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-pr
Andrei Pelinescu-Onciul writes:
> > one more thing i would like to add to the tuto if someone knows the
> > answer: how to capture/see sip over ssl packets in the proxy host using
> > wireshark, ngrep, or something?
>
> ssldump -k modules/tls/sip-router-selfsigned.key tcp and port 5061
i tried
On Sep 30, 2010 at 19:07, Simone Felici 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 I should have
On Sep 30, 2010 at 19:56, Juha Heinanen 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
> > (http://sip-router.org/docb
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
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
>
On Sep 30, 2010 at 18:44, Juha Heinanen 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 session is esta
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
sr-users@lists.sip-router.
On Sep 30, 2010 at 18:27, Juha Heinanen 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
>
> http://kamailio.org/dokuwiki/doku.php/tls:create-certif
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
http://kamail
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
_
On Sep 29, 2010 at 10:43, Simone Felici 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 call to our p
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
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 th
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:
http://www.for
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
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:
http://www.forum.nokia.com/info/sw.nokia.com/id/e8606
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
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
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 RE
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 atte
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 a
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
log
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.
Regards,
Ovidiu Sas
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@li
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 mor
Andrei Pelinescu-Onciul writes:
> Are you sure that it does not receive a negative reply from the gateway
> that triggers the failure route?
andrei,
as i mentioned, i don't have yet wireshark dump from gw side, but i see
from sr syslog entry that sr is receiving "186 session progress" from the
g
Hello,
On 9/30/10 2:54 PM, X- x -X wrote:
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
On Sep 30, 2010 at 14:10, Juha Heinanen 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 dump of tha
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
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 t
2010/9/30 Daniel-Constantin Mierla :
> 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
trigger a reload
Andrei,
30.09.10, 03:56, "Andrei Pelinescu-Onciul" :
>
> 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 know, is
On 9/30/10 12:57 AM, Iñaki Baz Castillo wrote:
2010/9/29 Carsten Bock:
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 actions (for exam
On 9/30/10 12:53 AM, Iñaki Baz Castillo wrote:
2010/9/29 Daniel-Constantin Mierla:
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 contacts"
should be:
Hello,
On 9/29/10 5:50 PM, Jon Farmer wrote:
Hi,
I currently have 1 OpenSer server in front of a number of Asterisk
boxes. I want to move to a scenario where I have multiple OpenSer
boxes serving the same domain name in front of the Asterisk boxes. The
UA could be registered on either of the O
36 matches
Mail list logo
