[OpenSIPS-Users] no ringback

[Bogdan-Andrei Iancu]

Hi Jinsong,

As per RFC3261, a single 180 Ringing is enough to switch the caller 
device to ringing state and keep it so until other higher reply code is 
received.

Can you check at signalling level if something else (other reply) is 
sent after the 180 ? Maybe there is something else in there.

Regards,
Bogdan

Jinsong Hu wrote:
> Hi, Bogdan:
>  Yes, I tried that . The phone rings once , and then keeps silent. I 
> tried to put more
> 180 there, but the phone still only rings once.
>  If there is a solution that sends 180 periodically at 2 to 3 seconds 
> interval until the
> callee answers, probably then it will work, but is there anyway to get 
> this done ?
>
> Jinsong
>
>
> ----- Original Message ----- From: "Bogdan-Andrei Iancu" 
> <bogdan at voice-system.ro>
> To: "Jinsong Hu" <jinsong_hu at hotmail.com>
> Cc: <users at lists.opensips.org>
> Sent: Thursday, August 13, 2009 2:31 AM
> Subject: Re: [OpenSIPS-Users] no ringback
>
>
>> 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
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>
>>
>
>