[Blink] SIP2SIP account won't work with Cisco

[Michael Nagie]

Dear Tijmen de Mes,

Can I ask you one more thing?
I've been struggling with encryption for a while now. How did you set it 
up?

If I choose:
Voice > Sip > SRTP Method : s-descriptor
Voice > Phone > Secure Call Serv : yes
Voice > User > Secure Call Setting: yes

Then I can initiate encrypted calls if the recipient supports it.
For example the 1233 Voice Mail does not support it so I can't call it.
It says: Not acceptable here. It's better to say; I can only initiate 
encrypted calls.

If I'm not mistaken if I chose x-sipura then I would need a mini 
certificate from the SIP2SIP server. I didn't find it anywhere, though.
X-sipura is said to be more secure. However I'm not sure that it would 
solve the fallback to RTP problem...

Thank you
Mike


On 17-01-19 18:23:44, Tijmen de Mes wrote:
> Hi,
> 
> If you use sip2sip.info as proxy setting, you should make sure that it uses DNS SRV.
> 
> In a cisco 303 I have this setting in the advanced mode in the section 'proxy and registration'.
> DNS SRV Auto Prefix and Use OB Proxy In Dialog are also enabled.
> 
> The sip port you configured is the local port of the Device. It is not the server port. You should leave it on
> 5060 which is the default.
> 
> Best regards,
> 
> Tijmen de Mes
> —
> AG Projects
> 
> > On 19 jan. 2017, at 13:55, Michael Nagie <promike1987 at gmail.com> wrote:
> > 
> > Hello,
> > 
> > I stuck and I need some help. I bought a Cisco SPA504G in the end and
> > I've been configuring SIP2SIP for about 5 hours but I still can't figure
> > out what is wrong.
> > 
> > I tried to follow the description:
> > http://sip2sip.info/settings/
> > 
> > But I'm failed.
> > 
> > My settings right now are:
> > 
> > Line Enable: yes
> > Sip Transport: TLS
> > Sip Port: 443
> > Proxy: sip2sip.info
> > Register Expires: 3600
> > Display name, User ID and Auth ID are my username without @sip2sip.info.
> > Password is my passwd.
> > And currently the prefered codec is g711u, but I don't think that really
> > matters.
> > 
> > So I get:
> > Line SIP2SIP
> > Failed (Not Reachable)
> > 
> > If I change the settings to SIP Transport to UDP 5060 then my line
> > button blinks:
> > Not Registered(?)
> > After a while 'Failed (No response)' then it changes to
> > 'Not Registered (No response)'
> > 
> > Thanks in advance!
> > _______________________________________________
> > Blink mailing list
> > Blink at lists.ag-projects.com
> > http://lists.ag-projects.com/mailman/listinfo/blink
> 



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