On 04.05.2013 16:37, John Chen wrote:
Hi,
I can configure kamailio + rtpproxy to enable calling between user
behind the NAT. Thanks for the example config, it is very easy to did it.
Do we need rtprpoxy for all kind of NAT?
From what i have read, only symmetric NAT that hard to be traversal-
6 maj 2013 kl. 09:49 skrev Klaus Darilion :
>
>
> On 04.05.2013 16:37, John Chen wrote:
>> Hi,
>> I can configure kamailio + rtpproxy to enable calling between user
>> behind the NAT. Thanks for the example config, it is very easy to did it.
>>
>> Do we need rtprpoxy for all kind of NAT?
>>
>
> When stating "stable" and "bug-free" - my intention was relativly to that
> feature in kamailio because it seems that it is still under development.
> Moreover - certainly if i will find any problem while testing i would
> submit a bug-report and bug-fixes if possible.
>
> and by the way, I fou
Am Freitag, 3. Mai 2013, 16:33:23 schrieb Leo Brown:
> So record-route goes like a stack and moves the current contact one down the
> chain.
>
> You could then presumably truncate the list of other routes for privacy if
> needed?
> [..]
Hello,
Kamailio can't provide "real" SBC functionality, but
Am Freitag, 3. Mai 2013, 16:54:29 schrieb Leo Brown:
> I added record_route() and now I see an extra record-route and Via: header:
>
> .9INVITE sip:44800800...@pstn-out.netfuse.net SIP/2.0
> Record-Route:
> Via: SIP/2.0/UDP 85.13.242.55;branch=z9hG4bK388f.04bc8632.1
> Via: SIP/2.0/UDP 81.
Good day!
Now I test new style AVP's for radius response storing. Today
additional radius responses return in special radpair:
SIP-AVP - name, string "email:sr-users@lists.sip-router.org session-timeout#161
nexthopip:h323/0001...@myvoip-gate.kamailio.org" - value.
I don't know why developers ch
Hi Khoa. The maximum number of concurrent TCP connections depend on the
computational capacity of the server. Kamailio can be regarded as an
application server of high performance SIP/VoIP. In the following post, the
Kamailio user set max TCP connections to 64K.
http://lists.sip-router.org/pipe
Is it possible to run on phones Cisco 79xx series BLFs? So to show the
status of the shared lines. Phones registered with Kamailio.
Andy
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Hello,
Is there a limit on the max number of pipes for pipelimit module?
Thanks,
Julia
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Hi,
I have built a kamailio based phone system along the lines of the
tutorial below. I have updated it for version 4.0 and it is running
successfully on Debian 7.0. I have set my system up with a Gandi SSL
cert rather than a self signed one.
http://
Hi,
As someone asked in
http://lists.sip-router.org/pipermail/sr-users/2010-December/066792.html about
TCP overload, Daniel answerd that
"if you have some lengthily operations that have to be applied to your sip
traffic and you get congestion because of that, the solution is to use a
kamailio as
Hi Yang Hong, thanks for your reply and those papers
1. When client make TCP connection, it usually get [FIN,ACK] and [RST] from
server. Does that mean that server has reach max TCP connection limit, or
server is busy handling other connections?
P/S: proposed that my server is very strong
On Mo
On Tue, May 7, 2013 at 10:10 AM, Khoa Pham wrote:
> Hi,
>
> As someone asked in
> http://lists.sip-router.org/pipermail/sr-users/2010-December/066792.html
> about TCP overload, Daniel answerd that
>
> "if you have some lengthily operations that have to be applied to your sip
> traffic and you get
hi,
i am chinese,i am not good at kamailio.i want to install kamailio with
websocket function and msrp function on ubuntu,could you give me a help?
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Hi,
I installed and configured the latest Kamalio and all the features works
except PRESENCE
Also it works with NAT clients with RTPProxy.
When I register with Jitsi or QuteCom I get
5(10515) ERROR: presence [publish.c:388]: No E-Tag and no body found
6(10517) INFO: presence [notify.c:1601]: NO
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