7;s Topics:
>
> 1. Re: Message Compression Feedback (Ionut Ionita)
> 2. Re: topology hiding not accepting BYE before 200 OK
> (Stuart Marsden)
> 3. 500 command 'dr_gw_status' failed (Aqs
> On 5 Oct 2015, at 17:47, Stuart Marsden wrote:
>
> Our actual case is
>
> phone A-> opensips -> our soft switch -> same opensips -> phone B
>
> INVITE —>
> <—RINGING
> 200 ——
Hi
we are experimenting with topology hiding on 2.1
I think we see the same issue once a call is set up if UAC and UAS both send
BYE at “the same time”
we cannot reproduce at will because of the small timing window required to
receive the 2 BYEs
Stuart
Hi
sorry if being thick, but I cant see what is wrong
what I actually want to do is check if the route (in this case loose
routed) is to me , ie one of my "listens"
Stuart
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Hi,
There is probably an obvious answer, so apologies in advance.
I have built a simple script for opensips which just proxies sip messages
(calls only) between two endpoints.
The only processing is that it strips out and adds a few sip headers.
I now need to keep one side registered (every
ing.
Regards,
Bogdan
Stuart Marsden wrote:
> Hi,
>
> I was interested in being able to make/ receive calls via Skype, but the
> phones etc will still have our voip numbers.
> You have to register with the Skype service using their credentials,
> hence the dual nature
Hi,
I was interested in being able to make/ receive calls via Skype, but the
phones etc will still have our voip numbers.
You have to register with the Skype service using their credentials,
hence the dual nature of the subscriptions
Stuart
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Use
Hi,
Has anyone had a go at building one of these?
Stuart
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ou can
make this process secure by restricting this operation to that subnet
Stuart
Ruud Klaver wrote:
> Hi,
>
> On 30 Jun 2009, at 19:42, Stuart Marsden wrote:
>
>>
>> The problem is the Sipura/Linksys/Cisco Phones do it the other way
>> round - the phone going
sorry - email went as html
Stuart Marsden wrote:
>> 3) detect this special case and change existing conntrack to point to
>> the MoH sever, putting it back later
> Ruud Klaver wrote:
>
> How would you detect changes at SIP level in this case? The reason I'm
> ask
Hi,
The problem we have, is the implementation of MoH to Invite the MoH
server and pass the SDP of an existing call is fixed in the
implementation of the SPA 94X phone - we can't change that.
In fact at the moment we struggle to get Linksys to fix bugs in the
Phones, never mind change the way t
ud Klaver wrote:
Hi Stuart,
On 30 Jun 2009, at 14:41, Stuart Marsden wrote:
Hi,
I am testing a mediaproxy sever using 2.3.4
The tests I have done so far have worked, however there is one scenario
where once the call has been setup a phone can generate Music on Ho
Hi,
I am testing a mediaproxy sever using 2.3.4
The tests I have done so far have worked, however there is one scenario
where once the call has been setup a phone can generate Music on Hold
It does this by making a 2nd call (unrelated to the first at the SIP
level) and passes what it thinks is
Hi,
I am just trying mediaproxy 2.3.4 for the first time. All works out the
box (which is impressive)
However when I try the media_sessions_phtml - I get an error in the
media-dispatcher
"Connection to Management interface client lost: A record packet with
illegal version was received"
any
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