On Thu, Nov 29, 2012 at 2:42 AM, Jeremy Jongepier wrote:
> On 11/28/2012 05:48 PM, Raj Roy Ghandhi wrote:
>>
>> Dear Peter,
>> Thansk for your fast response. I highly appreciate it.
>> Is there any way that I can convert the RTP/SAVPF into general media
>> profile that PSTN GW support ? So that I
Am 28.11.2012 20:58, schrieb Konstantin M.:
Jeremy, it is doesn't work at all. I've made a lot of changes to that
patched asterisk to make it working and no luck.
However, ast11 has fully supported webrtc, but I heard no voice during
a call.
Same here. I also tried the Doubango patch but it do
On 11/28/2012 08:58 PM, Konstantin M. wrote:
Jeremy, it is doesn't work at all. I've made a lot of changes to that
patched asterisk to make it working and no luck.
However, ast11 has fully supported webrtc, but I heard no voice during a
call.
Another issue is - sipml5 is sending a malformed Conta
Jeremy, it is doesn't work at all. I've made a lot of changes to that
patched asterisk to make it working and no luck.
However, ast11 has fully supported webrtc, but I heard no voice during a
call.
Another issue is - sipml5 is sending a malformed Contact field, and
asterisk is trying to contact to
On 11/28/2012 05:48 PM, Raj Roy Ghandhi wrote:
Dear Peter,
Thansk for your fast response. I highly appreciate it.
Is there any way that I can convert the RTP/SAVPF into general media
profile that PSTN GW support ? So that I can get that call working/
Best Regards,
Roy.
I think at the moment o
You need some sort of media gateway or server. I don't know of any that
currently support this.
Peter
Raj Roy Ghandhi wrote:
>___
>SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>sr-users@lists.sip-router.org
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Dear Peter,
Thansk for your fast response. I highly appreciate it.
Is there any way that I can convert the RTP/SAVPF into general media
profile that PSTN GW support ? So that I can get that call working/
Best Regards,
Roy.
On Wed, Nov 28, 2012 at 8:32 PM, Peter Dunkley <
peter.dunk...@crocodile-r
Unless your PSTN gateway supports the RTP/SAVPF media profile - I don't
know of any that do - this will not work.
Regards,
Peter
> Hi,
> From all of your support now I can call from
> 1. IP Phone <--> IP Phone
> 2. Web Page <--> Web Page
> 3. IP Phone -> PSTN without any issue
>
> But when I try
Hi,
>From all of your support now I can call from
1. IP Phone <--> IP Phone
2. Web Page <--> Web Page
3. IP Phone -> PSTN without any issue
But when I try to call from Web Page to PSTN then it tries to call
sip:00xx89...@mysipdomain.com and that time out.
Trying to figure out how to get this w
