I’ve just migrated my business phones numbers to SIP and have problem with incoming calling rules. This is on Asterisk 1.8 on a NAS Synology. It seems that one of the incoming rules (I have one for each trunk) operates as a default, all incoming call to each trunk are treated by the same incoming rule . I’ve tested with an other SIP provider, and the rule for this trunk is correctly routed. But all the other incoming rules from the same provider did not work correctly. Any idea ? Many thanks in advance for your help.
The only way that Asterisk can match the incoming calls with sip.conf entries is on the basis of the host IP address.
What you actually need is a single, true, direct in dialing trunk, so that you can separate the different lines extensions.conf. You might be able to simulate this by using a callback extension on the register.
Thanks for info.I’ve read on viewtopic.php?f=1&t=74822&start=0#p147153 something that could be a solution, but I really do not have the level to implement this Do you think it could be a solution ? I’ve just migrated from physical lines to SIP and would not be happy to change back…
I’ve got an answer from the provider. He set up the trunks to send the destination phone number and on the Asterisk’s side I’ve put all the calling rules on a single trunk, and set the destination phone number as pattern. Now all is fine, I can get calls to any of my numbers correctly routed