I want to connect to a SIP provider to make outbound call to US phone numbers from our existing platform and receive inbound call to DIDs. I tried to do that with JAIN SIP (java lib) but only the signal part was successful, I couldn’t stream audio through RTP somehow (audience just don’t hear any sound). So, I installed Asterisk 18 & FreePBX 15 as ultimate solution, but I can’t make any outbound call due to below error:
[2023-07-02 14:35:55] ERROR[200062][C-00000014]: pbx_functions.c:612ast_func_read: Dangerous function DB read blocked
[2023-07-02 14:35:55] ERROR[200062][C-00000014]: pbx_functions.c:655ast_func_read2: Dangerous function DB read blocked
[2023-07-02 14:35:55] NOTICE[200062][C-00000014]: chan_sip.c:30836sip_request_call: Asked to get a channel without offering any format
[2023-07-02 14:35:55] NOTICE[200062][C-00000014]: app_dial.c:2709dial_exec_full: Unable to create channel of type ‘SIP’ (cause 58 - Bearer capability not available) == Everyone is busy/congested at this time (1:0/0/1)
I sent the Originate Action as below (asterisk-java v3.39):
FreePBX isn’t supported here. First time users of FreePBX should use their distribution, which contains a consistent combination of OS, Asterisk and FreePBX.
chan_sip isn’t supported. Use chan_pjsip
What is in your sip.conf for peer 102. My guess is nothing.
What is in the dialplan to support this extension?
You should not be using chan_sip. The equivalent of this is only possible with chan_pjsip, and there will be no chan_sip at all in the next release of Asterisk. If you must do this for chan_sip, only include yyy.yy.yyyy.yy as the first section will hande calls from the x’s.
What’s an “outbound route”. Sounds like something from FreePBX (as is quoting bits of sip.conf without the section names).
You are missing the codecs for the second section.
nat=no is really only needed for Cisco. The default is OK.
canreinvite is a very obsolete name for directmedia