Hello all,
Just wondering if anyone know any tutorial on setting up HA+DRBD solution for kamailio.
Especially creating partitions, DRBD devices and mount points.
Thanks in advance,
-Sid
"May the light be with you." ______________________________________________
Siddhardha Garige
www.luminepixels.com
Hi all..
I have solve the problem .
Cheers,
vivi
From: vivi [mailto:vivi.hilton@gmail.com]
Sent: Saturday, June 06, 2009 11:29 PM
To: 'users(a)lists.kamailio.org'
Subject: /usr/local/sbin/kamdbctl create error
Hi all
I have specify the wanted db type (DBENGINE=MYSQL) in the
/usr/local/etc/kamailio/kamctlrc
then I using "/usr/local/sbin/kamdbctl create" to create MySQL database,
but I got this error:
ERROR: database engine not specified, please setup one in the config script
root@acer:/usr/local/src/kamailio-1.5.0/sip-server# vim
/usr/local/etc/kamailio/kamctlrc
root@acer:/usr/local/src/kamailio-1.5.0/sip-server# /usr/local/sbin/kamdbctl
create
MySQL password for root:
INFO: test server charset
ERROR 2002 (HY000): Can't connect to local MySQL server through socket
'/var/run/mysqld/mysqld.sock' (2)
ERROR 2002 (HY000): Can't connect to local MySQL server through socket
'/var/run/mysqld/mysqld.sock' (2)
Usage: grep [OPTION]... PATTERN [FILE]...
Try `grep --help' for more information.
/usr/local/lib/kamailio/kamctl/kamdbctl.mysql: line 112: [: =: unary
operator expected
INFO: creating database openser ...
ERROR 2002 (HY000): Can't connect to local MySQL server through socket
'/var/run/mysqld/mysqld.sock' (2)
ERROR: Creating core database and grant privileges failed!
Cheers,
vivi
Hello all,
I recently add a classical Audiocodes Mediant 2000 with 2x 8E1, the purpose
is to have several interconnections with PSTN.
I configured it like this :
Audiocodes registers as a gateway to the Kamailio, using a dedicated port
(5062).
Registration seems to be OK, and the pstn gw uses OPTIONS method to ping the
proxy.
I can attack the Audiocodes with a SIP phone behind Kamailio, no pbm.
But the audiocodes returns some errors about SIP headers sent by Kamailio :
( sip_stack)(44732 ) AcSIPParser: Problem in SIP Message Headers [Time:
12:30:26]
( sip_stack)(44733 ) !! [ERROR] AcSIPParser: Parse Error. Unexpected symbol
'0' in scheme. ALPHA expected
Here you have the example of an INVITE from a SIP phone to the PSTN :
** audiocodes debug **
4d:12h:30m:26s ( lgr_flow)(44730 ) ---- Incoming SIP Message from
77.246.81.132:5060 ----
INVITE sip:0323719001@77.246.81.136:5062;transport=udp SIP/2.0
Record-Route: <sip:77.246.81.132;lr=on;ftag=71078b346a20fb3eo0;nat=yes>
Via: SIP/2.0/UDP 77.246.81.132;branch=z9hG4bKdace.1ab1d59.0
Via: SIP/2.0/UDP
192.168.0.113:5060;rport=15170;received=77.246.81.162;branch=z9hG4bK-b432f96
From: "Sam" <sip:0123451010@sip.720.fr
<sip%3A0123451010(a)sip.720.fr>>;tag=71078b346a20fb3eo0
To: <sip:0323719001@sip.720.fr <sip%3A0323719001(a)sip.720.fr>>
Call-ID: 944d8aec-27503ee6(a)192.168.0.113
CSeq: 102 INVITE
Max-Forwards: 49
Contact: "Sam" <sip:0123451010@77.246.81.162:15170>
Expires: 240
User-Agent: Linksys/SPA941-5.1.8
Content-Length: 281
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER
Supported: 100rel, replaces
Content-Type: application/sdp
P-Asserted-Identity: <0123451010>
Remote-Party-ID: <0123451010>;party=caller;privacy=none;screen=yes
v=0
o=- 26933860 26933860 IN IP4 192.168.0.113
s=-
c=IN IP4 77.246.81.133
t=0 0
m=audio 35038 RTP/AVP 18 0 8 101
a=rtpmap:18 G729a/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
a=nortpproxy:yes
( sip_stack)(44732 ) AcSIPParser: Problem in SIP Message Headers [Time:
12:30:26]
( sip_stack)(44733 ) !! [ERROR] AcSIPParser: Parse Error. Unexpected symbol
'0' in scheme. ALPHA expected
( sip_stack)(44734 ) !! [ERROR] Message type: INVITE [Time: 12:30:26]
( sip_stack)(44735 ) !! [ERROR] Source header: [Time: 12:30:26]
( sip_stack)(44736 ) !! [ERROR] Line: 17. Column: 23 [Time: 12:30:26]
The outgoing INVITE from Kamailio is exactly the same received by the
AudioCodes.
When I searched over Google, I just found 2 answers about Asterisk /
Audiocodes unsolved problem, but no more informations.
I supposed that the problem is as indicated : " s=- " where source is empty
in place of "NULL" / "0" or something like this ...
Someone can confirm or already met the problem ?
Many thanks all :)
.Sam.
high.all!
i'm wondering if there is any support of uaCSTA in openser (planned)?
i'm just working on the integration of asterisk (*) environment to OCS 2007
environment, having openSER in the middle (mainly for TCP/UDP translation
and smoothing out the protocol deficienes on both sides). in this setup the
* having the openSER in front is talking to the OCS (and vice versa) via the
OCS mediation server, which is moreorless sending standard SIP messages,
which enables normal softphone (integration to *) of the office
communicator. this configuration is already working...
now i'm planning to go for the CTI integration, where there is no OCS
mediation server in between OCS and openSER, doing the translation of
SIP/CSTA to SIP. i'm thinking about using openSER for this task, that's why
i'm looking for a CSTA module or perl programm, which is capable of this
functionality.
afaik for the CTI communication there isn't the full complexity of CSTA
needed, just a subset mainly for call setup and call clearing.
anyone having experience on this topic?
thx & cheers
-hugo
Great Ideas for Small Devices
Hugo Koblmueller
Senior Staff Engineer Software Development COMNEON electronic
technology GmbH & Co. OHG
Freistaedter Strasse 400
4040 Linz
Austria
hugo.koblmueller(a)comneon.com
tel:
fax:
mobile:
Skype ID: +43 (5) 1777 - 15730
<http://www.plaxo.com/click_to_call?lang=en&src=jj_signature&To=%2B43+%285%2
9+1777+%2D+15730&Email=hugo(a)koblmueller.com>
+43 (5) 1777 - 15810
+43 (676) 82051280
<http://www.plaxo.com/click_to_call?lang=en&src=jj_signature&To=%2B43+%28676
%29+82051280&Email=hugo(a)koblmueller.com>
drhookson
Want to always have my latest info?
<https://www.plaxo.com/add_me?u=21475050628&src=client_sig_212_1_banner_join
&invite=1&lang=en> Want a signature like
<http://www.plaxo.com/signature?src=client_sig_212_1_banner_sig&lang=en>
this?
Hi
Using Kamailio 1.5 we periodically get the following errors:
Jul 1 09:54:41 localhost /usr/local/sbin/openser[3836]:
ERROR:core:tcp_blocking_connect: poll error: flags 18
Jul 1 09:54:41 localhost /usr/local/sbin/openser[3836]:
ERROR:core:tcp_blocking_connect: failed to retrieve SO_ERROR (110)
Connection timed out
Jul 1 09:54:41 localhost /usr/local/sbin/openser[3836]:
ERROR:core:tcpconn_connect: tcp_blocking_connect failed
Jul 1 09:54:41 localhost /usr/local/sbin/openser[3836]:
ERROR:core:tcp_send: connect failed
Jul 1 09:54:41 localhost /usr/local/sbin/openser[3836]: ERROR:tm:msg_send:
tcp_send failed
They are several emails in this mailing list explaining we should play with
tcp parameters. I will try to change my configuration and see if we can
solve this issue.
But when this kind of error happen we observe that the kamailio processes
are using 100% of CPU (top).
Our client is using SIP over TCP only (no UDP).
Do you have any hint on how to configure Kamailio to improve TCP support ?
Is it normal to kamailio processes to overload CPU ?
Regards,
Pascal
Hello,
I'm seeing an increasing amount of spurious timeouts on my proxies.
Even though the proxy receives a "100 Trying" response immediately
after relaying the request, tm will send retransmits and times out
eventually. Below is a commented log snippet. The timeout occurs on
the proxy named 'balancer1'. Both are on the same host.
I verified that the 100 Trying response is received in a t_on_reply
block, so it is recognized by tm as belonging to a transaction.
When needed, I'll provide a debug log when it occurs again.
Meanwhile, any idea what is causing this?
This is on kamailio 1.5+svn5768
fr_timer = 10
fr_timer_avp = 2
# Initial request received on balancer
May 15 12:32:01 balancer1[24170]: [2 INVITE] - New request. <sip:uac@domain> => <sip:uas@domain>
May 15 12:32:01 balancer1[24170]: [2 INVITE] Relaying
# Initial request received on proxy
May 15 12:32:01 proxy1[23803]: [2 INVITE] - New request. <sip:uac@domain> => <sip:uas@domain>
# Balancer receives 100 response
May 15 12:32:01 balancer1[24166]: [2 INVITE] - Reply. 100 Trying
# Proxy relays request
May 15 12:32:01 proxy1[23803]: [2 INVITE] Relaying
# Proxy receives retransmission from balancer ???
May 15 12:32:02 proxy1[23804]: [2 INVITE] - New request. <sip:uac@domain> => <sip:uas@domain>
# Balancer times out ???
May 15 12:32:02 balancer1[24171]: [2 INVITE] Failure (Status: 408)
May 15 12:32:02 balancer1[24171]: [2 INVITE] No more gateways
May 15 12:32:02 proxy1[23804]: [2 CANCEL] - New request. <sip:uac@domain> => <sip:uas@domain>
--
Greetings,
Alex Hermann
Hi all
I have specify the wanted db type (DBENGINE=MYSQL) in the
/usr/local/etc/kamailio/kamctlrc
then I using "/usr/local/sbin/kamdbctl create" to create MySQL database,
but I got this error:
ERROR: database engine not specified, please setup one in the config script
root@acer:/usr/local/src/kamailio-1.5.0/sip-server# vim
/usr/local/etc/kamailio/kamctlrc
root@acer:/usr/local/src/kamailio-1.5.0/sip-server# /usr/local/sbin/kamdbctl
create
MySQL password for root:
INFO: test server charset
ERROR 2002 (HY000): Can't connect to local MySQL server through socket
'/var/run/mysqld/mysqld.sock' (2)
ERROR 2002 (HY000): Can't connect to local MySQL server through socket
'/var/run/mysqld/mysqld.sock' (2)
Usage: grep [OPTION]... PATTERN [FILE]...
Try `grep --help' for more information.
/usr/local/lib/kamailio/kamctl/kamdbctl.mysql: line 112: [: =: unary
operator expected
INFO: creating database openser ...
ERROR 2002 (HY000): Can't connect to local MySQL server through socket
'/var/run/mysqld/mysqld.sock' (2)
ERROR: Creating core database and grant privileges failed!
Cheers,
vivi
if i have understood correctly, if uac_replace_from() is called
in "auto" mode, encoded from uri is added into an extra param of r-r
header when the request is forwarded towards uas.
does the module check that reply to the request from uas really includes
the added parameter in its r-r header or is correct operation of
in-dialog requests at the mercy of the uas?
-- juha
Hi,
I searched around the web to load balance asterisk servers and found
Kamailio for possible solution. Let's say I have two identical asterisk
servers with same dialplan and configuration and I want both servers look
like they have same IP address from clients. Based on my understanding, we
need 1 Kamailio as load balancer and 2 Asterisk servers as the real servers.
Let's say the setup is :
Kamailio load balancer 192.168.2.1
Asterisk Server #1 192.168.2.2
Asterisk Server #2 192.168.2.3
My question is, X-Lite softphone Configuration should be set to domain
192.168.2.1, right?
I also want to know the step by step configuration to set kamailio as load
balancer. I have not used Kamailio before. However after searching the
documentation, the step (based on my understanding) is somewhat like this :
1. Install Kamailio. I will use the step by step here :
http://kamailio.org/dokuwiki/doku.php/install:kamailio-1.5.x-from-svn
2. Then, using dipatcher module, I will configure it using this guide here :
http://kamailio.org/dokuwiki/doku.php/asterisk:load-balancing-and-ha
Then modify the dispatcher.list file to match the IP address of my asterisk
servers :
*1 sip:192.168.2.2:5060
1 sip:192.168.2.3:5060*
Am I missing some steps?
Do I also need to configure dialplan or any other file at Kamailio load
balancer? Or those two steps basically done it all for simple load balancing
configuration? Thanks for your responses.
Best Regard,
Kurt Weasel.
here is correct reqwest and reqwest via kamailio
is there any way to rewrite To: field in kamailio
because now it nto work
correct reqwest(asterisk)
RECEIVING FROM: 69.70.173.195:5060
CANCEL sip:34249@193.110.78.12:8484 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.180:5060;branch=z9hG4bK4c97bd4a
From: "Unknown"
<sip:Unknown@voip1.bravotelecom.com<sip%3AUnknown(a)voip1.bravotelecom.com>
>;tag=as366a0ba0
To: <sip:34249@193.110.78.12:8484>
Call-ID: 322db75c71be4ba45a61c7666b93e6be(a)voip1.bravotelecom.com
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
incorrect reqwest(multi-homed kamailio, asterisk originate to internal eth)
RECEIVING FROM: 69.70.173.195:5060
CANCEL sip:34249@193.110.78.12:8484 SIP/2.0
Via: SIP/2.0/UDP 69.70.173.195;branch=z9hG4bK1b1b.3c42cea6.0
From: "Unknown" <sip:Unknown@69.70.173.195 <sip%3AUnknown(a)69.70.173.195>
>;tag=as6075df60
Call-ID: 4d784fa972c4e9c9795032417f46cb26(a)voip1.bravotelecom.com
To: <sip:34249@192.168.2.170 <sip%3A34249(a)192.168.2.170>>
CSeq: 102 CANCEL
Max-Forwards: 70
asterisk
Content-Length: 0
--
Merkulov Alexander