If DAHDI is not really configured or chan_dahdi isn't loaded the the error
mesage would be "can not create channel of type DAHDI" but here its
not the
case. Dadhi module is indeed loaded but the DAHDI device is not working
properly.
On Thu, Oct 6, 2011 at 8:49 PM, Gohar Ahmed <gohar.ahmed at vopium.com>
wrote:
> Hey,****
>
> How?ve you configured your Outbound trunk ? DAHDI/1/04712527270 : What do
> you?ve in your dahdi configuration file ! I doubt this ?/1? is the culprit
> or else your DAHDI channel is not really working at all.****
>
> ** **
>
> Regards,****
>
> Gohar A.****
>
> ** **
>
> *From:* asterisk-users-bounces at lists.digium.com [mailto:
> asterisk-users-bounces at lists.digium.com] *On Behalf Of *michael k
> *Sent:* Thursday, October 06, 2011 8:46 PM
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] PSTN connectivity****
>
> ** **
>
> Hi All,
>
>
> I got a busy message like "all lines are currently busy and
> please try again later" in call to ZAP trunk. Please help me to
resolve
> this issue
>
>
> == Using SIP RTP TOS bits 184
> == Using SIP RTP CoS mark 5
> -- Executing [904712527270 at from-internal:1]
Macro("SIP/157-00000000",
> "user-callerid,SKIPTTL,") in new stack
> -- Executing [s at macro-user-callerid:1]
Set("SIP/157-00000000",
> "AMPUSER=157") in new stack
> -- Executing [s at macro-user-callerid:2]
GotoIf("SIP/157-00000000",
> "0?report") in new stack
> -- Executing [s at macro-user-callerid:3]
ExecIf("SIP/157-00000000",
> "1?Set(REALCALLERIDNUM=157)") in new stack
> -- Executing [s at macro-user-callerid:4]
Set("SIP/157-00000000",
> "AMPUSER=157") in new stack
> -- Executing [s at macro-user-callerid:5]
Set("SIP/157-00000000",
> "AMPUSERCIDNAME=Rojar S") in new stack
> -- Executing [s at macro-user-callerid:6]
GotoIf("SIP/157-00000000",
> "0?report") in new stack
> -- Executing [s at macro-user-callerid:7]
Set("SIP/157-00000000",
> "AMPUSERCID=157") in new stack
> -- Executing [s at macro-user-callerid:8]
Set("SIP/157-00000000",
> "CALLERID(all)="Rojar S" <157>") in new stack
> -- Executing [s at macro-user-callerid:9]
ExecIf("SIP/157-00000000",
> "0?Set(CHANNEL(language)=)") in new stack
> -- Executing [s at macro-user-callerid:10]
GotoIf("SIP/157-00000000",
> "1?continue") in new stack
> -- Goto (macro-user-callerid,s,19)
> -- Executing [s at macro-user-callerid:19]
Set("SIP/157-00000000",
> "CALLERID(number)=157") in new stack
> -- Executing [s at macro-user-callerid:20]
Set("SIP/157-00000000",
> "CALLERID(name)=Rojar S") in new stack
> -- Executing [s at macro-user-callerid:21]
NoOp("SIP/157-00000000",
> "Using CallerID "Rojar S" <157>") in new stack
> -- Executing [904712527270 at from-internal:2]
Set("SIP/157-00000000",
> "_NODEST=") in new stack
> -- Executing [904712527270 at from-internal:3]
Macro("SIP/157-00000000",
> "record-enable,157,OUT,") in new stack
> -- Executing [s at macro-record-enable:1]
GotoIf("SIP/157-00000000",
> "1?check") in new stack
> -- Goto (macro-record-enable,s,4)
> -- Executing [s at macro-record-enable:4]
ExecIf("SIP/157-00000000",
> "0?MacroExit()") in new stack
> -- Executing [s at macro-record-enable:5]
GotoIf("SIP/157-00000000",
> "0?Group:OUT") in new stack
> -- Goto (macro-record-enable,s,15)
> -- Executing [s at macro-record-enable:15]
GotoIf("SIP/157-00000000",
> "0?IN") in new stack
> -- Executing [s at macro-record-enable:16]
ExecIf("SIP/157-00000000",
> "1?MacroExit()") in new stack
> -- Executing [904712527270 at from-internal:4]
Macro("SIP/157-00000000",
> "dialout-trunk,1,04712527270,,") in new stack
> -- Executing [s at macro-dialout-trunk:1]
Set("SIP/157-00000000",
> "DIAL_TRUNK=1") in new stack
> -- Executing [s at macro-dialout-trunk:2]
GosubIf("SIP/157-00000000",
> "0?sub-pincheck,s,1") in new stack
> -- Executing [s at macro-dialout-trunk:3]
GotoIf("SIP/157-00000000",
> "0?disabletrunk,1") in new stack
> -- Executing [s at macro-dialout-trunk:4]
Set("SIP/157-00000000",
> "DIAL_NUMBER=04712527270") in new stack
> -- Executing [s at macro-dialout-trunk:5]
Set("SIP/157-00000000",
> "DIAL_TRUNK_OPTIONS=tr") in new stack
> -- Executing [s at macro-dialout-trunk:6]
Set("SIP/157-00000000",
> "OUTBOUND_GROUP=OUT_1") in new stack
> -- Executing [s at macro-dialout-trunk:7]
GotoIf("SIP/157-00000000",
> "0?nomax") in new stack
> -- Executing [s at macro-dialout-trunk:8]
GotoIf("SIP/157-00000000",
> "0?chanfull") in new stack
> -- Executing [s at macro-dialout-trunk:9]
GotoIf("SIP/157-00000000",
> "0?skipoutcid") in new stack
> -- Executing [s at macro-dialout-trunk:10]
Set("SIP/157-00000000",
> "DIAL_TRUNK_OPTIONS=") in new stack
> -- Executing [s at macro-dialout-trunk:11]
Macro("SIP/157-00000000",
> "outbound-callerid,1") in new stack
> -- Executing [s at macro-outbound-callerid:1]
ExecIf("SIP/157-00000000",
> "0?Set(CALLERPRES()=)") in new stack
> -- Executing [s at macro-outbound-callerid:2]
ExecIf("SIP/157-00000000",
> "0?Set(REALCALLERIDNUM=157)") in new stack
> -- Executing [s at macro-outbound-callerid:3]
GotoIf("SIP/157-00000000",
> "1?normcid") in new stack
> -- Goto (macro-outbound-callerid,s,6)
> -- Executing [s at macro-outbound-callerid:6]
Set("SIP/157-00000000",
> "USEROUTCID=") in new stack
> -- Executing [s at macro-outbound-callerid:7]
Set("SIP/157-00000000",
> "EMERGENCYCID=") in new stack
> -- Executing [s at macro-outbound-callerid:8]
Set("SIP/157-00000000",
> "TRUNKOUTCID=") in new stack
> -- Executing [s at macro-outbound-callerid:9]
GotoIf("SIP/157-00000000",
> "1?trunkcid") in new stack
> -- Goto (macro-outbound-callerid,s,12)
> -- Executing [s at macro-outbound-callerid:12]
ExecIf("SIP/157-00000000",
> "0?Set(CALLERID(all)=)") in new stack
> -- Executing [s at macro-outbound-callerid:13]
ExecIf("SIP/157-00000000",
> "0?Set(CALLERID(all)=)") in new stack
> -- Executing [s at macro-outbound-callerid:14]
ExecIf("SIP/157-00000000",
> "0?Set(CALLERID(all)=)") in new stack
> -- Executing [s at macro-outbound-callerid:15]
ExecIf("SIP/157-00000000",
> "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
> -- Executing [s at macro-dialout-trunk:12]
ExecIf("SIP/157-00000000",
> "0?AGI(fixlocalprefix)") in new stack
> -- Executing [s at macro-dialout-trunk:13]
Set("SIP/157-00000000",
> "OUTNUM=04712527270") in new stack
> -- Executing [s at macro-dialout-trunk:14]
Set("SIP/157-00000000",
> "custom=DAHDI/1") in new stack
> -- Executing [s at macro-dialout-trunk:15]
ExecIf("SIP/157-00000000",
> "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^))") in new stack
> -- Executing [s at macro-dialout-trunk:16]
Macro("SIP/157-00000000",
> "dialout-trunk-predial-hook,") in new stack
> -- Executing [s at macro-dialout-trunk-predial-hook:1]
> MacroExit("SIP/157-00000000", "") in new stack
> -- Executing [s at macro-dialout-trunk:17]
GotoIf("SIP/157-00000000",
> "0?bypass,1") in new stack
> -- Executing [s at macro-dialout-trunk:18]
GotoIf("SIP/157-00000000",
> "0?customtrunk") in new stack
> -- Executing [s at macro-dialout-trunk:19]
Dial("SIP/157-00000000",
> "DAHDI/1/04712527270,300,") in new stack
> == Everyone is busy/congested at this time (1:0/0/1)
> -- Executing [s at macro-dialout-trunk:20]
NoOp("SIP/157-00000000", "Dial
> failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE =
66")
> in new stack
> -- Executing [s at macro-dialout-trunk:21]
Goto("SIP/157-00000000",
> "s-CHANUNAVAIL,1") in new stack
> -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
> -- Executing [s-CHANUNAVAIL at macro-dialout-trunk:1]
> Set("SIP/157-00000000", "RC=66") in new stack
> -- Executing [s-CHANUNAVAIL at macro-dialout-trunk:2]
> Goto("SIP/157-00000000", "66,1") in new stack
> -- Goto (macro-dialout-trunk,66,1)
> -- Executing [66 at macro-dialout-trunk:1]
Goto("SIP/157-00000000",
> "continue,1") in new stack
> -- Goto (macro-dialout-trunk,continue,1)
> -- Executing [continue at macro-dialout-trunk:1]
> GotoIf("SIP/157-00000000", "1?noreport") in new stack
> -- Goto (macro-dialout-trunk,continue,3)
> -- Executing [continue at macro-dialout-trunk:3]
NoOp("SIP/157-00000000",
> "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 66 - failing
through to
> other trunks") in new stack
> -- Executing [continue at macro-dialout-trunk:4]
Set("SIP/157-00000000",
> "CALLERID(number)=157") in new stack
> -- Executing [904712527270 at from-internal:5]
Macro("SIP/157-00000000",
> "outisbusy,") in new stack
> -- Executing [s at macro-outisbusy:1]
Progress("SIP/157-00000000", "") in
> new stack
> -- Executing [s at macro-outisbusy:2]
GotoIf("SIP/157-00000000",
> "0?emergency,1") in new stack
> -- Executing [s at macro-outisbusy:3]
GotoIf("SIP/157-00000000",
> "0?intracompany,1") in new stack
> -- Executing [s at macro-outisbusy:4]
Playback("SIP/157-00000000",
> "all-circuits-busy-now&pls-try-call-later, noanswer") in new
stack
> -- <SIP/157-00000000> Playing 'all-circuits-busy-now.gsm'
(language
> 'en')
> -- <SIP/157-00000000> Playing 'pls-try-call-later.gsm'
(language 'en')
> -- Executing [s at macro-outisbusy:5]
Congestion("SIP/157-00000000",
> "20") in new stack
> == Spawn extension (macro-outisbusy, s, 5) exited non-zero on
> 'SIP/157-00000000' in macro 'outisbusy'
> == Spawn extension (from-internal, 904712527270, 5) exited non-zero on
> 'SIP/157-00000000'
> -- Executing [h at from-internal:1] Macro("SIP/157-00000000",
> "hangupcall") in new stack
> -- Executing [s at macro-hangupcall:1]
GotoIf("SIP/157-00000000",
> "1?skiprg") in new stack
> -- Goto (macro-hangupcall,s,4)
> -- Executing [s at macro-hangupcall:4]
GotoIf("SIP/157-00000000",
> "1?skipblkvm") in new stack
> -- Goto (macro-hangupcall,s,7)
> -- Executing [s at macro-hangupcall:7]
GotoIf("SIP/157-00000000",
> "1?theend") in new stack
> -- Goto (macro-hangupcall,s,9)
> -- Executing [s at macro-hangupcall:9]
Hangup("SIP/157-00000000", "") in
> new stack
> == Spawn extension (macro-hangupcall, s, 9) exited non-zero on
> 'SIP/157-00000000' in macro 'hangupcall'
> == Spawn extension (from-internal, h, 1) exited non-zero on
> 'SIP/157-00000000'
>
>
> Michael.k
>
>
> ****
>
> On Fri, Sep 30, 2011 at 10:33 AM, Sam Govind <govoiper at gmail.com>
wrote:**
> **
>
> Hey Warren I thought that these are the complete CLI logs for one call. It
> started like " == Using SIP RTP CoS mark 5" and from-internal
priority-1
> ..So that seemed legit to me. Yeah I too suspect that dialing rules are not
> being matched and thats why Gotoif's are failing.****
>
> On Thu, Sep 29, 2011 at 8:15 PM, Warren Selby <wcselby at
selbytech.com>
> wrote:****
>
>
> On Thu, Sep 29, 2011 at 7:51 AM, michael k <michael at inapp.com>
wrote:****
>
> Thanks for the update. but how do i resolve this issue ? can you help me
> please ? ****
>
>
> You didn't provide a full CLI trace of the outgoing call, you only
supplied
> the hangup portion of the call. Please try again.
>
> Also, what are the dialing rules like in your country? You only have
> outbound dial patterns setup to handle North American numbers (8+
NXXNXXXXXX
> or 8+ NXXXXXX).
> The Dial Pattern box in the Outbound Rules box is where you define what
> numbers you want to go out over this trunk. If you dial a number that
> doesn't match one of these
> patterns, FreePBX is going to look internally for a dial pattern to match
> against, and if it doesn't find one there, it will end the call.
>
>
> -- ****
>
> Thanks,
> --Warren Selby, dCAP
> http://www.SelbyTech.com <http://www.selbytech.com>
>
> ****
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users****
>
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users****
>
> ** **
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20111006/eaf1e85d/attachment.htm>