Hi Mark, is working for us more or less, because is strange situtation, if
the extension WSS received a incoming call working good but if a put the
call in on-hold inmediatly hangout, but if the same extension do the call to
a external number or a SIP extension or another Webrtc extension work
Hi Juan Carlos - I feel your pain!
I've finished for the year so I don't have access to the code at the
moment. I'll have a look at this in the new year to see if I can post
something sensible! :-)
Short version is that it turned out to be a horrible challenge to get it
working as desired. My
Hi, iam having the same situation, can you copy the opensips.cfg as
reference, iam having issue with incoming calls to webrtc extensions, my
opensips version is 3.1 and is working with mid_registar, i can do calls to
SIP extensions, SIP Trunk, but not working when incoming calls to extension
Hi Mark, can you share the configuration? i need to do something similar but
without success, thank you.
Using your example:
REGISTER: SIP softphone (LinPhone) -> OpenSIPS Mid-registrar -> Asterisk
= success
REGISTER: WebRTC webphone (Mizutech) -> OpenSIPS Mid-registrar -> Asterisk
Hi Mark,
Glad it helped. I actually did not know about chan_sip option
supportpath because I was using additional opensips for wws to webrtc.
Thanks for the details.
Good luck
On Thu, Jul 23, 2020 at 9:19 AM Mark Allen wrote:
>
> [SOLVED]
>
> Thanks Stas. Your workaround did solve the problem
[SOLVED]
Thanks Stas. Your workaround did solve the problem and I see that with 3.1
path support is baked into mid-registrar module as
options to mid_registrar_save().
Once we added in the path module functionality, at first it didn't work.
Looking at sngrep traces we could see that the path
Hi Alexey - thanks for responding.
I've seen past reports where NAT was causing this type of problem. I tried
your suggestion but, along with other tests such as forcing
fix_nated_register() or fix_nated_contact() on all messages, and after
trying Stas' suggestion, it still doesn't work for me. I
Hi Mark,
try this:
if (nat_uac_test("123")) {
if (is_method("REGISTER")) {
fix_nated_register();
} else {
fix_nated_contact();
}
}
-
---
Alexey Vasilyev
--
Sent from:
I see, Mark. It is true, in my case, I splitted webrtc to other
opensips (newer version) as our platform was too old.
I still think path module function should help:
https://opensips.org/docs/modules/3.1.x/path.html#func_add_path_received
Good luck
On Tue, Jul 14, 2020 at 11:48 AM Mark Allen
Thanks Stas - I'll have a look at that.
For clarification, we only have one OpenSIPS server acting as
mid-registrar. Endpoints register through it to extensions on Asterisk, and
Asterisk acts as B2BUA for calls from one extension to another. We've got a
lot of additional functionality linked to
Hello Mark,
I had a similar challenge. Using "path" module on both opensips helps
to overcome this problem.
https://opensips.org/docs/modules/3.2.x/path.html
In your mid-registerer you need to enable path support. See "save"
function params p0 and v.
in your webrtc opensips use path module and
I'm new to OpenSIPS and I've hit a problem I can't find a way past
We have a test setup with an OpenSIPS mid-registrar in front of an Asterisk
PBX. Mid-registrar is currently in mode 1 (registration throttling). We
have SIP and WebRTC endpoints that we want to use.
*Current state is:*
REGISTER:
12 matches
Mail list logo
