hello Carles Wang
This is the way to to successfully call through asterisk.
SER is Main Registrar. Asterisk is B2BUA.
SER will route its call to asterisk. now control is at asterisk side he can do any thing to that call (forword, voicemail, etc....). but actually asterisk dont know who is the user he ll route this call back to SER who is aware of the user. it is like U-Turn through Asterisk. try this and tell me
ser.cfg ------------------------ rewritehostport("asterisk:port"); force_rport(); fix_nated_contact(); force_rtp_proxy(); t_relay();
sip.conf ------------------------ [general] context=default port=5070 bindaddr=0.0.0.0 srvlookup=no
[2000] callerid=2000 type=friend host=dynamic canreinvite=no reinvite=no qualify=yes auth=plaintext nat=yes
[3000] type=friend host=dynamic callerid=3000 qualify=yes canreinvite=no reinvite=no auth=plaintext nat=yes
--- Charles Wang lazy.charles@gmail.com wrote:
I also have the same problem. I don't know how to fix it. Do you get any solution or sample config about it. I think it is a problem of asterisk.
On 4/28/05, Kamran Ahmad p_kami@yahoo.com wrote:
hi list
any person using ser and Asterisk as B2bUA.
what is the best way of using asterisk as B2BUA.
i can route calls to through asterisk its working
but
some time one UA tries to hangup the call and the other side is not able to receive Bye
what could be the best way of using asterisk for monitering voice
Thanks Kamran
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Best Regards Charles
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