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...le to a mobile phone) I see in disposition field the string "ANSWERED" when I reject the call and also when I really answer the call, while in the variable DIALSTAUS I have the correct status of the call (BUSY, CHANUNAVAIL, ANSWERED, NO ANSWER etc....). Can anyone help me? Bye, Damiano Bertuna. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070412/6b4c6d72/attachment.htm

Hi to everybody, I have an spa 3102 where i connected an analog phone (in the fxs port) and the pstn line (in the fxo port). This is my problem: the incoming call doesn't arrive to asterisk. In the spa web page i configured this dialplane: (<:line01@192.168.1.220:5060>) where line01 is the context in sip.conf, 192.168.1.220 is the asterisk ip and 5060 is the asterisk sip port.

Hi to everybody, I need some help in configuration of the spa 3102. I created an account for line 1 (user 208, sip port 5061) correctly registered in asterisk, then i create an account in sip.conf like this: [general] register = line01:pwdsipura:line01@192.168.1.222:5060/095377078 [line01] username = line01 fromuser = line01 secret = pwdsipura host = 192.168.1.222 fromdomain = 192.168.1.222

This is my problem: I have two server, server 1 in iax.conf [general] register=> isdn_server:isdn@169.254.68.200 [voip_server] type=friend username=damiano_voip auth=md5 host=dynamic

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Hi, I used Grandstream (gxp2000, gxp2020) and Snom (370) SIP Phones, but with 2 extensions BLF status does not work correctly. have someone ever tested a Sip Phone with more then 60 BLF without problems? Can someone suggest me model and brand? Thanks, bye. -------------- next part -------------- An HTML attachment was scrubbed... URL: