[OpenSIPS-Users] no ringback

[Bogdan-Andrei Iancu]

Hi Jimmy ,

There is a simple thing you can do:

 - just before relaying the INVITE the Asterisk, from OpenSIPS cfg, to a 
sl_send_reply("180","ringing"); to fire a local 180 - of course this is 
a bit bogus from logical perspective (as the end party does not actually 
ring, so you force some information that you cannot check).

Regards,
Bogdan

Jinsong Hu wrote:
> Hi, There:
>   I am using opensips/kamailio in front of asterisk pool. my user register 
> on the opensips, and pstn call are routed out via asterisk.  what I find out 
> is that when the caller calls callee, some of the UA doesn't generate ring 
> back. for example, if I use xlite, the ring back works fine. but if I use 
> sipura 3000,
> I don't hear anything until the callee picks up phone.
>   I did a debug and found that after INVITE, I get 200 back, and then the UA 
> sends out ACK. the callee never sends 180 or 183 back to the caller UA. so 
> before the callee pick up phone, all the caller can hear is just silence.
>
>   if my user registers directly on the asterisk, he can hear the ringback 
> because the Dial() command by default
> will send ring back to the UA.
>
>   How do I solve this problem in this case ? I searched all over internet 
> and don't see any body having any solution.
>
> Jimmy 
>
>
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