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
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?
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
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
2009/3/23 Alexandr Dubovikov <shurik(a)start4.info>:
> On Mon, Mar 23, 2009 at 11:15:52AM +0100, Iñaki Baz Castillo wrote:
>> 2009/4/1 Alexandr Dubovikov <shurik(a)start4.info>:
>> > On Fri, Mar 20, 2009 at 10:21:20PM +0100, Andreas Heise wrote:
>> >> Hello Iñaki,
>
> Hi Inaki,
>
>> >
>> > Hi all,
>> >>
>> >> you should ask Alexandr he has introdused this feature with rev5452,
>> >> but I'm not sure if he is on the lists all the time, so I'll forward your
>> >> question
>> >> to him...
>> >
>> > sorry, I am currently offline and couldn't answer directly to the list.
>> >
>> > anyway. I agreed, 180 seconds it's too big ping interval, but it couldn't be
>> > also less 32 seconds.
>> >
>> >
>> > http://www.ietf.org/rfc/rfc3261.txt
>> >
>> > 17.2.2 Non-INVITE Server Transaction
>> >
>> >
>> > T1 = 500 ms.
>> >
>> >
>> > Timer J 64*T1 for UDP Section 17.2.2 Wait time for
>> > 0s for TCP/SCTP non-INVITE request
>> >
>> > so, the timer for "completed" is 32 seconds and couldn't be less.
>> >
>> >
>> > of course, you can use less value, but on own risk :-)
>> > RFC 3261 are not recommendet it :)
>>
>> Thanks for pointing it out.
>> However, if a gw takes so long time (~32 seconds) to respond I prefer
>> to consider it as "offline" :)
>
> don't forget to disable ping for this gateway, because finaly you will have
> a DDOS effect of the "offline" gw. :-)
Well, If the host is already offline then it doesn't matter to kill it more XD
Thanks.
--
Iñaki Baz Castillo
<ibc(a)aliax.net>
Hello,
I'm using the set_dlg_profile function from the dialog module (kamailio 1.4).
I use it to control limit of simultaneous calls to clients.
I'm noticing that after this function is called, even if the call
terminates immediately, it takes some 3 to 4 seconds for the profile
to be cleared. Is there any reason for this?
I can see even if I call unset_dlg_profile on failure_route, the
profile will take that time span to be cleared.
regards,
mayama
hi
To install another instance of kamailio listening in another IP, does one need to do something in particular aside from install it in another directory?
Also, when it is time to run the script to create the mysql tables, how can I tell the script to create a db named "openser2" to avoid deleting the existing one named by default "openser" ? Just by changing the line DBNAME=openser in kamctlrc to DBNAME=openser2?
thank you
jp
I am having an issue while setting up Kamailio. I using the carrierroute modules to process calls between carriers. The configuration works great. However I have and issue. On the SIP messages We are forcing all the media through an RTP proxy, however my issue has to do with the Contact field on the Message Header. While I am able to use force_rtp_proxy to replace the contact and connection fields on the SDP part of the message, I can seem to modify the Contact on the Header part.
I tried using subst to change the Contact:, and while I was partly successful, it seems the ACKs going from one carrier to another are getting messed up and throwing Kamailio into a loop. We are trying to avoid providing Contact: information to our carriers. Basically keeping them from know about each other. Any ideas/help will be Greatly appreciated.
Hi,
I would like to run rtpproxy 1.2.0 on a server with Quad-core dual Xeon processor and load balance the load across the 4 processors.
Do you know how do I enable this feature pls. ( I am not an expert in this field so keep it simple pls. thanks)
Thanks in advance
Regards,
Tony
