[Users] TLS and asterisk ?

Bogdan-Andrei Iancu bogdan at voice-system.ro
Thu Apr 12 13:42:28 CEST 2007


Bodin,

please add an xlog() to print he RURI just before t_relay():
    xlog("-----ruri = $ru\n");

and let em know what uri you have. I want to see if the protocol is 
forced by a ruri parameter or not,

regards,
bogdan

Bodin Bruno wrote:
> Bogdan-Andrei Iancu a écrit :
>> Hi Bodin,
>>
>> I guess openser forwards the call to asterisk via TLS also - can you 
>> check this?
>>
>> regards,
>> bogdan
>>
>> Bodin Bruno wrote:
>>> Hi,
>>> I know asterisk doesn't support tls but that something strange in 
>>> SIP protocol :
>>>
>>> TLS UA -> Openser -> no tls UA : that work
>>> TLS UA -> Openser -> asterisk : that don't work
>>>
>>> any one know something about this ?
>>>
>>> to use asterisk I make rewritehostport.
>>>
>>> thank for help
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at openser.org
>>> http://openser.org/cgi-bin/mailman/listinfo/users
>>>
>>
> in openser log i found this :
> 10(4159) ERROR: tcp_blocking_connect: poll error: flags 18
> 10(4159) ERROR: tcp_blocking_connect: SO_ERROR (111) Connection refused
> 10(4159) ERROR: tcpconn_connect: tcp_blocking_connect failed
> 10(4159) ERROR: tcp_send: connect failed
> 10(4159) msg_send: ERROR: tcp_send failed
>
>
> is there another function like rewritehostport() to renew a SIP 
> exchange without tls ?
>





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