Hi Yagishita,
The error is not related to STUN - I see you configured 127.0.0.1:5060
as a TCP listening interface, but it seams other application is already
using it.
Regards,
Bogdan
Koichi Yagishita wrote:
Hi Bogdan,
Thank you for your response. The following is output of
Hi Nir,
the last command does create (if not present) or adds to (if already
present) the current CA to the CA list file.
Also, have you properly set the TLS related parameters in the config file?
Regards,
Bogdan
nir elkayam wrote:
hi,
i follow the script on :
Hi Alan,
Normally (using only sip_trace fct), the tracing can be activated by
flags (no username in sip_trace table) or by avp_traced_user (with name
in sip_trace table) - of course, using them both will generate multiple
records for the same SIP message.
But looking at the trace_dialog()
Hi Wesley,
if you set debug = 4, you will get a all the debug messages from the
module. It will give you some hints if at least is matching any rule.
But what I found strange is that tat the replt_exp field is empty -
that is the part to be returned .
Regards,
Bogdan
Wesley Volcov wrote:
Hi Mike,
All replied do get first in onreply_route (provisional, success or
failure). After that, only the negative replies trigger the failure route.
See : http://www.opensips.org/Resources/DocsCoreRoutes16
Regards,
Bogdan
Mike O'Connor wrote:
Hi All
Is there any way that I could have
Hi Nathaniel,
What version of opensips are you trying and what OS you are running on ?
Regards,
Bogdan
Nathaniel L Keeling wrote:
Hello,
I am trying to load the seas module and I am getting this error:
ERROR:core:sr_load_module: could not open module
hi,
attached the lines from the cfg file:
r...@:/usr/local/etc/opensips# cat opensips.cfg | grep tls
disable_tls = no
listen = tls:X.X.X.X:30100
tls_port_no = 30100
tls_verify_server = 0
tls_verify_client = 0
tls_require_client_certificate = 0
tls_method = TLSv1
tls_certificate =
Hi Jeff,
Jeff Pyle wrote:
Iñaki,
On Jan 9, 2010, at 5:00 PM, Iñaki Baz Castillo wrote:
El Sábado, 9 de Enero de 2010, Jeff Pyle escribió:
Hello,
The docs say that when using the b flag with lookup() when multiple
records are present, it will load only the one with the highest
Hi Yagishita,
If different than 5060 (default) the port is required in SIP. BTW, the
call to UA2 is done via lookup(location) ? if so, check via the
opensipsctl ul show UA2_AOR the contacts the UA2 has registered with
opensips.
Regards,
Bogdan
Koichi Yagishita wrote:
Thank you very much
Hi Mike,
Could describe in more details the call flow you have there ? I do not
understand your scenario here.
Regards,
Bogdan
Mike O'Connor wrote:
Hi All
Anytime I forward a call to the same instance of opensips the CPE
which was initially rung will re-ring, if I sent the call directly
Hi Ashwini,
Enable full logging (debug=4) and see if the BYE requests to match the
dialog (or post there the logging corresponding to BYE processing).
Regards,
Bogdan
ASHWINI NAIDU wrote:
Hi Bogdan,
The calls are terminated by the users. when i check the dlg_list
MI command i see the
Hi,
you do not need any loop - just set as key for profiling the DID
number and add to that profile the calls related to that DID.
Regards,
Bogdan
Johnson Pajayat wrote:
Hi Bogdan,
I was able to implement the channel limiting on one DID by using a
variable instead of AVP and replacing all
The f flag sounds fantastic. Thanks.
- Jeff
On Jan 18, 2010, at 9:24 AM, Bogdan-Andrei Iancu wrote:
Hi Jeff,
Jeff Pyle wrote:
Iñaki,
On Jan 9, 2010, at 5:00 PM, Iñaki Baz Castillo wrote:
El Sábado, 9 de Enero de 2010, Jeff Pyle escribió:
Hello,
The docs say that when using
yes, with same user-name and passwd i am able to login from opensips box
to remote mysql server.
On Mon, Jan 18, 2010 at 11:41 AM, ram-2-3 [via OpenSIPS (Open SIP Server)]
ml-node+4412061-17454...@n2.nabble.comml-node%2b4412061-17454...@n2.nabble.com
wrote:
are you able to login to remote
Hello Bogdan!
I think you could not see the repl_exp value because the line break, this
value is my email address (I'm using nabble.com and hiden the email).
About debud level, I'm already using debug = 4, but It's not working anyway.
I tested with debug =9, but the log appears the same.
When I
Bogdan,
Thanks for the info. I load the RPID with the modparam(auth_db,
load_credentials, rpid) and put it into $avp(s:rpid).
As long as OpenSIPS is in forked mode, it works fine. But when I was
running it in non-forked mode is when I saw the retention behavior.
Seems the RPID would stick when
Hi Wesley,
if you deleted the whole table, it is impossible to get that error (with
no rules to load). Check if you are loading form the right server/DB/table.
Regards,
Bogdan
Wesley Volcov wrote:
Hello Bogdan!
I think you could not see the repl_exp value because the line break, this
value
Hi List,
When a user hangs up a call (call comes into proxy, connects to PSTN) and if
the user that made the call hangups before a certain amount of time I want
to delay sending the BYE to the upstream carrier, but ACK the BYE to the
person they called and then have acc show the correct call
Ron,
Are you trying to avoid short-call charges from your carrier? It's not easy.
Even if this were possible, it wouldn't help if the far-end were to hang up
first. Even if they don't hang up first, they're likely going to hang up
during this 12-second window you're looking to create in
Jeff,
Yes, that's the goal anyways :)
I guess in my mind I thought if I could delay the BYE from going to the
upstream BUT send the BYE to the customer / ACK the BYE they sent then the
end user has no ideal what's going on and we just leave the channel open for
5 to 11 seconds and then send the
Ron,
No, I don't believe the theory is not correct.
I'm going to think you have a customer that gets hung up on a lot, generating
short-call surcharges from your carriers. You want to delay the BYE you send
to the carrier... except that's not what triggers the call disconnect. The
I am in the process of putting all the OpenSIPS modules and AG Projects
together to create a carrier-grade service and think this is something that
can be used. I am far from implementing what you speak of above, but it
would be very helpful. Nothing is better then saving money but not at the
Hi Bogdan,
Thank you very much for the response. This issue has been solved.
Regards,
Yagishita
Hi Yagishita,
The error is not related to STUN - I see you configured 127.0.0.1:5060
as a TCP listening interface, but it seams other application is already
using it.
Regards,
Bogdan
Hi Bodan,
Thank you very much for the response. This issue has been solved by upgrading
OpenSIPS from 1.5.0 to 1.5.3.
Regards,
Yagishita
Hi Yagishita,
If different than 5060 (default) the port is required in SIP. BTW, the
call to UA2 is done via lookup(location) ? if so, check via the
That would be a nice topic... We are hoping and thankful to hear from you
soon...
-
http://opensips.blogspot.com http://opensips.blogspot.com
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Sent from the OpenSIPS - Users mailing
Hi all
i debug on mediaproxy and see, mediaproxy create 4 ports but the real ports to
relay media are only 2 ports: 118.69.239.140:50012 - 118.69.239.140:50014
is it normal, can i config mediaproxy create only 2 ports
Thank you
Ha`
mediaproxy.mediacontrol.StreamListenerProtocol starting on
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