9 dec 2011 kl. 18:54 skrev Spencer Thomason:
> Hello all,
> Is it possible to configure Kamailio to reply to a request with a 503 if it
> cannot connect to a necessary database for the operation?
>
> I.e. Kamailio cannot connect to the database, when a REGISTER comes in and
> userloc can't fi
12 dec 2011 kl. 10:33 skrev Daniel-Constantin Mierla:
> Hello,
>
> On 12/10/11 11:36 AM, Olle E. Johansson wrote:
>> 9 dec 2011 kl. 18:54 skrev Spencer Thomason:
>>
>>> Hello all,
>>> Is it possible to configure Kamailio to reply to a request with a 503 i
12 dec 2011 kl. 10:45 skrev Daniel-Constantin Mierla:
> Hello,
>
> On 12/9/11 9:04 PM, Gautam Batra wrote:
>> Hello,
>>
>> I have a kamailio sip proxy server with freeswitch acting as SBC. I want to
>> redirect the call to freeswitch when hold is pressed so that i can play
>> music on hold. I
12 dec 2011 kl. 11:26 skrev Daniel-Constantin Mierla:
> Hello,
>
> On 12/12/11 10:49 AM, Olle E. Johansson wrote:
>> 12 dec 2011 kl. 10:33 skrev Daniel-Constantin Mierla:
>>
>>> Hello,
>>>
>>> On 12/10/11 11:36 AM, Olle E. Johansson wrote:
20 dec 2011 kl. 05:57 skrev sunny_day:
> There should be a syslog.conf file under the kamailio directory.
> But I haven't.
>
> I only want to see the contents generated by "xlog"
Syslog is a daemon in your operating system. The configuration file is propably
in the /etc
directory. Try runnin
reinvite in the
call.
/O
>
> Gautam
>
> On Mon, Dec 12, 2011 at 4:50 AM, Olle E. Johansson wrote:
>
> 12 dec 2011 kl. 10:45 skrev Daniel-Constantin Mierla:
>
> > Hello,
> >
> > On 12/9/11 9:04 PM, Gautam Batra wrote:
> >> Hello,
> >>
>
29 jan 2012 kl. 13:11 skrev Daniel Pocock:
>
>
> I found that my TLS client was not happy because my server cert is
> signed by an intermediate root.
>
> A quick search in Google found other people mentioning the same problem,
> but no solution or documentation.
>
> I've had a quick look in t
29 jan 2012 kl. 22:27 skrev Daniel Pocock:
>
>
> On 29/01/12 21:47, Iñaki Baz Castillo wrote:
>> 2012/1/29 Daniel Pocock :
>>> It's a little bit different in Apache, where the user specifies a file
>>> containing intermediate certs - many of the CAs give instructions for
>>> adding that file in
7 feb 2012 kl. 10:21 skrev Henning Westerholt:
> On Monday 06 February 2012, Serhat AKCA wrote:
>> I want to run Kamailio under FreeBSD. I compiled with the makefiles under
>> freebsd packages. But It gives alotof errors. I want to use 3.2.2 version
>> but I cannot compile. Could you help me To c
10 feb 2012 kl. 15:16 skrev Daniel-Constantin Mierla:
> Hello,
>
> On 2/9/12 3:51 PM, Yufei Tao wrote:
>> Hi
>>
>> When clients register to Kamailio over TCP/TLS, if I set
>> tcp_connection_lifetime to be quite small, like 30 seconds, and let
>> clients send refreshes every 20 seconds, for exam
15 feb 2012 kl. 18:29 skrev Stoyan Mihaylov:
> I did - registration is purely in Kamailio.
> In Asterisk - I created sip account for Kamailio based on IP address without
> username and password.
> This way - all calls from Kamailio go to Asterisk without problems.
> In Kamailio I allowed calls f
Friends,
I'm having issues with the xpath support. If I run Daniel's example in the
XMLOPS, xpath works as documented. But if I take the
body of a PUBLISH from the and run xpath, I don't get expected results. Now,
I'm no XPATH guru so I may be totally off the markup here...
Here's the test scrip
27 feb 2012 kl. 21:07 skrev Daniel-Constantin Mierla:
> Hello,
>
> an xpath invalid expression should be printed when the xpath expression is
> incorrect -- I cannot say what is wrong, not being an xpath expert by hart.
>
> However, there is a mismatching between the xml standards and SIP/SIMP
27 feb 2012 kl. 21:44 skrev Daniel-Constantin Mierla:
>
>
> On 2/27/12 9:28 PM, Olle E. Johansson wrote:
>> 27 feb 2012 kl. 21:07 skrev Daniel-Constantin Mierla:
>>
>>> Hello,
>>>
>>> an xpath invalid expression should be printed when the xp
Just to make sure we're in the flow, I've started a Facebook page for Kamailio.
http://www.facebook.com/kamailio
Feel free to like it if you're a Facebook user. There's also a Google+ page.
If you have news to share, you're welcome to add it on these pages.
/O
__
I've set
tos=IPTOS_RELIABILITY
in my configuration, which follows
https://www.kamailio.org/wiki/cookbooks/3.2.x/core#tos which has the following
examples:
tos=IPTOS_LOWDELAY
tos=0x10
tos=IPTOS_RELIABILITY
but I get a configuration error:
0(45338) : [cfg.y:3501]: parse error in config
rlin, Germany
>
> http://www.asipto.com/index.php/kamailio-advanced-training/
> ___
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users@lists.sip-router.org
&g
2 apr 2012 kl. 10:10 skrev Daniel-Constantin Mierla:
> Hello,
>
> based on last IRC devel meeting, the next major release was planned to happen
> before the summer holidays.
>
> I am considering freezing the development for v3.3.0 by April 23, test for
> 4-6 months and release by end of May/b
a - http://www.linkedin.com/in/miconda
>
>
> ___
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users@lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
---
* Ol
24 apr 2012 kl. 16:04 skrev Daniel-Constantin Mierla:
> Hello,
>
> On 4/24/12 3:55 PM, Olle E. Johansson wrote:
>> 24 apr 2012 kl. 14:27 skrev Daniel-Constantin Mierla:
>>
>>> Hello,
>>>
>>> thanks, I found that as well, but I didn't wante
Hello!
I will be running an Asterisk SIP Masterclass - the last one - in Barcelona in
June. During this week, I will organize a dinner for everyone working with or
interested in Asterisk, Kamailio and other Open Source platforms for realtime
communication. It's Wednesday, June 13th somewhere in
xpress Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users@lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
---
* Olle E Johansson - o...@edvina.net
* Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden
_
I got this question today that I can't find the answer on:
How many values are there in an avp?
If I have pushed X values into the AVP - is there a function that counts them
for me?
/O
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mai
I know that you can set the TOS for packets sent from Kamailio in the config
section.
I got the question if you can read the TOS in the IP packet of an incoming
response or reply in the config scripts?
And can you set it per message/transaction?
/O
__
15 jun 2012 kl. 05:56 skrev Jakson Kalsson:
>
>
> Hi all, I have a project for the 3G related, AMR and AMR-WB support.
>
> I'm using the client develop suite from the PortSIP(http://www.portsip.com),
> as their said
> support the AMR, AMR-WB with RFC4867.
>
> Now I have to setup a SIP server
15 jun 2012 kl. 08:44 skrev Daniel-Constantin Mierla:
> Hello,
>
> I am introducing Vicente as a new registered developer - he submitted lately
> very useful patches to ndb_redis module (e.g., array support in replies,
> redis free function for config), new ones being on the pipe -- he can pre
27 jun 2012 kl. 10:01 skrev Daniel-Constantin Mierla:
> the patch was backported to branch 3.2, it is recommended you use the latest
> version of that branch, as there might be other fixes that can hit you at
> some point. It is safe to do it, you don't need to change anything in
> database or
11 jul 2012 kl. 09:00 skrev Daniel-Constantin Mierla:
> Hello,
>
> On 7/11/12 4:23 AM, mike wrote:
>> dear kamilio developers :
>>
>> i read the kamilio 3.3.0' s new features , it say it support outbound , and
>> i have place to handle reg-id and sip.instance . so , i wonder if the
>> kamilio
le. A temporary GRUU is
only valid for as long as a device registers with the same Call-ID.
Please give me pointers to what I have missed!
Thanks,
/O
---
* Olle E Johansson - o...@edvina.net
* Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden
__
11 jul 2012 kl. 15:16 skrev Daniel-Constantin Mierla:
>
> On 7/11/12 9:28 AM, Olle E. Johansson wrote:
>> 11 jul 2012 kl. 09:00 skrev Daniel-Constantin Mierla:
>>
>>> Hello,
>>>
>>> On 7/11/12 4:23 AM, mike wrote:
>>>> dear kamili
11 jul 2012 kl. 15:28 skrev Daniel-Constantin Mierla:
> Hello,
>
> On 7/11/12 3:18 PM, Olle E. Johansson wrote:
>> Hey,
>> Been trying to understand the GRUU support in Kamailio. I see that the
>> registrar saves sip.instance - but I don't see any functions to
11 jul 2012 kl. 15:42 skrev Daniel-Constantin Mierla:
> Hello,
>
> On 7/11/12 3:36 PM, Olle E. Johansson wrote:
>> 11 jul 2012 kl. 15:28 skrev Daniel-Constantin Mierla:
>>
>>> Hello,
>>>
>>> On 7/11/12 3:18 PM, Olle E. Johansson wrote:
>>&
11 jul 2012 kl. 15:44 skrev Olle E. Johansson:
>
> 11 jul 2012 kl. 15:42 skrev Daniel-Constantin Mierla:
>
>> Hello,
>>
>> On 7/11/12 3:36 PM, Olle E. Johansson wrote:
>>> 11 jul 2012 kl. 15:28 skrev Daniel-Constantin Mierla:
>>>
>>>>
11 jul 2012 kl. 17:57 skrev Iñaki Baz Castillo:
> 2012/7/11 Olle E. Johansson :
>> I see that the gruu_enabled() parameter enables gruu if there's an instance
>> ID in the contact.
>>
>> The requirement is also that there has to be a supported: gruu
>>
&
11 jul 2012 kl. 18:06 skrev Daniel-Constantin Mierla:
> On 7/11/12 5:58 PM, Olle E. Johansson wrote:
>> 11 jul 2012 kl. 17:57 skrev Iñaki Baz Castillo:
>>
>>> 2012/7/11 Olle E. Johansson :
>>>> I see that the gruu_enabled() parameter enables gruu if there&#
20 aug 2012 kl. 10:06 skrev Daniel-Constantin Mierla:
> Hello,
>
> over the weekend I committed the code that allows to remove contacts from
> location table if the device does not respond to several attempts of SIP nat
> keepalives (usually OPTIONS requests).
>
> The feature works only for S
22 aug 2012 kl. 18:01 skrev Carlos Cruz:
> I'm just starting to evaluate Kamailio to be used as a front end to a couple
> of Asterisk Servers.
>
> Currently I have an Adobe Flex application that interfaces with Asterisk via
> AMI using a socket connection. I searched on how to interface with
I haven't heard anyone using Asterisk in large production systems in other
virtualization than OpenVZ. Asterisk depends a lot on timers and these aren't
reliable enough in Vmware and Xen when putting load on the media server.
I would love to hear about a successful implementation :-) on other pl
x27;s a matter of personal preferences. Good to hear that you have it
working in Xen!
/O
>
>
> 2012/8/28 Olle E. Johansson
> I haven't heard anyone using Asterisk in large production systems in other
> virtualization than OpenVZ. Asterisk depends a lot on timers and these ar
Good discussion!
In most, but not all, cases it's a political/business decision outside of the
scope of the technichal specifications. A commercial SBC delivers a cloud of
magic dust that makes some people feel better and more secure. I have audited
several SBC installations that are totally in
31 aug 2012 kl. 11:04 skrev Tayeb Meftah :
> Hello
> Sbc also ofer a propritary way of failover for itself
> If node1 die node2 will replace it
That's been done both with Asterisk and Kamailio (and propably SEMS) for a very
long time.
/O
smime.p7s
Description: S/MIME cryptographic signature
_
8 okt 2012 kl. 16:33 skrev Daniel-Constantin Mierla :
> Hello,
>
> some bits were left with the old project name OpenSER, not to disturb too
> much at that time and see where everything goes.
>
> Other voices expressed same idea in the past, now everything is stable and
> development goes smo
11 okt 2012 kl. 13:57 skrev SamyGo :
> :) "Soon..." But Not Today.
>
> Not everyone can afford the Gateways. Thanks for the replies. I was hoping
> maybe someone else be thinking of freeing the kamailio from Asterisks or
> Freeswitchs when it comes to interconnecting with PSTN.
>
Why? We ha
12 okt 2012 kl. 09:57 skrev Christophe ROY :
> I have created a SRV record for sip.domain.tld pointing to our kamailio
> server, the domain by itself resolves to our DNS IP.
If you check the SRV record standards - especially RFC 3263 - you see that you
need
_sip._udp.domain.tld
_sip._tcp.doma
15 okt 2012 kl. 13:24 skrev Peter Lemenkov :
> Hello.
>
> 2012/10/15 Kamal Palei :
>> Hi All
>> I am planning to enhance RTP proxy to support TLS and DTLS.
>> We have some requirements where we need to send RTP packets either over TLS
>> or over DTLS.
>
> Shouldn't it be better to rely on SRTP/
22 okt 2012 kl. 19:59 skrev Juha Heinanen :
> Daniel-Constantin Mierla writes:
>
>> For safety, i would use 48, to allow zero termination
>
> why 48 when max length of ipv6 addr is 39 chars? did you mean 40?
It's hard to judge the max length of IPv6, since there are many notations. With
IPv4
22 okt 2012 kl. 20:24 skrev Daniel-Constantin Mierla :
> Hello,
>
> that message is printed by the mechanism that tries to discover local
> hostnames to add them to aliases list.
>
> To turn that off, use '-a no' in command line or 'auto_aliases=no' in
> configuration file.
Which is a command
22 okt 2012 kl. 20:24 skrev Daniel-Constantin Mierla :
> Hello,
>
> that message is printed by the mechanism that tries to discover local
> hostnames to add them to aliases list.
>
> To turn that off, use '-a no' in command line or 'auto_aliases=no' in
> configuration file.
http://www.kamaili
25 okt 2012 kl. 13:20 skrev Andreas Granig :
> Hi,
>
> Sorry for hijacking this thread, but I've a similar but different
> question which bugs me since a while :)
>
> Is there a way in kamailio to statelessy forward a request without
> putting its own Via header into the message? Consider an ex
The marketing arm of the Kamailio project has without further discussion opened
up a Twitter account for Kamailio, to support our web site news, G+ and
Facebook pages. Feel free to contribute with case stories and news we can
publish on the web site and follow up in the social media flow.
The t
26 okt 2012 kl. 19:07 skrev Peter Dunkley :
> Hi,
>
> External anchors are not currently supported by the Kamailio RLS module. The
> presence of these do not explain all of the symptoms you are seeing - but you
> are not going to be able to get your resource-lists working properly if they
>
end
calls with a BYE based on a timer.
> Is there a cleanup process or a parameter to set in the config to
> avoid/delete the dead transactions?
The question here is if you mean calls without a bye or really mean a "dead"
SIP transaction?
/O
--
* Olle E. Johansson - o...@e
Hi!
I am looking for a way to add addresses to the IP blacklist in Kamailio at
startup. Is that possible?
I know I can do it with sercmd, but I fail to find a documented way to do it in
the configuration... Must be missing something simple.
Thank you for your help!
/O
31 okt 2012 kl. 06:39 skrev Jeremy Ardley :
> Hi,
>
> I'm after advice.
>
> I'm developing a script to provide a degree of PBX functionality for a
> small to medium office using Kamailio & SEMS.
>
> This includes mapping internal extensions to external PSTN numbers via
> the SIP trunk provider
erver
> config files.
> Kindly advise for required changes in Kamailio.
As far as I understand, Sylkserver is just a SIP server. Forward calls from
kamailio to sylkserver using any of the forwarding commands.
/O
--
* Olle E. Johansson - o...@edvina.net
* Kamailio & SIP Masterclass Miam
Friends,
The 2600Hz project that builds an API-driven telephony service have selected to
switch to Kamailio as a SIP proxy for their SBC. They blog about the switch,
comparing Kamailio with their previous choice and coming to a conclusion:
"To our minds, a more active development community mean
g.
Look for has_totag()
/O
--
* Olle E. Johansson - o...@edvina.net
* Kamailio & SIP Masterclass Miami FL December 2012
* http://edvina.net/training/
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@list
u have to check AFTER the block that
checks the existence of the to-tag.
Please check the sample configuration you got with Kamailio to learn more!
Regards,
/Olle
--
* Olle E. Johansson - o...@edvina.net
* Kamailio & SIP Masterclass Miami FL December 2012
* http://edvina.net/training/
31 okt 2012 kl. 14:19 skrev Alex Balashov :
> Inside if(loose_route()) will do.
The question was "where is the right place
>
>> to check the new INVITE?"
Proper answer would be after this block... :-)
/O
>
> if(has_totag()) {
> if(loose_route()) {
> if(is_method("INVITE")) {
>
Friends,
Dispatcher has a mode 8 - "use first destination" which makes sense for a
textfile driven dispatcher configuration.
Now if I use a database - will dispatcher first sort on priority, then take the
first?
It seems so from the documentation of the text file:
priority: sets the pri
No.
I am sure there are ways to use SQL views or some other magic to make it
happen...
If the databases align, you can patch the source (yes, you have full right to
the source)
so that one of them accepts the version number of the other. It requires some
testing
and verification on your end.
/
statelessly.
Cheers,
/O
>
>
> Regards,
>
> VJ++
>
> Message: 7
> Date: Wed, 31 Oct 2012 08:26:50 +0100
> From: "Olle E. Johansson"
> Subject: Re: [SR-Users] Kamailio and sylkserver integration
> To: "SIP Router - Kamailio \(OpenSER\) and SIP Expres
14 nov 2012 kl. 13:23 skrev Daniel-Constantin Mierla :
> Hello,
>
> I am thinking of having our next IRC devel meeting soon, to plan the next
> major release and review current stable releases and the environment around
> the project (e.g., if you can add anything else to the project to make l
simplest way would be to use a tool like SIPSAK to send
the MESSAGE to kamailio and let kamailio route it to the user.
Cheers,
/O
--
* Olle E. Johansson - o...@edvina.net
* Kamailio & SIP Masterclass Miami FL December 2012
* http://edvina.net/training/
___
15 nov 2012 kl. 11:58 skrev Christophe ROY :
> Hi everyone
>
> I'm trying to integrate Asterisk with Kamailio for voicemail.
> I tried to follow this tutorial:
> http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb
> BUT:
>
> - I had to adapt it because I use LDAP authen
if(is_method("REGISTER"))
> {
> t_relay_to("tls:115.114.48.75:443");
> exit();
> }
>
> if(is_method("INVITE|BYE|CANCEL|SUBSCRIBE|REFER|NOTIFY"))
> {
> xdbg("incoming req
19 nov 2012 kl. 15:06 skrev Andreas Granig :
> Hi David,
>
> On 11/19/2012 02:54 PM, David J wrote:
>> Is the database shared? If so maybe when they authenticate add a secure
>> token to the header that the second proxy can use for auth?
>
> No, the DBs are explicitely NOT shared in this scenar
19 nov 2012 kl. 15:40 skrev Carsten Bock :
> Maybe we should merge the docs from 1.5, the docs are much better:
>
> "Secret phrase used to calculate the nonce value.
>
> The default is to use a random value generated from the random source
> in the core.
>
> If you use multiple servers in your
19 nov 2012 kl. 23:19 skrev Andrew Mortensen :
>
> On Nov 19, 2012, at 4:56 PM, David J wrote:
>
>> This looks really awesome. Thanks for sharing
>
> Thanks, and you're welcome. I've added very simple installation instructions
> to the repo.
Do we have any idea on the licensing for this? Th
19 nov 2012 kl. 19:25 skrev Bolang :
> Hi all,
> is there any standard mechanism to create a temporary user?
> i know i can create it and then delete it after some amount of time.
> but, i'm looking for an established/standard way of doing this.
>
> Ideally, the user will be automatically delete
20 nov 2012 kl. 10:25 skrev Johan Wilfer :
> Hi,
>
> I've done some tests with the UAC module to authenticate to a remote proxy.
> I've based my config on this example:
> http://docs.huihoo.com/opensips/tutorials/uac/ar01s06.html (example 9)
>
> I have found that if I send a call from a aster
3 dec 2012 kl. 10:43 skrev Andreas Granig :
> Hi Klaus,
>
> On 12/03/2012 10:15 AM, Klaus Darilion wrote:
>> The request URI should look like the one which the user enters. E.g. if
>> user enters "sip:12...@example.com" then the request URI should be
>> "sip:12...@example.com" - regardless of th
4 dec 2012 kl. 05:26 skrev Ovidiu Sas :
> Hello all,
>
> For those who like running kamailio on routers and/or other small
> embedded systems, the latest kamailio stable is available for
> download.
> For more info, please check: http://www.nslu2-linux.org/wiki/Optware/HomePage
> For a list of s
Hi!
Daniel has set a code freeze date for the next major release and the Kamailio
community needs to work together to prepare the release.
That means you as well!
There will be a lot of work in the coming weeks to merge code from SIP router
into the Kamailio core to complete the merger process.
!
/O
Vidarebefordrat brev:
> Från: "Olle E. Johansson"
> Ämne: Realtime testers needed - Path header support (Oolong branch)
> Datum: 7 december 2012 11:07:09 CET
> Till: Asterisk Developers Mailing List
> Kopia: "Olle E. Johansson"
>
> Friends,
>
>
11 dec 2012 kl. 01:39 skrev Juha Heinanen :
> Iñaki Baz Castillo writes:
>
>> transport=tls has NEVER been real, no one RFC mentions it.
>
> transport=tls is very real. many sip UAs and proxies support it.
The interesting part was that it was deprecated in the text of RFC 3261, but
never exist
12 dec 2012 kl. 10:18 skrev Grant Bagdasarian :
> Hello,
>
> We have a cluster of three Asterisk machines. Each machine answers an
> incoming call and transfers it to unique remote destination.
> So, Asterisk01 transfers to Destination01, Asterisk02 to Destination02,
> Asterisk03 to Destinati
12 dec 2012 kl. 10:44 skrev Daniel-Constantin Mierla :
> Hello,
>
> On 10/9/12 3:59 PM, Andreas Granig wrote:
>> Hi,
>>
>> I'm playing around with xavp, but there are some things I can't wrap my
>> head around.
>>
>> What basically works is this:
>>
>> $xavp(a=>foo) = 'foo';
>> $xavp(a[0]=>ba
12 dec 2012 kl. 11:11 skrev Daniel-Constantin Mierla :
>
> On 12/12/12 11:08 AM, Olle E. Johansson wrote:
>>
>> 12 dec 2012 kl. 10:44 skrev Daniel-Constantin Mierla :
>>
>>> Hello,
>>>
>>> On 10/9/12 3:59 PM, Andreas Granig wrote:
>>
Hi!
I want to change METHOD on a SIP message from UPDATE to INVITE or REGISTER to
INFO.
Seems like $rm is read-only. Any other known way to change a requests method?
/O
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-us
Evil stuff happening here. Kamailio is a good test-tool.
Now I fail to change Contact: headers. The docs for textops say that remove_hf
can remove
contact and Remove_hf(Contact) returns true - but the old contact is still
there!
/O ;-)
>
> Cheers!
> Uriel
>
>
> On Wed, Dec
19 dec 2012 kl. 19:35 skrev Daniel-Constantin Mierla :
>
> On 12/19/12 7:20 PM, Andreas Granig wrote:
>> Hi Olle,
>>
>> On 12/19/2012 02:43 PM, Olle E. Johansson wrote:
>>> Yeah, I already have Kamailio sending all kinds of crazy stuff while
>>> testin
19 dec 2012 kl. 23:14 skrev Daniel-Constantin Mierla :
>
> On 12/19/12 10:28 PM, Olle E. Johansson wrote:
>> 19 dec 2012 kl. 19:35 skrev Daniel-Constantin Mierla :
>>
>>> On 12/19/12 7:20 PM, Andreas Granig wrote:
>>>> Hi Olle,
>>>>
&
18 dec 2012 kl. 11:44 skrev Daniel Tryba :
> On Tuesday 18 December 2012 09:21:24 Daniel-Constantin Mierla wrote:
>> you have to use it only once -- this is due to how the changes are done
>> to the sip message headers, but also because of adding a special
>> parameter to record-route header. I
20 dec 2012 kl. 15:17 skrev Daniel-Constantin Mierla :
> Hello,
>
> most of the duplicated or ser-specific modules were sorted out at this time,
> next is the situation with the remaining ones.
>
> A) modules that will be renamed using a prefix 'uid_', because they have a
> database schema us
21 dec 2012 kl. 16:55 skrev Daniel-Constantin Mierla :
> Hello,
>
> Kamailio is used a lot in enterprises. Apart of the media processing services
> (e.g., voicemail, audio conferencing), kamailio offers all needed in an
> enterprise, including instant messaging and presence.
We've done quite a
21 dec 2012 kl. 17:10 skrev Ovidiu Sas :
> If you just want to control the debug level externally, take a look at
> the debug parameter:
> http://www.kamailio.org/wiki/cookbooks/3.3.x/core#debug
> It can be controlled via sercmd (kamcmd in future versions).
>
> If you want to play with global fl
uration. Check syslog. There's also a pv
buffer size you can set in the kamailio configuration - check the core cookbook.
Kamailio xlog logs to your syslog daemon.
Guessing a bit here.
/O
>
> Are you aware of maximum length of this xlog message parameter? I cannot find
> it in d
30 dec 2012 kl. 10:39 skrev Techie Sup :
> Hello,
>
> My IP is 192.168.1.25 and I have added left zero padding i,e: 192.168.001.025
> in the configuration file and try to start kamailio, it works.. but along
> with that rtpproxy does not start. Once I revert back it works. My script
> require
2 jan 2013 kl. 10:15 skrev Daniel-Constantin Mierla :
> Hello,
>
> using v4.0.0 for the next major release was proposed and debated at previous
> development IRC meeting. First was about the addition of a new transport
> layer (websockets) along with other important new features.
>
> During t
4 jan 2013 kl. 07:22 skrev Shreyansh Purwar :
> Hello Everyone,
>
> I need a help, I was installing kamailio SIP server according to the
> instruction provided at wiki page
>
> http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.0.x-from-git
>
> but while compiling it(make all) i a
7 jan 2013 kl. 20:46 skrev Glenn O Larsen :
> On Fri, Jan 4, 2013 at 4:18 PM, Henning Westerholt wrote:
>> Am Donnerstag, 3. Januar 2013, 16:14:59 schrieb Glenn O Larsen:
>>>
>>> Is there a way to read modparam config from a database?
>
>> probably the configuration database module - cfg_db is
15 okt 2010 kl. 17.42 skrev JR Richardson:
> On Fri, Oct 15, 2010 at 10:22 AM, Fred Posner wrote:
>> Hey JR...
>>
>> I use this:
>>
>> #! /usr/bin/perl -w
>> use IO::Socket;
>> use POSIX 'strftime';
>>
>> my ($msg,$remotehost,$callid,$socket,$date,$branch,$localip,$dest);
>>
>> $remotehost =
Hello friends,
Is there any way from the outside (like from MI) I can check the existing
TCP/TLS connections?
Can I drop them somehow?
/O
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://
31 dec 2010 kl. 19.00 skrev Daniel-Constantin Mierla:
>
>
> Thanks for your support for the project! I wish you a great 2011 in personal
> life and business, enjoy tonight party!
>
A huge THANK YOU to all the Kamailio Developers. You've done a great job during
2010 and I wish we all can cont
Friends,
We've had an interesting discussion on the Asterisk-dev mailing list about
supporting the ;maddr and ;ttl attributes in the via header when sending
responses. We've agreed that it should be considered harmful and suggest making
it configurable whether to support it in Asterisk.
My ques
7 jan 2011 kl. 10.55 skrev Alex Balashov:
> On 01/07/2011 04:52 AM, Carsten Bock wrote:
>
>> the trick, when using "uac_replace_from" is, that it will
>> "automagically" change subsequest requests. This is required in order
>> to work with most SIP-Endpoints.
>> If you just want to change the "F
15 mar 2011 kl. 13.36 skrev Stefan Sayer:
> We are happy to announce the availability of the
> SIP Express Media Server version 1.4.0.
>
Congratulations! This new version sounds really exiting.
What's the status of SEMS and IPv6?
/O ;-)
___
SIP Ex
15 mar 2011 kl. 14.23 skrev Stefan Sayer:
> Hi Olle,
>
> o Olle E. Johansson on 03/15/2011 01:52 PM:
>>
>> 15 mar 2011 kl. 13.36 skrev Stefan Sayer:
>>
>>> We are happy to announce the availability of the
>>> SIP Express Media Server version 1.4
gt;
> --
> Iñaki Baz Castillo
>
>
> ___
> sr-dev mailing list
> sr-...@lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev
---
* Olle E Johansson - o...@edvina.net
* Cell phone +46 70 593 68 51,
1 - 100 of 473 matches
Mail list logo
