DM> so I assume is fine to compact old gcc version to something
DM> more generic...
The only thing I can think needs checking is whether openbsd still
uses an archaic gcc.
-JimC
--
James Cloos OpenPGP: 0x997A9F17ED7DAEA6
___
SIP Express Route
Hi andres
these are the invites traces
*Incoming INVITE from DID provider*
Session Initiation Protocol (INVITE)
Request-Line: INVITE sip:521234567890@kamailioServer:8080;user=phone
SIP/2.0
Message Header
Via: SIP/2.0/UDP
providerServer:5060;branch=z9hG4bK+5bef3865b99c5762a2c30f01
Ah, never mind. This was a stupid question.
In short-duration traffic land, most calls end with all branches
failing. I didn't have rtpengine_delete() cleanups in the
failure_route[] for that scenario.
The volume of calls is far too large for rtpengine's own RTP timeout
("garbage collector")
Hello,
I have a high-volume installation with 100% RTP relay using rtpengine.
We run rtpengine with a high and low port range of 8000 - 65000.
Generally the peak concurrent channel count is around 2000-4000 ports,
so in theory, so we should have enough ports for all calls currently in
process
On 2/2/16 1:36 PM, Rene Montilva wrote:
Daniel
checking the frame size trace, i notice when kamailio receive packets
less than 1100, it response with packets more than 1300 , but when the
packets are more than 1100, kamailio send packets less 1000 and
connection to softphone fail
You are go
Daniel
checking the frame size trace, i notice when kamailio receive packets less
than 1100, it response with packets more than 1300 , but when the packets
are more than 1100, kamailio send packets less 1000 and connection to
softphone fail
On Tue, Feb 2, 2016 at 12:19 PM, Rene Montilva
wrote:
Hi list
I have problem with invite packet in some wifi networks because the invite
packet is less than 1000bytes(frame size) and the access point not redirect
to softphone, this scenario happen with incoming calls to my DID, the
packets come from the provider to kamailio with a size more than 100
Hi Daniel
yes i'm sure always, for example i receive from provider a packet with 1259
and kamailio redirect with 158, but this issue is with some did provider
On Tue, Feb 2, 2016 at 11:33 AM, Daniel-Constantin Mierla wrote:
> Hello,
>
> are you sure it is happening when the packet is less than
Hello,
was there any error log messages printed by kamailio in the syslog?
What was exactly the issue, reg-id not being allowed to have the value 1?
Cheers,
Daniel
On 02/02/16 14:51, sgy wrote:
>
>
>
>
> Hi Daniel,
>
> I have solved the issue, the root cause is that parameters reg-id in
> Cont
Hello,
are you sure it is happening when the packet is less than frame size,
not when they are bigger than the frame size?
Cheers,
Daniel
On 02/02/16 15:46, Rene Montilva wrote:
>
> Hi list
>
> I have problem with invite packet in some wifi networks because the
> invite packet is less than 1000b
Hi Daniel,
I have solved the issue, the root cause is that parameters reg-id in Contact
line is not right. Thank you anyway.
Best Regards,
Shengy
At 2016-02-02 17:41:05, "Daniel-Constantin Mierla" wrote:
Hello,
can you upgrade the latest 4.2.x -- it seems you are running 4.2.0 and th
Hello,
It seems to work, now i have to solve some problems on my media server but the
Kamailio part is ok I think.
Thank you for your help !
Best regards,
Alexandre
De : sr-users [mailto:sr-users-boun...@lists.sip-router.org] De la part de Phil
Lavin
Envoyé : lundi 1 février 2016 14:41
À : Kamai
Hello,
the Fosdem organizers arranged video recording of all sessions at the
event, including the ones from Real Time Dev Room. There were also
problems from time to time, we have to wait and see when they become
available.
If you referred to the social event, it was not recorded of course, it
wa
Hello Daniel.
Thank you for your clarification,
BR
José
2016-02-02 9:53 GMT+00:00 Daniel-Constantin Mierla :
> Hello,
>
> On 28/01/16 17:01, José Seabra wrote:
> > Hello there,
> > Can anyone tell me what is the exactly memory size that kamailio use
> > for one sip register saved on memory?
> >
That is a good idea and would be great to have that information also from
all community.
When i finish these measurements i will share it here for sure.
Thank you
BR
José
2016-02-02 10:43 GMT+00:00 Daniel-Constantin Mierla :
> Hello,
>
> if there are many interested, maybe would be a good id
Hello,
On 28/01/16 17:01, José Seabra wrote:
> Hello there,
> Can anyone tell me what is the exactly memory size that kamailio use
> for one sip register saved on memory?
>
> How much memory i need allocate to kamailio in order to support
> 50.000 subscribers.
this depends a lot on your environm
Hello,
On 29/01/16 13:14, Вячеслав Завалко wrote:
> Is it possible to do a demonstration screen caller using Kamailio +
> Asterisk?
can you provide more details about what you expect to be done?
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/i
Hello,
On 29/01/16 07:03, Sharon Nathaniel wrote:
> Hi,
>
> I am not sure, if its a kamailio issue or something else, but our
> Kamailio 4.2 server crashes everyday. Throwing out these
> logs: http://pastebin.com/z1MWdk7q. We have to restart the server to
> get this fixed. Please help us understan
Hello,
if there are many interested, maybe would be a good idea to try to
collect such details per phone and scenario type, like:
- direct connection to server (nat and no-nat)
- connection via outbound proxy (nat and no-nat)
We can make a wiki page for it. If you do some measurements and share
Hello,
so I assume is fine to compact old gcc version to something more generic...
Cheers,
Daniel
On 27/01/16 12:05, Daniel-Constantin Mierla wrote:
> Hello,
>
> looking a bit over our makefiles, I am thinking of simplifying some
> parts related to compiler optimizations, as I noticed some of th
On 28/01/16 14:13, Вячеслав Завалко wrote:
> The adjustment was carried out by a manual that
> http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
> But the sharing screen is not working softphone on a given
> configuration. What can be done?
>
As said in another email, I
Hello,
On 29/01/16 08:26, Barış Şekerciler wrote:
> Hello everyone. I'm a student and I'm working on project which about SIP
> communication and PBXs with IP phones.
>
> So, my first test results were OK with JITSI. But I think I need Freeswitch
> for working with IP phones. In a word, I need co
Hello,
can you upgrade the latest 4.2.x -- it seems you are running 4.2.0 and
there were many fixes from that version, not sure if related to your
situation, but it is important to rule out the fixed issues.
Also, it is important to know the parameters you set for modules usrloc
and registrar (yo
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