Pin sets module cannot complete as dialed

Running a PIAF installation:
FreePBX Version = 2.10.1.19
Running Asterisk Version = 1.8.28.0

Pin set module version is 2.11.0.9

When I dial out of the trunk that has the pin set assigned I get the message “call cannot be completed as dialed”.

 == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Executing [<longdistnumber>@from-internal:1] ResetCDR("SIP/2311-0000087e", "") in new stack
    -- Executing [<longdistnumber>@from-internal:2] NoCDR("SIP/2311-0000087e", "") in new stack
    -- Executing [<longdistnumber>@from-internal:3] Progress("SIP/2311-0000087e", "") in new stack
    -- Executing [<longdistnumber>@from-internal:4] Wait("SIP/2311-0000087e", "1") in new stack
    -- Executing [<longdistnumber>@from-internal:5] Progress("SIP/2311-0000087e", "") in new stack
    -- Executing [<longdistnumber>@from-internal:6] Playback("SIP/2311-0000087e", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
    -- <SIP/2311-0000087e> Playing 'silence/1.gsm' (language 'en')
    -- <SIP/2311-0000087e> Playing 'cannot-complete-as-dialed.gsm' (language 'en')
  == Spawn extension (from-internal, <longdistnumber>, 6) exited non-zero on 'SIP/2311-0000087e'
    -- Executing [[email protected]:1] Hangup("SIP/2311-0000087e", "") in new stack
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/2311-0000087e'

The trunk provider says they’re receiving the traffic. I placed a password for the outbound route in the Password field and removed the pin set; that’s working correctly. The pin set module, isn’t, however. I looked around, and the only forum posts that I could find do not have a solution, or stated that there was a fix in an earlier version of the pin set module.

No one else has had this issue?

Also seeing the same problem since update to Pinset Version 2.11.0.9.

If I put the pin set to None, it works fine. If I add a pinset I get the recording, your call cannot be completed as dialed.

With PINSET:

*CLI>
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Executing [[email protected]:1] ResetCDR(“SIP/6275-000743fb”, “”) in new stack
– Executing [[email protected]:2] NoCDR(“SIP/6275-000743fb”, “”) in new stack
– Executing [[email protected]:3] Progress(“SIP/6275-000743fb”, “”) in new stack
– Executing [[email protected]:4] Wait(“SIP/6275-000743fb”, “1”) in new stack
– Executing [[email protected]:5] Progress(“SIP/6275-000743fb”, “”) in new stack
– Executing [[email protected]:6] Playback(“SIP/6275-000743fb”, “silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer”) in new stack
– <SIP/6275-000743fb> Playing ‘silence/1.ulaw’ (language ‘en’)
– <SIP/6275-000743fb> Playing ‘cannot-complete-as-dialed.ulaw’ (language ‘en’)
== Spawn extension (from-internal, 53072007900, 6) exited non-zero on ‘SIP/6275-000743fb’
– Executing [[email protected]:1] Hangup(“SIP/6275-000743fb”, “”) in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/6275-000743fb’

Same on two more boxes, both running asterisk 1.8/freepbx 2.10.

-- Executing [[email protected]:1] Macro("SIP/1011-0000c644", "user-callerid,LIMIT") in new stack
-- Executing [[email protected]:1] Set("SIP/1011-0000c644", "AMPUSER=1011") in new stack
-- Executing [[email protected]:2] GotoIf("SIP/1011-0000c644", "0?report") in new stack
-- Executing [[email protected]:3] ExecIf("SIP/1011-0000c644", "1?Set(REALCALLERIDNUM=1011)") in new stack
-- Executing [[email protected]:4] Set("SIP/1011-0000c644", "AMPUSER=1011") in new stack
-- Executing [[email protected]:5] Set("SIP/1011-0000c644", "AMPUSERCIDNAME=Test extension") in new stack
-- Executing [[email protected]:6] GotoIf("SIP/1011-0000c644", "0?report") in new stack
-- Executing [[email protected]:7] Set("SIP/1011-0000c644", "AMPUSERCID=1011") in new stack
-- Executing [[email protected]:8] Set("SIP/1011-0000c644", "CALLERID(all)="Test extension" <1011>") in new stack
-- Executing [[email protected]:9] GotoIf("SIP/1011-0000c644", "0?limit") in new stack
-- Executing [[email protected]:10] ExecIf("SIP/1011-0000c644", "1?Set(GROUP(concurrency_limit)=1011)") in new stack
-- Executing [[email protected]:11] GosubIf("SIP/1011-0000c644", "7?sub-ccss,s,1(from-internal,12027621069)") in new stack
-- Executing [[email protected]:1] ExecIf("SIP/1011-0000c644", "0?Return()") in new stack
-- Executing [[email protected]:2] Set("SIP/1011-0000c644", "CCSS_SETUP=TRUE") in new stack
-- Executing [[email protected]:3] GosubIf("SIP/1011-0000c644", "0?monitor_config,1(from-internal,12027621069):monitor_default,1(from-internal,12027621069)") in new stack
-- Executing [[email protected]:1] GotoIf("SIP/1011-0000c644", "0?is_exten") in new stack
-- Executing [[email protected]:2] StackPop("SIP/1011-0000c644", "") in new stack
-- Executing [[email protected]:3] Return("SIP/1011-0000c644", "FALSE") in new stack
-- Executing [[email protected]:12] ExecIf("SIP/1011-0000c644", "0?Set(CHANNEL(language)=)") in new stack
-- Executing [[email protected]:13] GotoIf("SIP/1011-0000c644", "1?continue") in new stack
-- Goto (macro-user-callerid,s,26)
-- Executing [[email protected]:26] Set("SIP/1011-0000c644", "CALLERID(number)=1011") in new stack
-- Executing [[email protected]:27] Set("SIP/1011-0000c644", "CALLERID(name)=Test extension") in new stack
-- Executing [[email protected]:28] Set("SIP/1011-0000c644", "CHANNEL(language)=en") in new stack
-- Executing [[email protected]:2] Set("SIP/1011-0000c644", "ROUTEUSER=1011") in new stack
-- Executing [[email protected]:3] GotoIf("SIP/1011-0000c644", "1?outbound-3-4,12027621069,2:outbound-allroutes,12027621069,2") in new stack
-- Goto (outbound-3-4,12027621069,2)

On an asterisk 11/freepbx 2.11, Pinset Version 2.11.0.9 is working as expected.

Ok so upgrade to 2.11. FreePBX 10 is receiving security fixes only.

freepbx 2.11 doesn’t have any issues running on asterisk 1.8?

@mwaters_atni_com

I got this working by downgrading to pinsets 2.10.0.1. I have no idea what implications this might have, so I’m not giving any advice here, I’m just saying that’s what got to work again.