On 21.08.2020 18:14, Liviu Chircu wrote:
Let me re-test this feature and come back with an update.
Johan,
I've successfully re-run my tests and both notify_on_event() and
async(wait_for_event()) worked just fine.
I only have one idea that may explain why it doesn't work for you: if
you
On 21.08.2020 16:40, johan wrote:
Hello, using opensips3.1 I don't arrive at catching event
E_UL_CONTACT_INSERT.
As you can see in below logs,
the route seems to be correctly armed, but the event is not triggered
on the registration of the user.
Hey Johan,
Let me re-test this feature
On 20.08.2020 09:37, Darpan Patel wrote:
But in my case after 40 seconds it's not trigger resume_call route, so
resume_call only called after the event will succeed ? I want to implement a
feature like if callee is not registered till 40 seconds then relay call to
PSTN Gateway .thanks alot in
following Volga's advice, I added lookup(location) after the subscribe
but to no avail, that event doesn't want to pop.
xlog("callid=$ci: Route[userlocation]:we call t_newtran and
subscribe for E_UL_CONTACT_INSERT");
# prepare transaction for branch injection; it is
Hello Mark,
In my case I do have a path in the location record. Here is my example from
"ul show" (I changed my real domain and IPs):
AOR:: 9...@example.com
Contact:: sip:suvp4v56@1p6pc0g6m3ml.invalid;transport=ws Q=
Expires:: 494
Hello, using opensips3.1 I don't arrive at catching event
E_UL_CONTACT_INSERT.
As you can see in below logs,
the route seems to be correctly armed, but the event is not triggered on
the registration of the user.
Can somebody give a hint on what I am overlooking ?
Do I need to enable the
What am I looking for?
INVITE from Asterisk to Opensips looks fine. Contact info from "location"
matches that seen in console for web phone.
Problem seems to be that the address is not recognised as a web socket
rather than a host name. It's not NATed but tried fix_nated_register() and
Please check contact header.
volga629
From: "Mark Allen"
To: "OpenSIPS users mailling list"
Sent: Friday, August 21, 2020 8:08:18 AM
Subject: Re: [OpenSIPS-Users] 3.1 - Mid_Registrar - AOR throttling with WebRTC
failing
I've not received any feedback on this regarding whether or not
I've not received any feedback on this regarding whether or not what I'm
doing should be working. Trying to find a workaround has just led to a
number of dead-ends. Can anyone please help me with this?
We are using mid-registrar with AOR Throttling talking to Asterisk/FreePBX.
We have OpenSIPS
On 21.08.2020 05:12, Robert Dyck wrote:
As a learning exercise I wanted to create a new database using
opensips-cli "database create sqlite:///tmp" or "database create
sqlite:///tmp/tmp.db". The response was invariably "ERROR: Bad URL, it
should resemble: sqlite:///path/to/db". Omitting the
Hello volga629,
Thanks to Liviu Chircu, #2161 is fixed.
If you are using tls_mgm in db mode, and the Postgresql connections are limited
to 1, it could be useful for you too.
Thanks for your emails, you helped me to find what to search for the bug report.
Regards,
--
Adrien Martin
Hello, I want to do dynamic configuration for fetching data from mongodb
database and putting it into opensips.
I am using openSIPS version 2.4 .
My script :
loadmodule "cachedb_mongodb.so"
modparam ("cachedb_mongodb",
"cachedb_url","mongodb:instance://localhost:27017/opensipsDB.login")
route{
12 matches
Mail list logo
