[sr-dev] No Media in SIP Incoming calls

2019-01-08 Thread Prashant Gupta
Hi, I have the following architecture - SIP provider <-> Kamailio <-> Asterisk servers Currently I have everything setup and incoming calls from Sip are routed to my asterisk server. The issue is however that when I answer the call, there is no media in the call. I have tried connecting with a

Re: [sr-dev] Cannot Register To SIP Provider

2019-01-05 Thread Prashant Gupta
boundproxy=4.3.2.1 >> >> That should send all outbound regs and calls via your kamailio. >> >> But if in your case, your provider will use that registration to send YOU >> calls, this will probably not work. You’d indeed need to register from >> kamailio. Something I understand is

Re: [sr-dev] Cannot Register To SIP Provider

2019-01-05 Thread Prashant Gupta
ever been able to do. > On Sat, 5 Jan 2019 at 06:07, Prashant Gupta wrote: > >> Thanks for the reply. Can you please explain how can I achieve this? >> In this scenario, will the incoming calls still be routed through the >> kamailio. What about the outgoing calls? >>

Re: [sr-dev] Cannot Register To SIP Provider

2019-01-04 Thread Prashant Gupta
just forward the registrations > and every asterisk will be registered normally. > > I know it’s not ideal, but I couldn’t make it work, and at least it’s now > working perfectly. > > On Fri, 4 Jan 2019 at 07:51, Prashant Gupta wrote: > >> Hi, >> I have the follo