Dear All,
.May be I couldn't express my doubt properly ijn my earlier mails.Actually
we were testing OfficeSIP server with SipML5 client for multiparty
videoconferencing with presence enabled for showing buddy list and it is
working properly.Now as we moved to kamailio with same client and
configu
16 sep 2013 kl. 19:11 skrev Steve Davies :
>> Why not forward the registration from the endpoint and let them handle
>> authentication?
>>
>
> I need to push the registration upstream. But I also need a local
> location record since I have an Asterisk on the side that also needs
> to send cal
Hello,
you can call save() for 200ok received from downstream. In case you deal
with nat or multiple local sockets, you need to call the functions for
fixing the registration as well as storing the local socket.
The processing can be like:
- forward register
- if getting 200ok, call save() fo
On 16 September 2013 21:30, Daniel-Constantin Mierla wrote:
>
> you can call save() for 200ok received from downstream. In case you deal
> with nat or multiple local sockets, you need to call the functions for
> fixing the registration as well as storing the local socket.
>
> The processing can be
On 16 September 2013 19:49, Olle E. Johansson wrote:
>
> 16 sep 2013 kl. 19:11 skrev Steve Davies :
>
> Why not forward the registration from the endpoint and let them handle
> authentication?
>
>
> I need to push the registration upstream. But I also need a local
> location record since I have
Hi,
Just managed to check code. Correct me if I'm wrong, but I think it won't
work with add_path() because the function uses substitution SUBST_SND_ALL
which of course refers to the source ip:port;transport=proto of the
outgoing message.
Regards,
Charles
On 16 September 2013 16:06, Charles Ch
On 16 September 2013 21:03, Olle E. Johansson wrote:
> I say you're making it too complex. Let the proxy be a proxy and force the
> client to behave properly.
>
>
>
The client is an ordinary SIP phone. I don't have any control over its
behaviour. But I'm not sure in what way it is behaving impr
16 sep 2013 kl. 20:54 skrev Steve Davies
:
>
>
>
> On 16 September 2013 19:49, Olle E. Johansson wrote:
>
> 16 sep 2013 kl. 19:11 skrev Steve Davies :
>
>>> Why not forward the registration from the endpoint and let them handle
>>> authentication?
>>>
>>
>> I need to push the registrati
16 sep 2013 kl. 18:08 skrev Steve Davies :
> Hi,
>
> I'm making slow but steady progress with my Kamailio project.
>
> My next task: I need to "relay" a registration to an upstream service. IE
> when one or more devices registers against my local registration service, I
> need to initiate a
I'm thinking about changing the architecture of the system I'm building. As
it stands now, my registrars are on the edge of the network and they share
a DB. If a user calls another user on another server, the callers registrar
sends an INVITE to the callee and the caller's rtpproxy is used by both
> Why not forward the registration from the endpoint and let them handle
> authentication?
>
I need to push the registration upstream. But I also need a local
location record since I have an Asterisk on the side that also needs
to send calls to the phones via the Kamailio instance.
The device o
Hi,
I'm making slow but steady progress with my Kamailio project.
My next task: I need to "relay" a registration to an upstream service. IE
when one or more devices registers against my local registration service, I
need to initiate a corresponding registration upstream. If all my local
devices
Peter - calling msg_apply_changes() after add_path() should indeed work,
I'm sure, so thanks for pointing it out. However, when testing I'm getting
parse errors in debug output, so am unable to confirm at this time.
Brian - which version are you using? Anything in log? After calling save(),
path c
In my test setup the registrar is at the edge and add_path() does in fact
do nothing. I also tried Charles' suggestion which didn't seem to add the
Path header either.
On Mon, Sep 16, 2013 at 10:04 AM, Peter Dunkley <
peter.dunk...@crocodilertc.net> wrote:
> I thought append_hf() didn't take aff
Hello,
I have a problem with Kamailio version 3.2.4 (tested also with 3.3.5)
regarding "binary data" in message bodies. According RFC3261 it is
explicitly allowed using binary data within MIME bodies:
RFC 3261 section 7.4.1: SIP messages MAY contain binary bodies or body
parts. When no explici
I thought append_hf() didn't take affect (unless you use
msg_apply_changes()) until the message left Kamailio too?
If that is the case, and msg_apply_changes() is called, doesn't that mean
the Path: header from add_path() would be added in that scenario?
On 16 September 2013 14:49, Charles Chanc
It is possible if the edge proxy and registrar are separate, as you say.
But if the registrar is at the edge with no separate proxy, add_path() does
nothing (because the message never leaves Kamailio for the header to be
added).
On 16 September 2013 14:42, Peter Dunkley wrote:
> Is there any rea
Is there any reason it isn't possible?
On 16 September 2013 14:21, Charles Chance wrote:
> Hi,
>
> Yes, you are right - and I agree, it would be better if this was possible
> :)
>
> Charles
>
>
>
>
> On 16 September 2013 14:15, Peter Dunkley
> wrote:
>
>> Hello,
>>
>> It'd be better if the "add
Hi Brian,
Another tip, if you do share the same database, make sure the clocks are in
sync ;)
Cheers,
Charles
On 16 September 2013 14:22, Brian Wallen wrote:
> Thanks for the tips guys. I'll try them out today and report back.
>
>
> On Mon, Sep 16, 2013 at 9:21 AM, Charles Chance <
> charle
Thanks for the tips guys. I'll try them out today and report back.
On Mon, Sep 16, 2013 at 9:21 AM, Charles Chance <
charles.cha...@sipcentric.com> wrote:
> Hi,
>
> Yes, you are right - and I agree, it would be better if this was possible
> :)
>
> Charles
>
>
>
>
> On 16 September 2013 14:15, Pe
Hi,
Yes, you are right - and I agree, it would be better if this was possible :)
Charles
On 16 September 2013 14:15, Peter Dunkley wrote:
> Hello,
>
> It'd be better if the "add_path()" function could be used here. That way,
> if using outbound (RFC5626), the flow-token (the userinfo part o
Hello,
It'd be better if the "add_path()" function could be used here. That way,
if using outbound (RFC5626), the flow-token (the userinfo part of the
Path-URI) would be present and there would be no need to add the
";received" parameter.
This would address the one issue remaining for SIP outbou
Hi,
This sounds like a case for sharing same database, and adding Path before
saving incoming register. That way, no need to replicate register message
to other servers and all subscribers use the same domain.
Add path something like this before calling save():
append_hf(
On Mon, Sep 16, 2013 at 7:34 AM, Daniel-Constantin Mierla wrote:
> Hello,
>
>
> On 9/12/13 10:08 PM, Brian Wallen wrote:
>
>> I currently have two independent kamailio servers. I'd like to set them
>> up in a way that user1 on server1 can make a call to user2 on server2.
>> After searching I've c
Hello,
On 9/12/13 10:08 PM, Brian Wallen wrote:
I currently have two independent kamailio servers. I'd like to set
them up in a way that user1 on server1 can make a call to user2 on
server2. After searching I've come up with two ways that this might be
able to be done. Can someone please sanit
16 sep 2013 kl. 08:41 skrev Hassan Wajahat :
> Hi,
>
> I followed your tutorial and the setup is running seamlessly. But the
> question is that I read somewhere that asterisk can make up to 300 to 400
> concurrent sip calls. Higher than that will cause problems. Will that issue
> appear here
Hello,
discussed last week during IRC development meeting, we are approaching
the time for getting out a new major release - to be numbered v4.1.x.
I would proposed October 7 as the last day do accept new features for
v4.1.0. That gives another three weeks to push new code. As a reminder,
th
Hi,
I followed your tutorial and the setup is running seamlessly. But the
question is that I read somewhere that asterisk can make up to 300 to 400
concurrent sip calls. Higher than that will cause problems. Will that issue
appear here as well since you are using asterisk in the sip calling
proces
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