[Spce-user] Call forwarding, no RTP

[stefanormc]

if this is still the issue:

> 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.
> 


then yes, we are talking about the same issue.

try pressing any dtmf when the call is established on callee side. in my case rtp starts going through





> On 19 Jan 2016, at 13:00, Alessandro Bolletta <alessandro at mediaspot.net> wrote:
> 
> 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
>> 
>> 
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>>  
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>>  
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