Hey Daniel,
Thanks for the answer. It makes more sense now.
Have a good one!
DanB
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Hello,
there were some workarounds to store bigint as a string in avps,
probably done in modules such as sqlops. The DB api itself supports this
type, but the AVP can store inside only string or integer values. In
this case, auth_db has to be updated to deal with this case.
Cheers,
Daniel
Hello,
On 6/1/13 10:30 PM, hiro wrote:
I tried for multiple hours to operate the debugger, also looking at
-ddd output at stdout for many days.
But I'm none the wiser.
voicemail works from route[location] (e.g. if extension is not
registered), but not after late errors like busy while ringing.
Hello,
On 6/2/13 10:57 PM, hiro wrote:
I'm still thinking about this issue and wondering:
is it even compliant to the RFC to go directly from ringing to session
progress and then OK?
if I understand correctly what you mean, then it is ok from RFC point
of view. Why you think would be a
Hi Isaac,
On 5/31/13 11:55 PM, Isaac A. McDonald wrote:
Hello All,
I'm a bit confused as to how Siremis interacts with Kamailio. From
what I understand, the dial-plan is done in the kamailio.cfg with
regex statements and requires a restart of the Kamailio process to
take effect. How then
At some point i got session progress and then loads of OKs back from
freeswitch, but either the phone didn't receive or didn't accept it.
the ringing tone would keep on playing in the phone instead. Randomly
the session progress seems to still get processed by the phone, i
would hear freeswitch
Dear 2.8 users,
this update includes the fixes for kamailio-lb config (multiple
sockets-related):
- added record-route param to store socket; review logic for relaying
in-dialog requests (fallback to default send socket if it's not defined
explicitly); fixed socket selection for ACK after 4xx and
Andrew Pogrebennyk writes:
Dear 2.8 users,
this update includes the fixes for kamailio-lb config (multiple
sockets-related):
i would prefer to keep sr-users list to generic kamailio/sip router
related messages.
-- juha
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SIP Express Router (SER)
Hello Dan,
You can also use the built in web provisioning interface:
http://kamailio.org/docs/modules/4.0.x/modules/xhttp_pi.html
You can build your own provisioning layout (you can preset fields) and
also pre-validate data before pushing it into the database (like URI
or socket type validation).
On 30 May 2013 21:09, Michael Leuker mich...@leuker.me wrote:
Thank you so much for pointing me in the right direction! It was the
missing alias. Now the extensions are working, but there's one more
problem: When I call the PBX (echo-test) or from one extension to another,
I get a hangup
On 3 June 2013 10:46, Ravindra Gowda ravin...@thrikasa.in wrote:
Dear All,
I am trying to integrate kamailio 4.0.1 server with Asterisk 11.4.0 in
Ubuntu 12.04 LTS version. I tried it by using the following link :
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
Hi!
Is there a simple way to:
- retrieve the RR headers (or Route/Contact ...) and to
- selectively remove them?
E.g. retrieve 2 and delete 3
I do not want to care about the format, eg:
Record-Route: 1, 2, 3
or
Record-Route: 1
Record-Route: 2
Record-Route: 3
Thanks
Klaus
On Monday 03 June 2013 20:54:41 Daniel-Constantin Mierla wrote:
can you try the patch from the commit linked next?
-
http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=a17a32e
5f7a3120c200d6e48fe91d7aa1dfd28b1
If all works fine, then I will backport.
Impressions from 45m
On Tuesday 04 June 2013 12:07:35 hiro wrote:
Sometimes it also seemed that kamailio was sending the INVITE to the
phone instead of to freeswitch, or when i played around between
changing $du or $ru the INVITE gets sends to freeswitch but with the
wrong URI pointing to the phone instead of
Finally I found it myself. For the records:
To address a certain header (regardless if headers are in a single line
or in separate lines) use the @hf_value select. The trick is to load the
textopsx module (this select used to be in ser's textops module). Note,
header names must use '_'
ok. right now from tcpdump I can see the session progress and OK
messages are sent to the correct ip:port of my phone, but either the
phone doesn't receive it or it doesn't process it.
I assume the problem to be the headers sent by freeswitch, and perhaps
not changed appropriately by kamailio,
actually, I now see my last message is wrong.
I've compared the 200 OKs that gets sent from freeswitch to my phone
after the busy error and after direct voicemail routing from LOCATION
when user is offline. Both 200 OKs look the same with one exception:
The one that doesn't work had nat=yes two
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