Peter Dunkley writes:
> There is a list of the missing presence features in Kamailio here:
> http://www.kamailio.org/wiki/devel/completing_presence
thanks for the pointer.
> BTW, I don't know of any presence implementation that supports
> everything :-)
sure, no problems there. just wanted to
On 5/3/13 2:59 PM, Vassilis Radis wrote:
Yes, I used the term proxy to include statefullness and dialog awareness, which
makes me think: What is the point of being transaction-aware without being
dialog-aware? I am trying to find a use for it, but I cant.
Things are simpler: restartig transact
No.
There is a list of the missing presence features in Kamailio here:
http://www.kamailio.org/wiki/devel/completing_presence
BTW, I don't know of any presence implementation that supports everything :-)
Regards,
Peter
On 3 May 2013, at 20:48, Juha Heinanen wrote:
> does xcap_sever module s
does xcap_sever module support xcap-diff, i.e., is it able to publish
xcap-event when some modification of a document happens?
-- juha
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists
Daniel-Constantin Mierla writes:
> if it was something that was not working as it should, go ahead and
> backport if you tested the patch.
i tested it with xcap documents generated by sipclients, which should be
available also for macosx if you need access to rls capable sip client.
-- juha
__
Hello,
if it was something that was not working as it should, go ahead and
backport if you tested the patch.
Cheers,
Daniel
On 5/3/13 7:01 PM, Juha Heinanen wrote:
Juha Heinanen writes:
it clearly shows that the reason for not finding the uri are the escaped
chars in xcap rls-services doc.
Juha Heinanen writes:
> it clearly shows that the reason for not finding the uri are the escaped
> chars in xcap rls-services doc.
i just committed a patch to rls module (master version) that added
support for escaped chars in rls services document.
is it ok to cherry-pick the commit also to 4.0
Hi Henning,
I added record_route() and now I see an extra record-route and Via: header:
.9INVITE sip:44800800...@pstn-out.netfuse.net SIP/2.0
Record-Route:
Via: SIP/2.0/UDP 85.13.242.55;branch=z9hG4bK388f.04bc8632.1
Via: SIP/2.0/UDP 81.88.163.210:5060;rport=5060;branch=z9hG4bK82ae6ced
Ace,
So record-route goes like a stack and moves the current contact one down the
chain.
You could then presumably truncate the list of other routes for privacy if
needed?
I don't see why all "record-route" entries need to be maintained if the
conversation is mediated by Kamailio.
Leo
On 3
Am Freitag, 3. Mai 2013, 16:13:36 schrieb Leo Brown:
> [..]
> MVNO Carrier --> Our Edge Switch --> Our PSTN Switch --> Our
> carrier's switch
> |---> Our internal routing
> |switch
>
> The is
Hi
My application is for mobile (MVNO) users making calls, which will generally
end up on the PSTN via our carriers.
MVNO Carrier --> Our Edge Switch --> Our PSTN Switch --> Our carrier's
switch
|
Thank's Klaus,
now I understand how to use the extra_id_pv parameter... I already use the
rtproxy_manage in branch_route to process every branch separately, but I
was with difficult to understand how set extra_id_pv to every branch of
forking. Your example helped me to eliminate my doubts (I think
On 03/05/13 14:48, ? wrote:
I've tried to implement the first method you've stated. it seems ok
but i've found a more fundamental problem:
The event_route[websocket:closed] is called only when i teminate the
sip stack in my browser, but if i close the browser, without a
"regular dis
Hi!
Disclaimer: I never used the 'b' parameter. comments inside
On 03.05.2013 15:44, Bruno Bresciani wrote:
Hi All again,
Somebody can help me about my doubts? I can't get understanding how set
extra_id_pv parameter of the rtpproxy module... The documentation show
the bellow line:
modparam("r
On 03.05.2013 11:59, Vassilis Radis wrote:
Thank you jiri,
I totally agree, but I have a question that occured to me now and I cant
find answer:
If Kamailio receives a CANCEL from a UAC after kamailio has responded
with a 200 to the corresponding INVITE, what does t_relay_cancel() do in
the f
I've tried to implement the first method you've stated. it seems ok but
i've found a more fundamental problem:
The event_route[websocket:closed] is called only when i teminate the sip
stack in my browser, but if i close the browser, without a "regular
disconnect" then the w*ebsocket:closed event
Hi All again,
Somebody can help me about my doubts? I can't get understanding how set
extra_id_pv parameter of the rtpproxy module... The documentation show the
bellow line:
modparam("rtpproxy", "extra_id_pv", "$avp(extra_id)")
What means value "$avp(extra_id)"? I don't understand how set correc
On 05/03/2013 08:59 AM, Vassilis Radis wrote:
What is the point of being transaction-aware without being
dialog-aware? I am trying to find a use for it, but I cant.
Lots of uses! Retransmission dampening, serial forking (branches),
and many more.
-- Alex
--
Alex Balashov - Principal
Eva
Yes, I used the term proxy to include statefullness and dialog awareness,
which makes me think: What is the point of being transaction-aware without
being dialog-aware? I am trying to find a use for it, but I cant.
On Fri, May 3, 2013 at 3:34 PM, Jiri Kuthan wrote:
> Hi Bill,
>
> plain-kamailio
@Olle
To be more specific, I only want UDP with SIP encryption
On Fri, May 3, 2013 at 7:56 PM, Khoa Pham wrote:
> @Olle. thanks for your reply
>
> 1. I only want stream encryption to avoid SIP ALG (which can modify SIP
> message wrongly)
> 2. What is the max concurrent TCP connections can Kama
@Olle. thanks for your reply
1. I only want stream encryption to avoid SIP ALG (which can modify SIP
message wrongly)
2. What is the max concurrent TCP connections can Kamailio handle ? I heard
there is a tcp_max_connections, is that the answer ?
On Fri, May 3, 2013 at 7:20 PM, Olle E. Johansson
Hi Bill,
plain-kamailio cannot send BYEs (not sure if some module can). The point is the
proxy is
sort of "passive element" and doesn't initiate transactions on its own.
Why isn't it enough to have the BYEs sent by UAC? I mean sometimes there can be
some
confusing situations (say forking downs
TLS has two phases - key exchange and encryption.
Seems like you only want stream encryption, which means you will have to go
deep down in the
TLS module and the OpenSSL library.
Why on earth do you want to use a static key? That seems to contradict the need
for protection.
/O
3 maj 2013 kl.
@Daniel
TLS is OK, but many TCP connections will makes client suffering from 503
Service Unavailable error. Please consider this as a feature request :)
On Tue, Apr 2, 2013 at 11:18 AM, Khoa Pham wrote:
> Hi,
>
> Currently, I'm using TLS and it works fine. But eventually, TLS is just
> used to
Thank you jiri,
I totally agree, but I have a question that occured to me now and I cant
find answer:
If Kamailio receives a CANCEL from a UAC after kamailio has responded with
a 200 to the corresponding INVITE, what does t_relay_cancel() do in the
following 2 cases:
1. CANCEL received before th
On 5/3/13 11:04 AM, Vassilis Radis wrote:
Hello all,
In the documentation of the t_relay_cancel() (TM module) there is an example
that reads:
if (method == CANCEL) {
if (!t_relay_cancel()) { # implicit drop if relaying was successful,
# nothing to do
# corr
Hello all,
In the documentation of the t_relay_cancel() (TM module) there is an
example that reads:
if (method == CANCEL) {
if (!t_relay_cancel()) { # implicit drop if relaying was successful,
# nothing to do
# corresponding INVITE transaction found but error o
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