Hi ,
I have changed my instance.cfg like bellow .
[10.0.0.1]
prompt_name: opensips-cli@10.0.0.1
communication_type: http
url: http://10.0.0.1:/json
With this config above I am getting some error . I am not able to switch to
the remote instance as well .
https://pastebin.com/1fDuHbBk
if I
Hi,
yes, you are right. The 404 does not come from the module itself. It is
triggered by the routing logic script as the b2b-related functions are not
triggered:
For the second REFER we have the following message in the log:
No dialog found, callid= [123@1.2.3.4], method=REFER
(from
Hi Carsten,
"404 not here" can't be generated by the b2b module itself, it means that
"404 not here" came from another part, I think from part C.
Best way to check this is to capture sip traffic.
--
Nick
пн, 29 мар. 2021 г. в 22:08, Carsten Bock :
> Hi,
>
> I have a question. I am trying to
Hi,
I have a question. I am trying to handle REFER requests as per RFC 4579 in
OpenSIPS B2BUA Module.
User-A wants to initiate a conference with User-B and User-C.
step 1: Create Conference:
=> INVITE sip:conference@conference-url
<= 200 OK
=> ACK
--
This INVITE is forwarded to FreeSwitch to
Hello Giovanni,
I updated the post! There's no need to create the transaction at the
beginning of the route. Simply enabling the dialog tracing takes care
of the outgoing ACK.
Thanks a lot for your valuable feedback,
Ovidiu
On Wed, Mar 24, 2021 at 10:31 AM Giovanni Maruzzelli wrote:
>
> Ciao
Hi Stas,
thanks for that. I'll try it out
cheers,
Mark
On Mon, 29 Mar 2021 at 15:51, Stas Kobzar wrote:
> Hello Mark,
>
> IMO, it is Asterisk side. Of course it depends on your setup but probably
> you need Asterisk sip peer to opensips. Do not know for pjsip, for older
> sip_chan it would
Hello Mark,
IMO, it is Asterisk side. Of course it depends on your setup but probably
you need Asterisk sip peer to opensips. Do not know for pjsip, for older
sip_chan it would be something like:
[opensips]
type=friend
deny=0.0.0.0/0.0.0.0
permit=OPENSIPS_IP/255.255.255.255
host=OPENSIPS_IP
You
We have a DID. If an incoming INVITE goes via OpenSIPS, Asterisk returns
'401 Unauthorized' requesting authorization credentials. If we map the DID
direct to Asterisk it doesn't ask for authorization. Our setup is...
DID ---> OpenSIPS 3.1 Mid_registrar ---> Asterisk (FreePBX)
Is there something
Hi,
Thanks - yes I had realised that but it was the string manipulation of
the header itself that I was mainly thinking about.
I'm thinking that using a regex match on what is between "sip:" and "@"
would hopefully be OK with no false matches anywhere else in the
string, but there is no
Hi, Kingsley!
The only way to do this is to remove the header and add a new one using
remove_hf/append_hf functions.
Best regards,
Răzvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 3/26/21 6:09 PM, Kingsley Tart wrote:
Hi,
I'm using OpenSIPS 3.1.
I've set up
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