[Spce-user] Call forwarding, no RTP

[Alessandro Bolletta]

No, i'm not using Digitel. Also I don't think that we are talking about the same issue.


Da: stefanormc <stefanormc at gmail.com>
Inviato: 19/gen/2016 12:36
A: Marco Teixeira
Cc: Alessandro Bolletta; spce-user at lists.sipwise.com
Oggetto: Re: [Spce-user] Call forwarding, no RTP

I’m experiencing the same problem. I have 2 providers: one works fine the other not. Sipwise setting are the same.

The non working provider is Digitel; is that the same of yours, Alessandro?

Apparently is a RFC compliance issue

stefano

On 19 Jan 2016, at 12:34, Marco Teixeira <admin at marcoteixeira.com<mailto:admin at marcoteixeira.com>> wrote:

So i think my initial assumption "(...) If you are allowing direct RTP between endpoints (...)" does not verify then.
My next step would be to take a SIP trace and check what's going on...


---
​B​
est regards
Marco
---


On Tue, Jan 19, 2016 at 11:24 AM, Alessandro Bolletta <alessandro at mediaspot.net<mailto:alessandro at mediaspot.net>> wrote:
That setting was already enabled.

Alessandro

Da: Marco Teixeira [mailto:admin at marcoteixeira.com<mailto:admin at marcoteixeira.com>]
Inviato: martedì 19 gennaio 2016 12.12

A: Alessandro Bolletta <alessandro at mediaspot.net<mailto:alessandro at mediaspot.net>>
Cc: spce-user at lists.sipwise.com<mailto:spce-user at lists.sipwise.com>
Oggetto: Re: [Spce-user] Call forwarding, no RTP


Goto your peering server for that "Ext number"
Click Preferences
and set use_rtpproxy to always
I think that should do make the RTP for calls to that peer to always proxy via SPCE
<image001.png>


---
Best regards
Marco
---


On Tue, Jan 19, 2016 at 10:37 AM, Alessandro Bolletta <alessandro at mediaspot.net<mailto:alessandro at mediaspot.net>> wrote:
Yes, the “Ext number” gateway refuses RTP from that endpoint for sure.

Could Sipwise be set in order to proxy that RTP flow?

Da: Marco Teixeira [mailto:admin at marcoteixeira.com<mailto:admin at marcoteixeira.com>]
Inviato: martedì 19 gennaio 2016 10.51
A: Alessandro Bolletta <alessandro at mediaspot.net<mailto:alessandro at mediaspot.net>>
Cc: spce-user at lists.sipwise.com<mailto:spce-user at lists.sipwise.com>
Oggetto: Re: [Spce-user] Call forwarding, no RTP

If you are allowing direct RTP between endpoints, your "external number" gateway/provider might be refusing RTP from the original endpoint and expecting audio to come from your SPCE.


---
Best regards
Marco
---


On Mon, Jan 18, 2016 at 4:09 PM, Alessandro Bolletta <alessandro at mediaspot.net<mailto:alessandro at mediaspot.net>> wrote:
Hi,

if I configure a subscriber to unconditionally forward any incoming call to an external number, get the forwarded call estabilished but I get no RTP flows on both sides.

The peer used for the call forwarding is already working flawlessly for other purposes, so I can imagine that there’s something with call forwarding’s procedure.

Hoping to get your help,
Alessandro Bolletta

_______________________________________________
Spce-user mailing list
Spce-user at lists.sipwise.com<mailto:Spce-user at lists.sipwise.com>
https://lists.sipwise.com/listinfo/spce-user



_______________________________________________
Spce-user mailing list
Spce-user at lists.sipwise.com<mailto:Spce-user at lists.sipwise.com>
https://lists.sipwise.com/listinfo/spce-user

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20160119/269c70b9/attachment-0001.html>