Cannot make outbound calls from public IP, but can from priv

Asterisk Asterisk Support
1

Hello All!

I am pretty new to the whole Asterisk PBX thing and my apologies, if my questions seems dumb.

My problem is this. After installing the newest version with CentOS 6.5 from Downloads and configure my SIP and extensions, I can make outbound calls. But when I put the machine with Public IP, I cannot make any outbound calls, except to other extensions.

After digging the logs for hours, the only thing I am constantly seeing are those errors.

[2014-04-25 13:22:46] VERBOSE[14969][C-00000001] netsock2.c: == Using SIP RTP TOS bits 184
[2014-04-25 13:22:46] VERBOSE[14969][C-00000001] netsock2.c: == Using SIP RTP CoS mark 5
[2014-04-25 13:22:46] VERBOSE[14969][C-00000001] app_dial.c: – Called SIP/ITDxxxx/0876xxxxxx
[2014-04-25 13:22:46] NOTICE[14766][C-00000001] chan_sip.c: Failed to authenticate on INVITE to ‘;tag=as6daa5a86’
[2014-04-25 13:22:46] VERBOSE[14969][C-00000001] app_dial.c: – SIP/ITD8378-00000003 is circuit-busy
[2014-04-25 13:22:46] VERBOSE[14969][C-00000001] app_dial.c: == Everyone is busy/congested at this time (1:0/1/0)
[2014-04-25 13:22:46] VERBOSE[14969][C-00000001] pbx.c: – Executing [s@macroEmail links icon-dialout-trunk:23] NoOp(“SIP/9112-00000002”, “Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 21”) in new stack
[2014-04-25 13:22:46] VERBOSE[14969][C-00000001] pbx.c: – Executing [s@macroEmail links icon-dialout-trunk:24] GotoIf(“SIP/9112-00000002”, “0?continue,1:s-CONGESTION,1”) in new stack
[2014-04-25 13:22:46] VERBOSE[14969][C-00000001] pbx.c: – Goto (macro-dialout-trunk,s-CONGESTION,1)
[2014-04-25 13:22:46] VERBOSE[14969][C-00000001] pbx.c: – Executing [s-CONGESTION@macroEmail links icon-dialout-trunk:1] Set(“SIP/9112-00000002”, “RC=21”) in new stack
[2014-04-25 13:22:46] VERBOSE[14969][C-00000001] pbx.c: – Executing [s-CONGESTION@macroEmail links icon-dialout-trunk:2] Goto(“SIP/9112-00000002”, “21,1”) in new stack
[2014-04-25 13:22:46] VERBOSE[14969][C-00000001] pbx.c: – Goto (macro-dialout-trunk,21,1)
[2014-04-25 13:22:46] VERBOSE[14969][C-00000001] pbx.c: – Executing [21@macroEmail links icon-dialout-trunk:1] Goto(“SIP/9112-00000002”, “continue,1”) in new stack
[2014-04-25 13:22:46] VERBOSE[14969][C-00000001] pbx.c: – Goto (macro-dialout-trunk,continue,1)
[2014-04-25 13:22:46] VERBOSE[14969][C-00000001] pbx.c: – Executing [continue@macroEmail links icon-dialout-trunk:1] NoOp(“SIP/9112-00000002”, “TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 21 - failing through to other trunks”) in new stack
[2014-04-25 13:22:46] VERBOSE[14969][C-00000001] pbx.c: – Executing [continue@macroEmail links icon-dialout-trunk:2] Set(“SIP/9112-00000002”, “CALLERID(number)=9112”) in new stack
[2014-04-25 13:22:46] VERBOSE[14969][C-00000001] pbx.c: – Executing [0876xxxxxx@fromEmail links icon-internal:7] Macro(“SIP/9112-00000002”, “outisbusy,”) in new stack
[2014-04-25 13:22:46] VERBOSE[14969][C-00000001] pbx.c: – Executing [s@macroEmail links icon-outisbusy:1] Progress(“SIP/9112-00000002”, “”) in new stack
[2014-04-25 13:22:46] VERBOSE[14969][C-00000001] pbx.c: – Executing [s@macroEmail links icon-outisbusy:2] GotoIf(“SIP/9112-00000002”, “0?emergency,1”) in new stack
[2014-04-25 13:22:46] VERBOSE[14969][C-00000001] pbx.c: – Executing [s@macroEmail links icon-outisbusy:3] GotoIf(“SIP/9112-00000002”, “0?intracompany,1”) in new stack
[2014-04-25 13:22:46] VERBOSE[14969][C-00000001] pbx.c: – Executing [s@macroEmail links icon-outisbusy:4] Playback(“SIP/9112-00000002”, “all-circuits-busy-now&pls-try-call-later, noanswer”) in new stack
[2014-04-25 13:22:46] VERBOSE[14969][C-00000001] file.c: – Playing ‘all-circuits-busy-now.gsm’ (language ‘en’)
[2014-04-25 13:22:47] VERBOSE[14969][C-00000001] file.c: – Playing ‘pls-try-call-later.gsm’ (language ‘en’)

If someone needs more specific logs, I can provide them.
Thank you in advance!

Regards

2

Cause 21 is call rejected. In combination with the authentication issue I think you are routing the INVITE back to the originating Asterisk system.

3

Thank you for the reply, but could you please explain me a bit more, like an for an idiot, please. It would be very helpful.

Thank you in advance!

Regards

4

The IP packets you are sending appear to be being retuned, by your router, etc., to the Asterisk box, which is then rejecting the request.

5

How this is is possible? Where I can look at and fix or provide more info, so you could suggest me something.

6

At your router configuration.

Otherwise you will need to provide network topology information, details of the router, the contents of sip.conf (and, as that looks like a GUI dialplan, users.conf) and the output of sip set debug on for the call (INVITE and all responses to it, and the request that has the authentication failure).