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I have a trunk set up for the DID from my provider: [my_provider] type=registration outbound_auth=my_provider server_uri=sip:sip.example.com client_uri=sip:my_username at sip.example.com retry_interval=60 [my_provider] type=auth auth_type=userpass password=123456 username=my_username [my_provider] type=aor contact=sip:sip.example.com:5060 [my_provider] typ...

...?welcome to company this is Cameron?) -> their voice flows 2 seconds later and we hear ?ameron? If we talk they can't here anything either at the beginning.' My * version: 1.6.2 sip.conf: .... nat=yes canreinvite=no .... extensions.conf: exten => my_nr,1,Dial(SIP/my_mobile_nr at my_provider,,r) Thanks in advance for any help. Kind regards, tylmann

...header if it can't find a match by ip address. Here's some info on it anyway. If you're provider puts something unique and constant in the headers, like a User-Agent string that doesn't change, you can also try using the "match_header" parameter to an identify object. [my_provider] type = identify match_header = User-Agent: Something Unique 1.0.0 endpoint = provider It has to be an exact match though, no wildcards or regular expressions. I opened an issue[1] on separating ip matching from header matching so they can be re-ordered. [1] https://issues.asterisk.org/jira/b...

Thanks George I originally didn?t have the 1002@ for the identify. Changed that when things were not working. I changed it back. Unfortunately, the system I am connecting with doesn?t seem to support the line support. Looking at the SIP packets, I see Asterisk send it. Unfortunately, they do not send the line information as part of the INVITE. I checked with some developers of that system