Benjamin Stocker
2006-Jun-06 03:31 UTC
[Asterisk-Users] Asterisk Realtime and SIP Registration
Hi! I use the following configuration to register my asterisk server to my SIP provider: register => 12345:passwd@sip.provider.com/12345 sip.conf: [sipout-test] type=peer username=12345 fromuser=12345 fromdomain=provider.com secret=passwd insecure=very host=sip.provider.com qualify=yes context=test-incoming extensions.conf: exten => 12345,1,Dial(SIP/10) exten => _0NXZXXXXXX,1,Dial(SIP/${EXTEN}@sipout-test) This works fine when I put it into the config files. I can dial other numbers via my provider and receive calls. Wenn I put everything into Realtime tables (except the register command), incoming calls work only after * I make at least one outgoing call - or - * Somebody calls me twice On incoming calls, the caller first gets a 'user unavailale' from my SIP provider. When hanging up and calling again, the connection establishes successfully and I see this when entering 'sip show peers': sipout-test/12345 IP.AD.DR.ESS 5060 UNKNOWN This line does not show up when I registering my phone to my asterisk server. But it shows up immediately after registerung the phone when I use config files instead of RTA. I don't know wheter this is RTA- or a config-problem. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060606/c8aa04aa/attachment.htm
Douglas Garstang
2006-Jun-15 07:53 UTC
[Asterisk-Users] Asterisk Realtime and SIP Registration
Kevin Fleming has said on numerous ocassions that this is known not to work, and is not supported. -----Original Message----- From: Benjamin Stocker [mailto:bstocker@gmail.com] Sent: Tuesday, June 06, 2006 4:31 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk Realtime and SIP Registration Hi! I use the following configuration to register my asterisk server to my SIP provider: register => 12345:passwd@sip.provider.com/12345 sip.conf : [sipout-test] type=peer username=12345 fromuser=12345 fromdomain= provider.com secret=passwd insecure=very host= sip.provider.com <http://sip.provider.com> qualify=yes context=test-incoming extensions.conf: exten => 12345,1,Dial(SIP/10) exten => _0NXZXXXXXX,1,Dial(SIP/${EXTEN}@sipout-test) This works fine when I put it into the config files. I can dial other numbers via my provider and receive calls. Wenn I put everything into Realtime tables (except the register command), incoming calls work only after * I make at least one outgoing call - or - * Somebody calls me twice On incoming calls, the caller first gets a 'user unavailale' from my SIP provider. When hanging up and calling again, the connection establishes successfully and I see this when entering 'sip show peers': sipout-test/12345 IP.AD.DR.ESS 5060 UNKNOWN This line does not show up when I registering my phone to my asterisk server. But it shows up immediately after registerung the phone when I use config files instead of RTA. I don't know wheter this is RTA- or a config-problem. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060615/4c9099d4/attachment.htm