Yes, have shared the complete traces in this thread itself.
Thanks.
Regards
Kashish
On Thu, May 27, 2021 at 6:44 PM Kashish Raheja
wrote:
> Haven't been able to sort this out yet. Anything am I missing here?
>
> Thanks.
> Regards
> Kashish
>
> On Fri, May 21, 2021
Haven't been able to sort this out yet. Anything am I missing here?
Thanks.
Regards
Kashish
On Fri, May 21, 2021 at 1:44 AM Kashish Raheja
wrote:
> Hi Daniel,
>
> Sorry it took some time for me to make these changes.
>
> I have made all the changes as suggested by you howe
Hi Daniel,
Sorry it took some time for me to make these changes.
I have made all the changes as suggested by you however it still doesn't
seem to work. No audio in the outbound call however incoming call works
fine.
Here are the SIP traces after making the changes:
*INVITE: Asterisk to Kamailio
│
20:24:29.914667 │ │
│ 200 OK│
│ │
│ ──> │
Thanks.
Regards
Kashish
On Mon, May 10, 2021 at 8:26 PM Kashish Raheja
wrote:
>
890 IN IP4 10.0.X.X
s=SBC call
c=IN IP4 10.0.X.X
t=0 0
m=audio 37874 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:8 PCMA/8000/1
Regards
Kashish
On Mon, May 10, 2021 at 2:37 PM Kashish Raheja
wrote:
> Hi All,
>
> I have set up Kamailio in the following manne
Hi All,
I have set up Kamailio in the following manner:
Kamailio (Physical Server: Register to Telecom Operator Carrier SIP trunk)
---> Asterisk Server (on Cloud having public IP)
I am successfully able to route the call to Asterisk server on Cloud when I
make a call to the number provided by th
