( I posted a same message through gmane interface. gmane is not
working right now at my place, So i can't respond to the replies i got
there).
I have now these users :
arif@khost:~$ kamctl db show subscriber
database engine 'MYSQL' loaded
Control engine 'FIFO' loaded
+----+----------+---------------+------------+---------------+----------------------------------+----------------------------------+------+
| id | username | domain | password | email_address | ha1
| ha1b | rpid |
+----+----------+---------------+------------+---------------+----------------------------------+----------------------------------+------+
| 1 | test | 192.168.7.143 | testpasswd | |
ad87b307789553f95799738d87246ca0 | e211f539e22a1ad2cc0f3c07056c3517 |
NULL |
| 4 | test | 127.0.0.1 | ps | |
bca93be084f6c7a291c98fe9b0077b0b | d9ce12a4ec42448f65287de912cb9d43 |
NULL |
+----+----------+---------------+------------+---------------+----------------------------------+----------------------------------+------+
Now still if i send OPTION query
sofsip> o sip:test@127.0.0.1:5070
UA: OPTIONS to sip:test@127.0.0.1:5070
sofsip> UA: OPTIONS 404 Not Found
And from kamailio log :
(5827) DEBUG: registrar [lookup.c:158]: lookup(): 'test' Not found in usrloc
--
-Cheers
-Arif
Hello ,
Is the $DLG_lifetime supposed to be accessed in theevent_route[tm:local-request] ?
We are trying to access it and it always reports null with kamailio 4.0.0 .
We found a reference about this here:
http://www.kamailio.org/pub/kamailio/3.3.3/ChangeLog
.....
dialog(k): run event route after setting cfg dlg vars - in this way they (e.g., $DLG_lifetime) should be accessible in event route (cherry picked from commit 2cdded28d9968a0b78f5ec8329ae6983d9ea77a9)
.....
Apparently it should have been possible.
Any hints on this ?
Thank you!
Regards,
Dragos & Federico
Hi,
I am trying to customize the reason phrase using t_reply.
I noticed that we can not set the reason phrase when the code is in the 6xx
range
Kamailio TM is setting the reason phrase to "unknown" or to a default one.
For example :
t_reply("603","My custom reason phrase");
Will send "603 Decline"
Hi guys,My name is Gustavo Salazar, I'm from Quito-Ecuador (South-America), and I need some info about Kamailio. I'm studying a Master degree of Networking and Data Communications, and I want to build a VoIP project with Kamailio.I want to integrate Kamailio with Trixbox to improve the security levels of the IP-PBX, also I want to use my cisco ip phones 7960G with them, but I need some help with how and where to configure the Kamailio to be connect to the Trixbox server.I made some research about it, and in some pages I read that I can install Kamailio over some Linux Distro like Ubuntu or CentOS, and in other that I can install it on the asterisk (Trixbox) server itself. Could you help me with some examples of configuration, where to configure the extensions. I think is in Kamailio.cfg file, but I don't know how to do it.Please help me with it. I'll be so grateful with you for this support.Regards from Ecuador Gustavo Salazar
Good morning Martin.
RFC3261 says:
"
.... [Page 85]
RFC 3261 SIP: Session Initiation Protocol June 2002
...
If the server retransmits the 2xx response for 64*T1 seconds without
receiving an ACK, the dialog is confirmed, but the session SHOULD be
terminated. This is accomplished with a BYE, as described in Section
15.
"
Why 32 seconds? Because almost always T1 = 500 ms. -> 64 * 500ms = 32 segs
Why 200 OK are being re-transmited? Because someone in someplace is stopping
200 OKs or ACKs.
Look for a router/FW and disable ALG for SIP support.
Look for errors on headers.
Best regards.
--
Eduardo Lejarreta.
-----Mensaje original-----
De: sr-users-bounces(a)lists.sip-router.org
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sr-users-request(a)lists.sip-router.org
Enviado el: viernes, 21 de junio de 2013 0:18
Para: sr-users(a)lists.sip-router.org
Asunto: sr-users Digest, Vol 97, Issue 79
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Today's Topics:
1. Fwd: Call breaks after 32 seconds (Martin Mou?ka)
----------------------------------------------------------------------
Message: 1
Date: Wed, 19 Jun 2013 14:36:53 +0200
From: Martin Mou?ka <moucka.m(a)gmail.com>
To: sr-users(a)lists.sip-router.org
Subject: [SR-Users] Fwd: Call breaks after 32 seconds
Message-ID:
<CAJY3ant7LG1xgO4ZjmSa0j-HfBbZtPUSvZ3kDkWiq7KD6XuhEw(a)mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Hello,
we have problem here with outbound calls. Many devices for example Zyxel,
WELL 8820IP, Interbell and some software clients Ekiga v3.3.2 and SJ Phone
drops call after 32 seconds. I think it could be some new standard from
2007, because Ekiga v4.0 seems ok and mentioned hardware devices don't have
new firmware since this year. Can you confirm that please? I'm attaching
some captures.
Thank you
Good morning,
I have serveral servers who are registered with the same AOR in Kamailio.
Every server registers with a Q-value indicating its availability. When a
user sends an INVITE request through Kamailio to that AOR, Kamailio should
send the request to the server with the best Q-value.
Using t_load_contacts I obtain a q-ordered destination set but the
t_next_contacts uses parallel forking when there are two servers with the
same q-value.
Is there any way to tell t_next_contacts to use only ONE destination?
Thank you in advance,
Regards,
--
*Jan **Gaida*
Ingeniero Desarrollo Software C/ Marconi 3 (PTM)
28760 Tres Cantos
Spain
jan.gaida(a)grupoamper.com | www.grupoamper.com
--
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Hello,
We have a system which Kamailio fails to compile mysql support on, even
though the library is installed.
The mysql packages are:
MySQL-client.x86_64
5.5.28-1.rhel5 installed
MySQL-devel.x86_64
5.5.32-1.rhel5 installed
MySQL-server.x86_64
5.5.28-1.rhel5 installed
MySQL-shared-compat.x86_64
5.5.28-1.rhel5 installed
The error from the Kamailio install is:
/usr/bin/ld: cannot find -lmysqlclient
collect2: ld returned 1 exit status
make[1]: *** [db_mysql.so] Error 1
make: *** [install-modules] Error 1
make: Leaving directory `/usr/src/kamailio-3.3.2'
Would anyone be able to advise please? Thanks in advance.
--
David Cunningham, Voisonics
http://voisonics.com/
USA: +1 213 221 1092
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 8063 9019
My kamailio implementation sits in front of a group of asterisk servers, when a call comes into kamailio it does a series of checks on the location of the src and dst to determine if the call is allowed. If the src is PSTN and called user is found in the location table the call is passed to an asterisk server for further processing of call forwarding options / voicemail / ring time setting etc. However if the called account is not found (not currently registered) the call is then discarded and voicemail etc can never be accessed.
I can add a avp_db_query to fetch user from the subscriber table vs location table lookup. Just wondering if there are any built in functions or other efficient ways to handle this that others are doing.
Thanks for any input
-Dan
Hi,
I'm trying to understand intricacies of SIP protocol.
I'm installed a stock kamailio from git repo.
kamcmd> core.version
kamailio 4.0.2 (x86_64/linux) f87866
Now i'm trying to send OPTION request by "sipsak".
I've added two users :
arif@khost:~$ kamctl db show user
...........
+----+----------+---------------+------------+---------------+----------------------------------+----------------------------------+------+
| id | username | domain | password | email_address | ha1
| ha1b | rpid |
+----+----------+---------------+------------+---------------+----------------------------------+----------------------------------+------+
| 1 | test | 192.168.7.143 | testpasswd | |
ad87b307789553f95799738d87246ca0 | e211f539e22a1ad2cc0f3c07056c3517 |
NULL |
| 2 | test2 | localhost | testpasswd | |
46d8618f5b652e4aeff3ffe52f373028 | bb8564b436dcfa183a8228244d8527ea |
NULL |
+----+----------+---------------+------------+---------------+----------------------------------+----------------------------------+------+
Now The OPTION message i'm trying :
OPTIONS sip:test_local@localhost SIP/2.0
Via: SIP/2.0/UDP 127.0.1.1:60518;branch=z9hG4bK.374c1583;rport;alias
From: sip:sipsak@127.0.1.1:60518;tag=74766f41
To: sip:test_local@localhost
Call-ID: 1953918785(a)127.0.1.1
CSeq: 1 OPTIONS
Contact: sip:sipsak@127.0.1.1:60518
Content-Length: 0
Max-Forwards: 70
User-Agent: sipsak 0.9.6
Accept: text/plain
kamailio is respoding with :
SIP/2.0 404 Not Found
After digging in kamailio log :
2(17356) DEBUG: registrar [lookup.c:158]: lookup(): 'test_local' Not
found in usrloc --
The code reads :
155 ul.lock_udomain(_d, &aor);
156 res = ul.get_urecord(_d, &aor, &r);
157 if (res > 0) {
158 LM_DBG("'%.*s' Not found in usrloc\n",
aor.len, ZSW( aor.s));
So it seems i'm doing something very wrong.
Although in my config :
106 #!define WITH_MYSQL
107 #!define WITH_AUTH
108 #!define WITH_USERLOCDB
After going through the RFC3261's section 11.2,
It seems this OPTION message should work and generate a 200(OK).
What it is i'm doing wrong?
--
-Cheers
-Arif
Hi,
I'm trying to understand intricacies of SIP protocol.
I'm installed a stock kamailio from git repo.
kamcmd> core.version
kamailio 4.0.2 (x86_64/linux) f87866
Now i'm trying to send OPTION request by "sipsak".
I've added two users :
arif@khost:~$ kamctl db show user
...........
+----+----------+---------------+------------+---------------+----------------------------------+----------------------------------+------+
| id | username | domain | password | email_address | ha1
| ha1b | rpid |
+----+----------+---------------+------------+---------------+----------------------------------+----------------------------------+------+
| 1 | test | 192.168.7.143 | testpasswd | |
ad87b307789553f95799738d87246ca0 | e211f539e22a1ad2cc0f3c07056c3517 |
NULL |
| 2 | test2 | localhost | testpasswd | |
46d8618f5b652e4aeff3ffe52f373028 | bb8564b436dcfa183a8228244d8527ea |
NULL |
+----+----------+---------------+------------+---------------+----------------------------------+----------------------------------+------+
Now The OPTION message i'm trying :
OPTIONS sip:test_local@localhost SIP/2.0
Via: SIP/2.0/UDP 127.0.1.1:60518;branch=z9hG4bK.374c1583;rport;alias
From: sip:sipsak-jEV4NMQqubUqMp+WYRx65w@public.gmane.org:60518;tag=74766f41
To: sip:test_local@localhost
Call-ID: 1953918785-jEV4NMQqubUqMp+WYRx65w(a)public.gmane.org
CSeq: 1 OPTIONS
Contact: sip:sipsak-jEV4NMQqubUqMp+WYRx65w@public.gmane.org:60518
Content-Length: 0
Max-Forwards: 70
User-Agent: sipsak 0.9.6
Accept: text/plain
kamailio is respoding with :
SIP/2.0 404 Not Found
After digging in kamailio log :
2(17356) DEBUG: registrar [lookup.c:158]: lookup(): 'test_local' Not
found in usrloc --
The code reads :
155 ul.lock_udomain(_d, &aor);
156 res = ul.get_urecord(_d, &aor, &r);
157 if (res > 0) {
158 LM_DBG("'%.*s' Not found in usrloc\n",
aor.len, ZSW( aor.s));
So it seems i'm doing something very wrong.
Although in my config :
106 #!define WITH_MYSQL
107 #!define WITH_AUTH
108 #!define WITH_USERLOCDB
After going through the RFC3261's section 11.2,
It seems this OPTION message should work and generate a 200(OK).
What it is i'm doing wrong?
--
-Cheers
-Arif