___
>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>>> sr-users@lists.sip-router.org
>>>> <javascript:_e(%7B%7D,'cvml','sr-users@lists.sip-router.org');>
>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
at the venue or in a
nearby pub.
Full details here: www.meetup.com/London-VoIP-User-Group-LVUG/
Hope to see you there!
Richard
--
Richard Brady
E: rnbr...@gmail.com
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip
Hi Alexandr
On 28 March 2014 15:36, Alexandr Usov blessen...@gmail.com wrote:
I am already have some practice to integrate Kamailio with Asterisk, when
all users creates and registers in Kamailio, and calls go to/from Asterisk
with static host=kamailio_ip settings for each user on Asterisk
Thanks Henning!
Should anything be done for 3.x? Feels like this is enough of a bug to warrant
reverting to branch=0.
Richard
On 24 May 2013, at 14:54, Henning Westerholt h...@kamailio.org wrote:
Am Donnerstag, 23. Mai 2013, 12:20:00 schrieb Henning Westerholt:
[...]
And should the next
Hi folks
The syn_branch global parameter results in the use of a synonym
branch parameter in the Via header for statelessly forwarded requests
as a performance optimisation.
This was originally done by setting branch=0 which, while not strictly
compliant with 3261 (8.1.1.7 and 16.6 item 8),
Hi folks
If an empty or invalid JSON string is passed to the json_get_field function
from the the json module, it causes a segfault.
I have attached a patch.
Regards,
Richard
--
Richard Brady
M: +44 (0)7771 623 348
T: +44 (0)20 8144 8160
E: rnbr...@gmail.com
json_segfault_fix.patch
Patch attached.
Should this be cross posted to [sr-dev] if it contains a patch?
Richard
On 7 January 2013 01:10, Richard Brady rnbr...@gmail.com wrote:
Agreed, doesn't make sense to me either.
The code is in the decode2format function in siputils/contact_ops.c
Also can a flow fail temporarily?
For example a broadband router with a NAT timeout of 60 seconds and a UA
with a keep-alive interval of 120s. Would the flow succeed for the first 60
seconds after each keep-alive and then fail for 60 seconds until the next
keepalive?
And would this generate a
Also can a flow fail temporarily?
For example a broadband router with a NAT timeout of 60 seconds and a UA
with a keep-alive interval of 120s. Would the flow succeed for the first 60
seconds after each keep-alive and then fail for 60 seconds until the next
keepalive?
Yes. That's a
Hi Tuan
Path doesn't do NAT traversal (although it plays a role) so it isn't a
replacement for fix_nated_contact and fix_nated_register. The best approach
is to replace them with add_contact alias and handle_ruri_alias.
Also be sure to set the advertised address on External A:
Hi Owen
The module overview describes this behaviour:
Every time when a user registers with Kamailio, the module is looking in
database for offline messages intended for that user. All of them will be
sent to contact address provided in REGISTER request.
The REGISTER request contact header
i didn't find in rfc5626 a requirement that registrar should remove 430
flow contact,
Closest I can find is:
EP1 no longer has a flow to Bob, so it responds with a 430 (Flow
Failed) response. The proxy removes the stale registration and tries
the next binding for the same instance.
Hi Peter
Great work on this! We'd like to help you test.
The test I have in mind, which we could create using SIPp, would be to
register multiple contacts with the same instance-id (i.e. sip.instance
param) but different reg-id params. Then send an INVITE to that AoR and
make sure the forking is
Agreed, doesn't make sense to me either.
The code is in the decode2format function in siputils/contact_ops.c:
if (((*pos) == '')||(*pos == ';'))
{
/* invalid chars inside username part */
Hey Klaus
The way you described works for me (on EC2) and I think is a good solution.
Be sure to set mhomed=1 in your config.
Richard
On 4 January 2013 17:57, Ovidiu Sas o...@voipembedded.com wrote:
Hello Klauss,
I use record_route_preset for this kind of scenarios:
Can anyone point to a good example of Kamailio performing the function of
Re-direct server.
We would like to forward messages onto another domain for authentication.
Just to be clear, are you sure you want to redirect it, i.e. send a
302 response back to the client?
The other option being to
can anyone give me some pointers or a list of some of these complications
so that I can further research to see how its is done.
There are 3 things you should do:
1. Stop Kamailio from acting as the registrar.
2. Make sure that NAT traversal information is captured and relayed to
the
Hi Andrew
well, the RURI of remote ACK has proxy IP address 10.200.70.100 so proxy
thinks that previous hop was a strict router.
Aha, that makes sense. But why would that make Kamailio apply strict routing?
Now I understand this! Kamailio is not copying the *next* Route URI
into the R-URI
Thanks Andrew
well, the RURI of remote ACK has proxy IP address 10.200.70.100 so proxy
thinks that previous hop was a strict router.
Aha, that makes sense. But why would that make Kamailio apply strict routing?
I can't think of any
workaround that would not be an ugly hack at the moment,
--
Richard Brady
M: +44 (0)7771 623 348
T: +44 (0)20 8144 8160
E: rnbr...@gmail.com
___
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
A bit late to dig up this old conversation but my thoughts:
The simplest SIP relationship is from one UA to another UA without proxies.
I love proxies, I really do, they are so powerful. But putting a proxy
between UAs (or B2BUAs) places higher interoperability requirements on
them. For example
, Daniel-Constantin Mierla wrote:
Hello,
On 9/10/12 3:52 PM, Richard Brady wrote:
The other thing to consider within the flow switching part is the
ability to provide a limited version of a STUN server on the same
interface and UDP port as SIP.
this is implemented (or at least partial) - I had
The other thing to consider within the flow switching part is the ability
to provide a limited version of a STUN server on the same interface and
UDP port as SIP.
That sounds like heavy going. Would it be doable within Kamailio?
I think this is a great RFC so would be happy to help with testing.
Klaus / Daniel
Thanks again for assistance with this.
I've tried the solution based on add_contact_alias() and
handle_ruri_alias() and it works perfectly.
Richard
On 22 June 2012 13:47, Klaus Darilion klaus.mailingli...@pernau.at wrote:
On 22.06.2012 13:50, Richard Brady wrote:
Thanks
Hi Reda
A bit late for a reply but I found your post recently and it helped me
to solve a similar problem, so I wanted to share one possible
solution.
On 21 January 2012 23:19, Reda Aouad reda.ao...@gmail.com wrote:
After endless tests, I tried to replace record_route_preset with insert_hf,
Thanks guys, fantastic answers.
You mention that NAT detection happens before save() and the flag is set by
lookup() which makes much more sense. However, if Kamailio is not the
registrar, as is the case with my current project, those functions are not
called, so an alternative is needed. There
to be because it doesn't have the
FLB_NATB flag set:
if (is_reply()) {
if(isbflagset(FLB_NATB)) {
fix_nated_contact();
}
}
Any help clarifying would be much appreciated.
Thanks,
Richard
--
Richard Brady
27 matches
Mail list logo
