Hi all
I'm using a database with some translation rules in the dialplan table. How
can I reload the updated information in the db without restarting Opensips?
I tried
dp_reload();
dp_translate(1,$rU/$rU);
but dp_reload() in the opensips.cfg
Hi Jayesh,
The dialog topology hiding works at dialog level and you cannot use it
in a per-branch manner. So, you cannot set or not topo hiding for some
branches. Solutions are:
1) simply do topo hiding when routing, so you will do it for all
branches.
2) set a separate opensips
Hi,
dp_reload is not a script function, but a MI (Management Interface)
function - such functions are called from outside opensips.
You can trigger the functions like:
opensipsctl fifo do_reload
Regards,
Bogdan
On 11/23/2011 10:27 AM, wüber wrote:
Hi all
I'm using a database with some
Hi Bogdan
Thank you very much for your answer.
Could you please tell me if this is the only possibility to load the new
translation rules (or in general the new database info) from the database?
Regards,
Carmelo
--
View this message in context:
Hi Carmelo,
This command is particular to dialplan module. On the other hand, most
of the modules that load and cache data from DB do have a similar
command, like Dynamic Routing module has dr_reload, permissions module
has address_reload, nathelper has nh_reload, etc...
You need to check
Sarò assente fino al 25 Novembre compreso. Per urgenze rivolgersi direttamente
ad assiste...@longwave.eu o chiamare lo 0522375500. Saluti
I will be out of office till November 25th 2011.
___
Users mailing list
Users@lists.opensips.org
Hi Bogdan
That's clear. Thanks for your support.
I think I'll try to create a mi datagram interface and execute the dp_reload
command remotely.
Regards, Carmelo
--
View this message in context:
http://opensips-open-sip-server.1449251.n2.nabble.com/dp-reload-tp7023627p7024030.html
Sent from
Bogdan,
I don't think the uac_redirect module in this case is helpful. The Contact
data that comes back from an LRN DIP's 302 isn't a real SIP URI, but rather
just some routing data that happens to be using a 302's Contact field as a
transport mechanism.
Kent,
Sorry for the late reply... I
Hello,
I committed a fix for this in trunk and 1.7 branch, so while using
topology hiding, requests should be properly routed even when the dialog
is in the early state.
Regards,
Vlad Paiu
OpenSIPS Developer
On 11/22/2011 04:18 PM, Saul Ibarra Corretge wrote:
Hi Bogdan,
On Nov 22, 2011,
Hi Jeff,
Well, according to RFC3261, a Contact hdr must carry a valid SIP URI -
now, in dip lookups, the answer is added as params to the SIP URI or to
the CT SIP hdr...depending...
If you uac_redirect does not server your purpose (like answer in in CT
hdr params), you can access the hdr
I'm using Opensips as a Load balancer and as a registrar, so basically
all phones are registered to the Opensips, all Incoming calls hit the
opensips server which forwards the call to asterisk with load
balancing, asterisk decides what to do with the call ie IVR voicemail
etc and if the call needs
Sarò assente fino al 25 Novembre compreso. Per urgenze rivolgersi direttamente
ad assiste...@longwave.eu o chiamare lo 0522375500. Saluti
I will be out of office till November 25th 2011.
___
Users mailing list
Users@lists.opensips.org
Hi all,
*OpenSIPS 1.7.1* - a minor release on the 1.7 main branch - was release
today.
This release contains only bug fixing (additional to 1.7.0) - crashes,
malfunctions and compliance issues - but no functionality, scripting or
interfacing were changed.
*OpenSIPS 1.7.1* contains
Hi Schneur,
What you have to do is to change the way you distribute the call among
the asterisk boxes in such a way that all calls in which a user is
involved to be on the same box (so that the transfers will work).
How to do that? with a mixed routing logic. When you receive a new call, do:
I am use the B2B modules with the topology hiding scenario and
periodically see the following error in my opensips log:
ERROR:b2b_entities:b2b_tm_cback: No dialog found reply 200 for method BYE
I did a capture to find out what was happening and it appears that
after receiving a BYE opensips will
Hi Ryan,
My first guess is that it is something wrong with the 200 OK you receive
for the BYE - could you post somewhere the trace (or send it offline to
me) ?
Regards,
Bogdan
On 11/23/2011 07:52 PM, Ryan Bullock wrote:
I am use the B2B modules with the topology hiding scenario and
Hi,
I'm having problem with hanging dialogs.
Nov 23 20:46:18 V0P034-VoIP-LB /usr/local/sbin/opensips[16945]: New request
- M=INVITE RURI=sip:ora517xxx...@lb-gw.sip.int.ccig.pl
F=sip:519xx@10.0.130.161 T=sip:ora517xxx...@lb-gw.sip.int.ccig.pl
IP=10.0.130.161
Hi Damien,
The behavior that you get is the correct one. If you publish with MI the
presence server will consider as if there is another device publishing for
the same account. So when sending Notify, it will aggregate what you have
sent with what it has received from the phone.
Regards,
Anca
18 matches
Mail list logo