hi everyone,
Actually i'm new in OpenXcap, i have installed it with a mysql database, and
it's runnind fine, but when i test with test_auth.py ... it displaying 8
failures ...
FAILED (failures=8)
devoteam@user-desktop:/usr/share/pyshared/xcap/test$ ./test_auth.py
test_global_auth
I've moved to 1.6.4 and there is no more leak.
Hi,
I am using opensips 1.6.2.
I have modified the opensips.cfg script and it generates a memory leak.
The memory leak seems to be on pkmem since when I use the following
command
'opensipsctl fifo get_statistics pkmem: shmem: | grep
A dummy question...
Every time i restart OpenSIPS i have to reset the phones so they can send
the REGISTER to be able to register again the Proxy...is this a normal
behavior? per the RFC it seems to be OK, because the UAC is 100% responsible
for sending the REGISTER (obviously...) but it's weird
Hi,
Depending how you use db_mode for usrloc:
http://www.opensips.org/html/docs/modules/1.6.x/usrloc.html#id292952
If you user db_mode 0, then you have to restart the phone every time you
restart the opensips.
Dani
On 02/04/11 11:40, Toyima Dias wrote:
A dummy question...
Every time i
2011/2/4 Toyima Dias toyim...@gmail.com:
Every time i restart OpenSIPS i have to reset the phones so they can send
the REGISTER to be able to register again the Proxy..
is this a normal behavior?
Not at all.
per the RFC it seems to be OK
It's not ok.
because the UAC is 100% responsible
On 4/2/11 9:51 AM, SRITI Mehdi wrote:
hi everyone,
Actually i'm new in OpenXcap, i have installed it with a mysql database, and
it's runnind fine, but when i test with test_auth.py ... it displaying 8
failures ...
FAILED (failures=8)
devoteam@user-desktop:/usr/share/pyshared/xcap/test$
Hello list,
I'm using:
Solaris 11 x86 (nv-b91)
OpenSIPS 1.6.4 with TLS
...and am logging dialogs with the siptrace module.
The problem is that every time I want to look at the traced messages
in the sip_trace table, I do:
$ psql opensips opensips
opensips= select * from sip_trace
Hi List Thank you very much Bogdan for the very helpful reply! Now seems
I'll need to find ways to increase the limitation of the max number of
allowed TCP connections for Ubuntu - for the VM that opensips runs on
and the host. I'll be very grateful if anyone would like to share their
experience
Hello Iñaki,
2011/2/4 Iñaki Baz Castillo i...@aliax.net
2011/2/4 Toyima Dias toyim...@gmail.com:
Every time i restart OpenSIPS i have to reset the phones so they can send
the REGISTER to be able to register again the Proxy..
is this a normal behavior?
Not at all.
per the RFC it
Hello,
On Feb 4, 2011, at 11:03 AM, Iñaki Baz Castillo wrote:
[...]
And the server is responsible for keeping the registrations and not
loosing them when restarting.
[...]
Just a related question: what about the nat binding ?
Does the nathelper module keep sending the keep-alive message for
Hello Vallimamod,
The best way to keep a user behind nat, is sending OPTIONS:
modparam(nathelper, natping_interval, 30)
modparam(nathelper, ping_nated_only, 0)
modparam(nathelper, sipping_bflag, 7)
modparam(nathelper, sipping_from, sip:pinger@PROXY_IP)
From the book: Building Telephony Systems
Hi Tyler,
Unfortunately it is not so simply as mic problem :) - the recording was
a done from a different location than where I was (so, across the wild
internet)
I will try to re-register the webinar, just to have a good audio.
Regards,
Bogdan
Tyler Merritt wrote:
Dave,
The audio on
Hi Max,
Considering you still wan to use the dr_rules table (for the mapping of
users to groups), I would manually do the query (with avp_db_query() )
from the script (to get the group id) and only for certain values of the
group ID I will do do_routing(group_id)
Regards,
Bogdan
Max
Hi,
Thanks for advice, i already did this for something similar, so i will
probably do another avp_db_query for this. I thought there was another
way, but guess it is fine this way.
Best Regards
Max M.
Am 04.02.2011 13:51, schrieb Bogdan-Andrei Iancu:
Hi Max,
Considering you still wan
Maybe we should deprecate the flag, to avoid confusion in the future
and keep the code simpler.
Regards,
Ovidiu Sas
On Fri, Feb 4, 2011 at 12:32 AM, Bogdan-Andrei Iancu
bog...@opensips.org wrote:
Hi Ronald,
there is no problem with using the flag and create_dilalog() in the same
time - the
Hi Brian,
opensips siptrace module inserts the message as text blob. My guess is
that your issue is psql related (as in mysql you do not have this
problem) - maybe the select function prints the BLOBs in hexaI do
not think it is a hexa storage, but a hexa display.
Regards,
Bogdan
I tried this:
remove_hf(Record-Route, Server, Supported);
And it blows up with this:
Feb 4 22:11:42 [32421] DBG:core:find_cmd_export_t: remove_hf not found
Feb 4 22:11:42 [32421] CRITICAL:core:yyerror: parse error in config file,
line 612, column 23-24: unknown command remove_hf, missing
Hi,
We've got three parties for this example: A, B, C
A - Asterisk end-point Polycom
B - Asterisk end-point Polycom
C - Outside end-point Uniden
OpenSIPs sits in front of the Asterisk servers and communicates with a
carrier C5 switch directly (same local area network inside a lab facility)
Greetings,
I've been working with a provider on a Kamailio-based project for some
time. We really need to be able to switch traffic to an outgoing SIP
provider (who does not require more than IP-based authorization). Using
the PDT module appears to be the preferred way of doing this. Could
Hi James,
it is a know issue that most of UAC do not properly implement 3xx for
REGISTER requests... Talk to pjsip guys, they are really good and fast
in fixing their stack.
About other solutionscomplicated ones...If P1 receives the REGISTERs
it needs to deal with the NAT issues and act
Hi Tyler.
remove_hf() does not accept multiple headers. What would be the issue
with listing multiple remove_hf() for each header you need to take out?
Regards,
Bogdan
Tyler Merritt wrote:
I tried this:
remove_hf(Record-Route, Server, Supported);
And it blows up with this:
Feb 4
Hi Quinn,
As your question is Kamilio related, maybe you would get appropriate
answers if from its mailing list. We are dealing with OpenSIPS here, a
different project.
Thanks and Regards,
Bogdan
Quinn Ebert wrote:
Greetings,
I've been working with a provider on a Kamailio-based project
Hi Ovidiu,
I agree on that - let's first add a deprecated warning when using the
flag, to force people (while still compatible) to drop flag usage in
favour for the create_dialog()
Regards,
Bogdan
Ovidiu Sas wrote:
Maybe we should deprecate the flag, to avoid confusion in the future
and
Hi Tyler,
So ngrep-ing on proxy, you do not see the second re-INVITE (which leads
to one way audio)A possibility is that the re-INVITE may by-pass
your opensips. Do you do record-routing also for sequential requests ?
There are some bogus UAC/UAS that continuously update the route set,
What did you mean by the document validation ? the credentials of test account
in ~/.xcapclient.ini ?
And another question, what's the difference between test_global and thes_users
??
Thank you :)
No issue at all - that's what I've been doing. It's just a selfish less
code is better motive for posting the question.
It works fine as I have it configured ~ :)
On Fri, Feb 4, 2011 at 10:21 PM, Bogdan-Andrei Iancu bog...@opensips.orgwrote:
Hi Tyler.
remove_hf() does not accept multiple
On 4/2/11 2:39 PM, SRITI Mehdi wrote:
What did you mean by the document validation ? the credentials of test account
in ~/.xcapclient.ini ?
That's an option in OpenXCAP configuration.
And another question, what's the difference between test_global and thes_users
??
Test fetch a global
I had a similar issue in a project, and what I did it to simulate the
list by using M4 macro (to expand the list of remove_hf() from a list of
hdr names). It works like a charm ;)
Regards,
Bogdan
Tyler Merritt wrote:
No issue at all - that's what I've been doing. It's just a selfish
less
Hi Bogdan,
The UAC - P1 - P2 scenario was something I was hoping to avoid
because it puts a reliance on P1 being up, even though its really not
needed.
I've been trying to play with DNS SRV, and returning some sort of
failure code so that a UAC would at least move to the next server with
the same
Hi Bogdan, you changed the mic once when i asked you, after the audio
was perfect. Do you remember ?
2011/2/4 Bogdan-Andrei Iancu bog...@opensips.org:
Hi Tyler,
Unfortunately it is not so simply as mic problem :) - the recording was a
done from a different location than where I was (so,
Hello...
2011/2/5 Robin Malhotra rocky...@gmail.com
When I enter
opensipsctl start
INFO: Starting OpenSIPS :
ERROR: PID file /var/run/opensips.pid does not exist -- OpenSIPS start
failed
Did you copy the /opensips_x/packaging/debian/opensips.init to /etc/init.d/
? and restart with
Hello - We have an Opensips 1.4 server that routes incoming calls to a
couple of different Asterisk servers and to upstream providers. All
working great and with the current config, the Opensips server only
handles the SIP traffic - all of the audio is between the UAs and
Asterisk servers.
Am
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