in advance for your attention !
>>>>
>>>> Best regards,
>>>>
>>>> Anton
>>>>
>>>>
>>>>
>>>>
>>>> ___
>>>> SIP Express Router (SER) and Kama
t the venue or in a
nearby pub.
Full details here: www.meetup.com/London-VoIP-User-Group-LVUG/
Hope to see you there!
Richard
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Richard Brady
E: rnbr...@gmail.com
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Hi Alexandr
On 28 March 2014 15:36, Alexandr Usov 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 side.
>
> I can't
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 wrote:
> Am Donnerstag, 23. Mai 2013, 12:20:00 schrieb Henning Westerholt:
>> [...]
>>> And should the next major release
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), wou
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 wrote:
> Agreed, doesn't make sense to me either.
>
> The code is in the decode2format function in siputils/contact_ops.c:
>
>
Hi Klaus
Thanks for posting the working solution.
You are right that mhomed=1 won't help if the sockets both have same IP
address. Though I think if you used it with two different IPs and the
correct routing tables on the OS then it would work.
In any case your solution is a good one.
Richard
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 he
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:
listen=192.168.1.1
> 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 m
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 430
> 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 i
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 wrote:
> Hello Klauss,
>
> I use record_route_preset for this kind of scenarios:
> http://kamailio.org/docs/modules/3.3.x/m
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 */
> 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 registra
> 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
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-U
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,
ce would be hugely appreciated. I can always paste logs /
configs / traces.
Richard
--
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-user
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 t
in 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
>>> interf
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 testi
Hi guys, sorry for the delay.
> Do you have an example of the cfg you can share?
Sure thing. See attached. This doesn't contain the Path header, as I have
statically configured fs_path in the dialplan on FreeSWITCH.
I am exploring a possible bug in FreeSWITCH where is advertising the wrong
trans
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 wrote:
>
>
> On 22.06.2012 13:50, Richard Brady wrote:
>>
>> Than
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 wrote:
> After endless tests, I tried to replace record_route_preset with insert_hf,
> writing the complete
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 ar
0 OK coming from behind NAT it is not subjected to
fix_nated_contact(), and this seems to be because it doesn't have the
FLB_NATB flag set:
if (is_reply()) {
if(isbflagset(FLB_NATB)) {
fix_nated_contact();
}
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