Outgoing PSTN not working - all circuits are busy

[brunski]

Hi, in every call i make thru my pstn i'm getting 'all circuits are busy'.
I only have 2 trunks: 1 with gizmo and 1 with pstn
I'm aso using spa 3102 and followed the tutorials in this forum to set it up. So far incoming to my PSTN work, but outgoing doesnt. Also gizmo to 0101xxxxxx is working.

I'm attaching my log, if you want me to get other information, please let me know and also how to get it (paths) as i'm not too familiar with the whole piaf setup. Thanks in advance for your help.

 

[jmullinix]

This part:

[2009-02-22 16:05:37] VERBOSE[1504] logger.c: -- Executing [s@macro-dialout-trunk:19] Dial("SIP/2004-b7d05cd0", "SIP/1-pstn/2815152339|300|") in new stack
[2009-02-22 16:05:37] WARNING[1504] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
[2009-02-22 16:05:37] VERBOSE[1504] logger.c: == Everyone is busy/congested at this time (1:0/0/1)
[2009-02-22 16:05:37] DEBUG[1504] app_macro.c: Executed application: Dial
[2009-02-22 16:05:37] VERBOSE[1504] logger.c: -- Executing [s@macro-dialout-trunk:20] Goto("SIP/2004-b7d05cd0", "s-CHANUNAVAIL|1") in new stack
[2009-02-22 16:05:37] VERBOSE[1504] logger.c: -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
[2009-02-22 16:05:37] DEBUG[1504] app_macro.c: Executed application: Goto
[2009-02-22 16:05:37] VERBOSE[1504] logger.c: -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] GotoIf("SIP/2004-b7d05cd0", "1?noreport") in new stack
[2009-02-22 16:05:37] VERBOSE[1504] logger.c: -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,3)
[2009-02-22 16:05:37] DEBUG[1504] app_macro.c: Executed application: GotoIf
[2009-02-22 16:05:37] VERBOSE[1504] logger.c: -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:3] NoOp("SIP/2004-b7d05cd0", "TRUNK Dial failed due to CHANUNAVAIL (hangupcause: 3) - failing through to other trunks") in new stack

indicates that the SPA is not talking to the PBX correctly. Look at your trunk setup for the SPA on FreePBX and the credentials setup on the SPA for the PSTN tab.

 

[brunski]

thanks for your repply, but, i checked and double checked the passwords and usernames in both the trunk and the pstn tab of the spa - everything matches. Could there be something elese?

 

[wardmundy]

There's already a very long thread on the SPA3102. Start there.

 

[brunski]

Thank you Ward, i've been reading that thread, have not made a breakthru yet. But i have made some progress - i dont get the 'circuits busy' recording, but instead after the connection is made, there is a clicking sound and a very long pause 10 secs or so, then i get the recording from the telco that 'the call did not go thru' - here is the code

 ==  fixlocalprefix: Dialpattern 281|NXXXXXX matched. 2812060165 -> 2060165
    -- AGI Script fixlocalprefix completed, returning 0
    -- Executing [s@macro-dialout-trunk:13] Set("SIP/2004-095bb968", "OUTNUM=2060165") in new stack
    -- Executing [s@macro-dialout-trunk:14] Set("SIP/2004-095bb968", "custom=SIP/1-pstn") in new stack
    -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/2004-095bb968", "0|Set|DIAL_TRUNK_OPTIONS=M(setmusic^)") in new stack
    -- Executing [s@macro-dialout-trunk:16] Macro("SIP/2004-095bb968", "dialout-trunk-predial-hook|") in new stack
    -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/2004-095bb968", "") in new stack
    -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/2004-095bb968", "0?bypass|1") in new stack
    -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/2004-095bb968", "0?customtrunk") in new stack
    -- Executing [s@macro-dialout-trunk:19] Dial("SIP/2004-095bb968", "SIP/1-pstn/2060165|300|") in new stack
    -- Called 1-pstn/2060165
    -- SIP/1-pstn-095bd590 is ringing
    -- SIP/1-pstn-095bd590 answered SIP/2004-095bb968
    -- Packet2Packet bridging SIP/2004-095bb968 and SIP/1-pstn-095bd590
  == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on 'SIP/2004-095bb968' in macro 'dialout-trunk'
  == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on 'SIP/2004-095bb968'
    -- Executing [h@macro-dialout-trunk:1] Macro("SIP/2004-095bb968", "hangupcall|") in new stack
    -- Executing [s@macro-hangupcall:1] ResetCDR("SIP/2004-095bb968", "w") in new stack
    -- Executing [s@macro-hangupcall:2] NoCDR("SIP/2004-095bb968", "") in new stack
    -- Executing [s@macro-hangupcall:3] GotoIf("SIP/2004-095bb968", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,6)
    -- Executing [s@macro-hangupcall:6] GotoIf("SIP/2004-095bb968", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] GotoIf("SIP/2004-095bb968", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,11)
    -- Executing [s@macro-hangupcall:11] Hangup("SIP/2004-095bb968", "") in new stack
  == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/2004-095bb968' in macro 'hangupcall'
  == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/2004-095bb968'

 

[brunski]

a quick note, here in Houston the callsmust be dialed as 10 digits, i see this in the code: 2812060165 -> 2060165 is this a change to 7 digits when dials out?

 

[MGD4me]

When you say "after the connection is made, there is a clicking sound and a very long pause" and reviewing the log above, it shows that indeed you ARE attempting to place a call using your PSTN line as a trunk. The clicks are noises from the telco, so that is a good sign. If your telco REQUIRES that you dial all 10 digits as you say, then look at the very first line of the log, and you will note that either your outbound route or trunk is 'stripping off' the 281 portion of the dialed number. Just get rid of the '281|' portion of the pattern.

 

[brunski]

your outbound route or trunk is 'stripping off' the 281 portion of the dialed number. Just get rid of the '281|' portion of the pattern.

Hi MGD4me Thanks!!! you nailed it. This time it did work.

 

[brunski]

new problem with outgoing..

can someone please help me dicipher this log? - my outgoing stopped working again.

Thanks!!!

root@pbx:/etc/asterisk $ asterisk -r
Connected to Asterisk 1.4.21.2 currently running on pbx (pid = 21269)
Verbosity is at least 3
    -- Executing [2812060165@from-internal:1] ResetCDR("SIP/2004-08412300", "") in new stack
    -- Executing [2812060165@from-internal:2] NoCDR("SIP/2004-08412300", "") in new stack
    -- Executing [2812060165@from-internal:3] Wait("SIP/2004-08412300", "1") in new stack
    -- Executing [2812060165@from-internal:4] Playback("SIP/2004-08412300", "silence/1&cannot-complete-as-dialed&check-number-dial-again|noanswer") in new stack
    -- <SIP/2004-08412300> Playing 'silence/1' (language 'en')
    -- <SIP/2004-08412300> Playing 'cannot-complete-as-dialed' (language 'en')
    -- <SIP/2004-08412300> Playing 'check-number-dial-again' (language 'en')
  == Spawn extension (from-internal, 2812060165, 4) exited non-zero on 'SIP/2004-08412300'
    -- Executing [h@from-internal:1] Macro("SIP/2004-08412300", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] ResetCDR("SIP/2004-08412300", "w") in new stack
    -- Executing [s@macro-hangupcall:2] NoCDR("SIP/2004-08412300", "") in new stack
    -- Executing [s@macro-hangupcall:3] GotoIf("SIP/2004-08412300", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,6)
    -- Executing [s@macro-hangupcall:6] GotoIf("SIP/2004-08412300", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] GotoIf("SIP/2004-08412300", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,11)
    -- Executing [s@macro-hangupcall:11] Hangup("SIP/2004-08412300", "") in new stack
  == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/2004-08412300' in macro 'hangupcall'
  == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/2004-08412300'

 

[edounn]

I had this problem, and made sure that i had enough channles set to my outbounds. Might need to make another outbound trunk too in pbx. Not sure why, mine doesn't even use it. But, it stopped this no channles available thing. I'm also using ptsn