip-router.org>
Sent: Thursday, February 26, 2015 9:56:48 PM
Subject: Re: [SR-Users] kamailio asterisk
Hello Daniel,
I tried $rz option on top of request route and that where I see wrong request uri like sip:sips : . And as far I can see it happenes only for SUBSCRIBE INVITE and NOTIFY.
if($rz=="sips") {
$ru = "sip" + $(ru{s.substr,4,0});
}
Slava.
Sent from mobile device typos are expected.
From: Daniel-Constantin Mierla <miconda(a)gmail.com>
Sent: Feb 25, 2015 1:04 PM
To: Slava Bendersky
Cc: sr-users
Subject: Re: [SR-Users] kamailio asterisk
Hello,
On 25/02/15 17:19, Slava Bendersky wrote:
BQ_BEGIN
Hello Daniel,
substr you suggested didn't worked.
See my previous email.
your previous email didn't say anything about the results. That's why I asked. Be sure you don't have those spaces that are in the email you wrote. Also, I had more parenthesis in the parameter of the subst_uri().
Or you can try the alternative with:
if($rz=="sips") {
$ru = "sip" + $(ru{s.substr,4,0});
}
I asked for more details from the backtrace to confirm that what I found is the cause for the crash in this case -- see one of my previous emails from today.
Cheers,
Daniel
BQ_END
--
Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015
Berlin, Germany - http://www.kamailioworld.com
BQ_END
--
Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015
Berlin, Germany - http://www.kamailioworld.com
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
In terms of implementing "group" calling via the append_branch feature of both
alias_db_lookup followed by lookup_branches, I'm looking for a reliable way to
ensure that if the caller happens to be a member of the group (list of
branches), the branch that's created to the original caller is dropped.
Originally, I was thinking of comparing $rU and $fU in branch_route, but this
would limit the ability for one contact of an AOR to call another contact of
the same AOR.
Can anyone offer an example of an efficient method to accomplish this?
--
Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery
8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E
