Hi,
The document http://kb.asipto.com/kamailio:presence:k31-made-simple
describes that XCAP can be sent over SIP instead of over HTTP. I tried to do
however, the server is unable to update presence rules. I was using PUBLISH
method.
I was using content-type = text...
Has anyone used SIP to send over XCAP? Appreciate if someone could share a
link or configuration changes required.
Thanks
Krish
Sep 22 09:49:53 siptest /usr/sbin/kamailio[3160]: ERROR: registrar
[common.c:75]: failed to parse Address of Record
Sep 22 09:49:53 siptest /usr/sbin/kamailio[3160]: ERROR: registrar
[save.c:822]: failed to extract Address Of Record
Is there a problem when AoR contains "_"
for example is AoR krish_kura(a)sip.org is not allowed?
Thanks
Krish Kura
Hello,
while building the v3.1.5 RPMs, the perl module failed to compile on
centos 5 and 6, with following error:
error: `PL_unitcheckav' undeclared (first use in this function )
so its packaging was disabled. It works fine on opensuse and fedora.
Someone on twitter tried with the package perl-ExtUtils-Embed (like in
fedora) and worked. However, OpenSUSE Build System fails to find that
package.
Is anyone here with latest centos 5 or 6 that can try compilation of
perl module in v3.1.5 and let me know the result? Few months ago, it
didn't fail for v3.1.4.
Thanks,
Daniel
--
Daniel-Constantin Mierla -- http://www.asipto.com
Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kathttp://linkedin.com/in/miconda -- http://twitter.com/miconda
But FreeSwitch don't use MySQL like Asterisk?
So adapting isn't that easy or...???
On 21/09/2011, at 16.18, David <david(a)styleflare.com> wrote:
You can adapt pretty easily.
On 9/21/11 3:46 PM, Henrik Aagaard Sørensen wrote:
Does anyone know if there somewhere exists a tutorial about Kamailio and
FreeSWITCH realtime integration?
I have Googled a lot and found:
http://kb.asipto.com/kamailio:index and
http://kb.asipto.com/freeswitch:index
On the same site there is a tutorial for Kamailio and Asterisk realtime
integration, but not for FreeSWITCH.
_______________________________________________
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_______________________________________________
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Hello,
I tried to reproduce with a very simple config to handle register
requests. So I sent a REGISTER with expires 60 (very much the same as
the one in the trace you sent), waited a while, then sent two REGISTERs
having the CSeq incremented. All went fine.
I tested with master and 3.1 branches (so the last is actually 3.1.5).
Can you upgrade to 3.1.5 first and try again? If still fails, then
update config to print the return code for save() function, something like:
$var(r) = save("location");
xlog("code returned by save is $var(r)\n");
if($var(r) < 0)
send_reply("500", "Failure");
Let me know the value of the returned code for the register that does
not have a contact in reply.
Cheers,
Daniel
On 9/20/11 2:10 PM, Alejandro Mingo wrote:
>
> Hi Daniel-Constantin,
>
> I send you the PCAP and LOG files corresponding to one of these
> situations. As you can see in the LOG file, one thread (5444) handles
> the first REGISTER, other thread (5447) handles the second REGISTER
> (retransmission) and a third thread (5451) indicates that the binding
> has expired, six seconds after the arrival of both REGISTERs.
>
> Thanks again.
>
> Alejandro.
>
>
>
> El 20/09/2011 10:54, Daniel-Constantin Mierla escribió:
>> Hello,
>>
>> On 9/20/11 10:13 AM, Alejandro Mingo wrote:
>>> Hi,
>>>
>>> Since we installed last version of Kamailio (3.1.4)
>>
>> the last version is now 3.1.5, but this is not really the most
>> important aspect. However, it is recommended to upgrade, there were
>> some fixed to registration handling when register requests have
>> multiple contacts.
>>> we have been experiencing a big problem with REGISTER
>>> retransmissions. When the server receives a retransmitted REGISTER
>>> it removes the binding and the UAC remains unregistered until next
>>> refreshing period. I include a PCAP capture that shows this behaviour.
>>>
>>> I am not 100% sure that this problem never happened with older
>>> versions.
>>>
>>> Has anybody experienced the same problem?
>> I will try to reproduce and investigate myself. Anyhow, meanwhile
>> maybe you can provide more hints faster. Do you have error messages
>> in the syslog? Can you run with debug=4 in config file and send the
>> output in syslog for such a case?
--
Daniel-Constantin Mierla -- http://www.asipto.com
Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kathttp://linkedin.com/in/miconda -- http://twitter.com/miconda
Does anyone know if there somewhere exists a tutorial about Kamailio and
FreeSWITCH realtime integration?
I have Googled a lot and found:
http://kb.asipto.com/kamailio:index and
http://kb.asipto.com/freeswitch:index
On the same site there is a tutorial for Kamailio and Asterisk realtime
integration, but not for FreeSWITCH.
Hey Phillip,
On 20.09.2011 13:48, Phillman25 Kyriacou wrote:
> Thanks for your email.
>
> Yes dlg_manage(); has to now be called on INVITE and BYE/CANCEL messages.
> Where would i have to call loose_route()? Only on INVITE?
On *all* in-dialog requests, i.e., all requests which contain a To tag.
This may include Re-INVITEs too (but not initial INVITEs).
> My configuration did not change between 3.1.2 and 3.1.5.
>
> Call flow example:
> ==============
>
> Cisco PGW ===> Kamailio 3.1.5 ===> VOIP PROVIDER or ASTERISK PABX
>
>
> The below is my configuration.
Let's take a look at it:
> #!KAMAILIO
> #
[...]
> if(is_method("BYE|CANCEL"))
>
> {
>
> dlg_manage();
>
>
>
> # per request initial checks
> route(REQINIT);
>
> # NAT detection
> route(NAT);
>
> # handle requests within SIP dialogs
> route(WITHINDLG);
[...]
> # Handle requests within SIP dialogs
> route[WITHINDLG] {
> if (has_totag()) {
> # sequential request withing a dialog should
> # take the path determined by record-routing
> if (loose_route()) {
[...]
So route[WITHINDLG] contains the logic to track in-dialog requests.
However, you seem to call that route only from BYE and CANCEL requests,
missing out ACKs, which is likely the reason why things go wrong. So
make sure you run all in-dialog requests through that route, and you
should be fine (hopefully).
Cheers,
--Timo
Hello,
since I am attending GSoC'11 Mentors Summit and come to Bay Area again,
it is time to bring the developer training in USA, therefore I am
planning to organize a two-day seminar focused on developing Kamailio
extensions in C.
Among topics to be approached:
- internal architecture
- SIP parser
- memory manager
- locking manager
- database API
- config file language interpreter
- RPC interface
- pseudo-variables and transformations framework
- module interface – write your own extensions in C as modules
- inter-module APIs - tm, presence, a.s.o.
- documentation docbook format
This kind of training follows the pattern from the past, trying to keep
a low attendee fee, targeting just to cover the expenses with the
organization. I am discussing with several people for a good location at
a convenient price, however, I am open for new alternatives, including
companies/institutes that want to host the event at no cost in exchange
of sending few attendees free of charge. Contact me if you can help here.
Training is planned for October 24-25, right now the estimated fee per
participant is 350USD (it will not exceed 400USD). The policy to accept
people is first come, first served.
You can register at this time to reserve your seat(s), emailing to me or
to registration(a)kamailio.org. Number of seats is usually 10 to 20, last
one organized in February in Barcelona filled up quickly. Once there
will be enough interested people and a location selected, I will have
more precise details regarding the participation to this seminar and I
will send updates.
Looking forward to meeting many of you and get new US developers on
board of the project in the near future.
Cheers,
Daniel
--
http://www.kamailio.org/w/daniel-constantin-mierla/
trying to hack around the possible fix_nated_sdp("1") issue that i
outlined in previous message, i thought to remove video media
description from sdp. in sdpops module there is sdp_with_media(type),
but for some reason no sdp_remove_media(type) function. is there some
other means to get rid of sdp video media description?
-- juha