On 9/30/10 12:53 AM, Iñaki Baz Castillo wrote:
2010/9/29 Daniel-Constantin Mierlamico...@gmail.com:
I published a tutorial of how to implement a SIP SIMPLE Presence XCAP
server with latest Kamailio/SER version. You can find it at:
http://bit.ly/btrJij
Hi, a typo:
body lists - your
On 9/30/10 12:57 AM, Iñaki Baz Castillo wrote:
2010/9/29 Carsten Bockli...@bock.info:
I published a tutorial of how to implement a SIP SIMPLE Presence XCAP
server with latest Kamailio/SER version. You can find it at:
http://bit.ly/btrJij
Hi, the DELETE action should also trigger refresh
Andrei,
30.09.10, 03:56, Andrei Pelinescu-Onciul and...@iptel.org:
In the meantime I managed to reproduce it.
It should be fixed in all the versions now.
Andrei
That's great! Because I could not reproduce this problem on system compiled
with this debug flags.
By the way, I don't
2010/9/30 Daniel-Constantin Mierla mico...@gmail.com:
I wonder if a delete can change the status of watcher. Usually is translation
from none to pending to active, but can be other way around triggered by
xcap operation?
In OMA specs *any* change (PUT/DELETE) in resource-lists document must
there is a broken grandstream sip ua that sometimes sends two initial
invites back-to-back. according to wireshark, sr received them about 10
microseconds apart. sr then forwards the request to pstn gw, but i
don't have wireshark dump of that side of the traffic. anyway, from the
ua side i see
Hi Daniel.
Thanks. It works like a charm. Very expressive tutorial.
but just looking at the logs I see stuff like this which is erroring.
doesnt crash the server or stuff just some error
Tested with sip-communicator-1.0-alpha6-nightly.build.2976-linux.bin
And the specs/conf on that tutorial
On Sep 30, 2010 at 14:10, Juha Heinanen j...@tutpro.com wrote:
there is a broken grandstream sip ua that sometimes sends two initial
invites back-to-back. according to wireshark, sr received them about 10
microseconds apart. sr then forwards the request to pstn gw, but i
don't have wireshark
Daniel-Constantin Mierla writes:
SC and xcap support is in an early stage as well, I mentioned in the
tutorials some things, like auto-adding accept rule for an watcher when
you subscribe to it (you become watcher for your watcher). I am waiting
for a bug to be fixed in bria 3.1 for more
On Thursday 30 September 2010, Ovidiu Sas wrote:
The jabber module is obsolete and it still exists in both (s) and (k)
module version.
We should remove this module from the upcoming 3.1 release.
Hello Ovidiu,
i'd second this. I've added a warning this february to the README and startup
Hi,
I would like to know if the following is possible in Kamailio, I've
tried with OpenSIPs but I don't think it is ideal for my needs.
I would like to load balance multiple asterisk boxes which terminate
and originate calls. To transfer calls by attended transfer any new
calls originating from
Ross,
On 09/30/2010 10:07 AM, Ross Beer wrote:
I would like to know if the following is possible in Kamailio, I've
tried with OpenSIPs but I don't think it is ideal for my needs.
I would like to load balance multiple asterisk boxes which terminate
and originate calls. To transfer calls by
Ross,
On 09/30/2010 10:14 AM, Ross Beer wrote:
The phone sets up another call using INVITE and then uses REFER when
the transfer takes place.
Please try to keep the list copied, as a matter of good practice.
If I understood your scenario correctly, you don't have a choice but
to send the
On 9/30/10 3:27 PM, Juha Heinanen wrote:
Daniel-Constantin Mierla writes:
SC and xcap support is in an early stage as well, I mentioned in the
tutorials some things, like auto-adding accept rule for an watcher when
you subscribe to it (you become watcher for your watcher). I am waiting
for a
Hello to all!
I need a little help with our ser installation (ser-2.0.0-rc1).
The continuous groving up of our infrastructure and using even more codecs, causes the INVITE (udp) to be over 1500bytes. An
external-incoming call to our proxy sip comes in with a size of ~1300 and will be forwarded
Hello to all!
I need a little help with our ser installation (ser-2.0.0-rc1).
The continuous groving up of our infrastructure and using even more codecs, causes the INVITE (udp) to be over 1500bytes. An
external-incoming call to our proxy sip comes in with a size of ~1300 and will be forwarded
Daniel-Constantin Mierla writes:
what are the models for these nokia phones? I have some with sip client
inside, but they might be old -- i haven't seen any xcap settings
there.
daniel,
the phone should have VoIP Release 3.0 or later:
now that 3.1 has async tls support, i decided (first time ever) to try
to test tls. things went quite smoothly when i followed Create
Certificates to be used with Kamailio document
http://kamailio.org/dokuwiki/doku.php/tls:create-certificates#using_the_certificates_with_tls
during the process,
On 9/30/10 5:13 PM, Juha Heinanen wrote:
Daniel-Constantin Mierla writes:
what are the models for these nokia phones? I have some with sip client
inside, but they might be old -- i haven't seen any xcap settings
there.
daniel,
the phone should have VoIP Release 3.0 or later:
I added note about configuring Snom phones to connect over TLS and
created a section from that part:
http://kamailio.org/dokuwiki/doku.php/tls:create-certificates#using_tls_and_the_certificates_with_sip_phones
Also, in my configs I set:
tcp_connection_lifetime=3610
Which is slightly higher
Hello,
On 9/30/10 4:03 PM, Henning Westerholt wrote:
On Thursday 30 September 2010, Ovidiu Sas wrote:
The jabber module is obsolete and it still exists in both (s) and (k)
module version.
We should remove this module from the upcoming 3.1 release.
Hello Ovidiu,
i'd second this. I've added a
On Sep 29, 2010 at 10:43, Simone Felici s.fel...@mclink.eu wrote:
Hello to all!
I need a little help with our ser installation (ser-2.0.0-rc1).
The continuous groving up of our infrastructure and using even more
codecs, causes the INVITE (udp) to be over 1500bytes. An
external-incoming
one question about the certificate tutorial: is something else needed in
the config or certificate business, when sr talks over tls with another
sip proxy, e.g. another sr? namely in that case sr may be in client
role when tls session is established.
-- juha
You are right.
Thanks for fixing my bugs :-)
Klaus
Am 30.09.2010 17:27, schrieb Juha Heinanen:
now that 3.1 has async tls support, i decided (first time ever) to try
to test tls. things went quite smoothly when i followed Create
Certificates to be used with Kamailio document
On Sep 30, 2010 at 18:27, Juha Heinanen j...@tutpro.com wrote:
now that 3.1 has async tls support, i decided (first time ever) to try
to test tls. things went quite smoothly when i followed Create
Certificates to be used with Kamailio document
Daniel-Constantin Mierla writes:
Also, in my configs I set:
tcp_connection_lifetime=3610
so do i. i added that line to the wiki doc.
-- juha
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
On Sep 30, 2010 at 18:44, Juha Heinanen j...@tutpro.com wrote:
one question about the certificate tutorial: is something else needed in
the config or certificate business, when sr talks over tls with another
sip proxy, e.g. another sr? namely in that case sr may be in client
role when tls
Andrei Pelinescu-Onciul writes:
enable_tls=1
tcp_async=no # do not include in 3.1
listen=udp:0.0.0.0:5060
listen=tcp:0.0.0.0:5060
it should not be 0.0.0.0 but an actual IP.
If you use 0.0.0.0 you _must_ set adevertised_adress or
you will
Andrei Pelinescu-Onciul writes:
However if you want to have different certificates in function of the
role (server or client, or who are you talking with, you need to use a
separate tls config
file
(http://sip-router.org/docbook/sip-router/branch/master/modules/tls/tls.html#config)
ok
On Sep 30, 2010 at 19:56, Juha Heinanen j...@tutpro.com wrote:
Andrei Pelinescu-Onciul writes:
However if you want to have different certificates in function of the
role (server or client, or who are you talking with, you need to use a
separate tls config
file
On Sep 30, 2010 at 19:07, Simone Felici s.fel...@mclink.eu wrote:
Thank you a lot for your answer!
I'll try these modifications, starting from some tune on actual
config of ser-2.0 to bring all working correctly and then ending
with a test phase of the ser-3.0 version.
I'll let you know if
Juha Heinanen writes:
i tried with command
ssldump -i any -k /etc/sip-proxy/certs/sip-proxy/key.pem tcp and port 5061
where /etc/sip-proxy/certs/sip-proxy/key.pem is the same file as
specified as tls module private key:
modparam(tls, private_key, /etc/sip-proxy/certs/sip-proxy/key.pem)
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