The problem is, I have set the SIP Trunk port to 5069, but in the debug, freepbx still sends 5060. Why is it sending 5060 and how can I get it to send 5069 instead which is what my provider is accepting. Thanks!
INFO
On my router:
5069 is forwarded to freepbx (SIP Port)
100000 - 20000 forwarded to freepbx (RTP ports)
In the debug,
I changed some numbers for privacy
the called number is 00027606001234
My external IP is 1.2.3.4
My extension is 200
My Trunk Settings:
Host=x.x.x.x
Bindport=5069
Type=peer
Disallow=all
Allow=ulaw&g729
Dtmfmode=rfc2833
Qualify=yes
Call Debug (abbrviated):
=========================================================================
Connected to Asterisk 11.13.1 currently running on localhost (pid = 2013)
[2014-11-25 16:24:12] DEBUG[8552][C-0000d579]: chan_sip.c:6207 sip_call: Outgoing Call for 7606001234
[2014-11-25 16:24:12] DEBUG[8552][C-0000d579]: chan_sip.c:13085 add_sdp: ** Our capability: (ulaw|g729) Video flag: False Text flag: False
[2014-11-25 16:24:12] DEBUG[8552][C-0000d579]: chan_sip.c:13086 add_sdp: ** Our prefcodec: (ulaw)
[2014-11-25 16:24:12] DEBUG[8552][C-0000d579]: chan_sip.c:3367 initialize_initreq: Initializing initreq for method INVITE - callid [email protected]:5060
â Called SIP/SIPTrunk-2/7606001234
[2014-11-25 16:24:12] DEBUG[2288]: manager.c:5262 process_message: Running action âCommandâ
[2014-11-25 16:24:12] DEBUG[2086][C-0000d579]: chan_sip.c:4457 __sip_semi_ack: (Provisional) Stopping
retransmission (but retaining packet) on â[email protected]:5060â Request 102: Found
[2014-11-25 16:24:12] DEBUG[2086][C-0000d579]: chan_sip.c:4378 __sip_ack: Acked pending invite 102
[2014-11-25 16:24:12] DEBUG[2086][C-0000d579]: chan_sip.c:4416 __sip_ack: Stopping retransmission on â[email protected]:5060â of Request 102: Match Found
[2014-11-25 16:24:12] WARNING[2086][C-0000d579]: chan_sip.c:23019 handle_response_invite: Received response: âForbiddenâ from âsip:[email protected];tag=as2afbd62dâ
-- Executing [continue@macro-dialout-trunk:1] NoOp("SIP/200-0000219f", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 21 - failing through to other trunks") in new stack
[2014-11-25 16:24:12] DEBUG[8552][C-0000d579]: app_macro.c:434 _macro_exec: Executed application: Noop
[2014-11-25 16:24:12] DEBUG[8552][C-0000d579]: pbx.c:4883 pbx_extension_helper: Launching âSetâ
â Executing [continue@macro-dialout-trunk:2] Set(âSIP/200-0000219fâ, âCALLERID(number)=200â) in new stack
[2014-11-25 16:24:12] DEBUG[8552][C-0000d579]: res_rtp_asterisk.c:3308 ast_rtp_write: Ooh, format changed from unknown to ulaw
[2014-11-25 16:24:12] DEBUG[8552][C-0000d579]: res_rtp_asterisk.c:3343 ast_rtp_write: Created smoother: format: ulaw ms: 20 len: 160
[2014-11-25 16:24:12] DEBUG[8552][C-0000d579]: res_rtp_asterisk.c:3205 ast_rtp_raw_write: Starting RTCP transmission on RTP instance â0x7f86e06333f8â
[2014-11-25 16:24:12] DEBUG[8552][C-0000d579]: channel.c:3594 ast_settimeout_full: Scheduling timer at (50 requested / 50 actual) timer ticks per second
â <SIP/200-0000219f> Playing âall-circuits-busy-now.ulawâ (language âenâ)
[2014-11-25 16:24:12] DEBUG[8552][C-0000d579]: res_rtp_asterisk.c:4178 ast_rtp_read: 0x7f86e0cbc160 â Probation learning mode pass with source address 10.1.1.14:10002
> 0x7f86e0cbc160 â Probation passed - setting RTP source address to 10.1.1.14:10002
[2014-11-25 16:24:14] DEBUG[8552][C-0000d579]: channel.c:3594 ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2014-11-25 16:24:14] DEBUG[8552][C-0000d579]: channel.c:3594 ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2014-11-25 16:24:14] DEBUG[8552][C-0000d579]: channel.c:3594 ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2014-11-25 16:24:14] DEBUG[8552][C-0000d579]: channel.c:3594 ast_settimeout_full: Scheduling timer at (50 requested / 50 actual) timer ticks per second
â <SIP/200-0000219f> Playing âpls-try-call-later.ulawâ (language âenâ)
localhost*CLI> exit
Asterisk cleanly ending (0).