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See https://code.google.com/p/sipservlets/issues/detail?id=270
Log Time
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I have earlier used MMS 3.0 and "SIPME Media server" together with a B2BUA SIP Servlet application. To retrieve the "local" side SDP, MGCP LocalConnectionOptions is set to TypeOfNetwork.Local.
This…
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Hi Dave,
I investigated a "caller couldn't hear me" report from two internal users.
The call was being processed via B2BUA with the simringer.
In the trace - which I can send - you can see th…
elbow updated
4 years ago
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We want to implement add call and conference in our app
Currently we are using this plugin for audio call with SIP protocol using this plugin. now we going to integrate conference call but not able …
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In SipDialogController::notifyTerminateStaleDialog when calling nta_outgoing_tcreate the dialog routeUri is not used and instead null is just passed in.
I was able to correct it by setting the routeU…
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We're looking for brave testers to validate this tool and its approach.
#### Requirements
All you need is a *HOMER* setup alongside one of the listed platforms with a TLS socket configured and HEP…
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I use the uac_auth module in a B2BUA topology in OpenSIPS 1.11.4.
All is working as expected, INVITES are authenticated but now a new device also challenges the BYE requests sent by OpenSIPS.
Those…
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Scenario:
1. Application is B2BUA
2. Call to some test@example.com to server that supports forking
3. Server returns 2 forked responses
4. My application receives 2 responses one in usual session and …
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Hi there,
I see multiple issues - in makeRequest method, there is no argument for message body, yet it adds a Content-Length, but without a Content-Type for multiple modules.
Did you ever valida…
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In previous versions (0.15.XX) refer to target (blind transfer) or session (attended transfer) terminates (or send termination event?) current session.
Now the session is left idle for a minute unles…