>>> > exten => _9X.,1,Dial(Dongle/dongle800/#31#${EXTEN:1},120,KT)My suggestion would be to add a pause or two before dialing the phone number exten => _9X.,1,Dial(Dongle/dongle800/#31#ww${EXTEN:1},120,KT) D(digits): After the called party answers, send digits as a DTMF stream, then connect the call to the originating channel (you can also use 'w' to produce .5 second pauses). You can also provide digits after a colon - all digits before the colon are sent to the called channel, all digits after the colon are sent to the calling channel (all digits are sent to the called channel if there is no colon present). Doug
>>> My suggestion would be to add a pause or two before dialing the phone numberLooks like using w for a pause is no longer supported. Doug
after adding the ww:
root at Pbx: /etc/asterisk $ asterisk -rvvv
Asterisk 11.25.3, Copyright (C) 1999 - 2013 D == Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5 -- Executing
[9211123456 at AllCalls:1] Goto("SIP/500-00000003",
"DefaultPlan,9211123456,1") in new stack
--
Goto (DefaultPlan,92105727105,1)
-- Executing [9211123456 at DefaultPlan:1]
Dial("SIP/500-00000003",
"Dongle/dongle800/#31#ww211123456,120,KT") in new stack
[2018-04-10
13:23:46] WARNING[1327][C-00000003]: channel.c:79 parse_dial_string:
Invalid destination '#31#ww211123456' in chan_dongle, only
0123456789*#+ABC
allowed [2018-04-10 13:23:46] WARNING[1327][C-00000003]:
app_dial.c:2455 dial_exec_full: Unable to create channel of type
'Dongle'
(cause 88 - Incompatible destination)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [9211123456 at DefaultPlan:2]
Hangup("SIP/500-00000003",
"88") in new stack == Spawn extension (DefaultPlan, 9211123456, 2)
exited
non-zero on 'SIP/500-00000003'
Pbx*CLI>
On Tue, Apr 10, 2018 at 1:30 PM, Doug Lytle <support at drdos.info> wrote:
> >>> > exten =>
_9X.,1,Dial(Dongle/dongle800/#31#${EXTEN:1},120,KT)
>
> My suggestion would be to add a pause or two before dialing the phone
> number
>
> exten => _9X.,1,Dial(Dongle/dongle800/#31#ww${EXTEN:1},120,KT)
>
> D(digits): After the called party answers, send digits as a DTMF stream,
> then connect the call to the originating channel (you can also use
'w' to
> produce .5 second pauses). You can also provide digits after a colon - all
> digits before the colon are sent to the called channel, all digits after
> the colon are sent to the calling channel (all digits are sent to the
> called channel if there is no colon present).
>
> Doug
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
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so any ideas, please? On Tue, Apr 10, 2018 at 1:46 PM, Atux Atux <atuxnull at gmail.com> wrote:> after adding the ww: > root at Pbx: /etc/asterisk $ asterisk -rvvv > Asterisk 11.25.3, Copyright (C) 1999 - 2013 D == Using SIP RTP TOS bits > 184 > == Using SIP RTP CoS mark 5 -- Executing > [9211123456 at AllCalls:1] Goto("SIP/500-00000003", > "DefaultPlan,9211123456,1") in new stack -- > Goto (DefaultPlan,92105727105,1) > -- Executing [9211123456 at DefaultPlan:1] Dial("SIP/500-00000003", > "Dongle/dongle800/#31#ww211123456,120,KT") in new stack > [2018-04-10 13:23:46] WARNING[1327][C-00000003]: channel.c:79 > parse_dial_string: Invalid destination '#31#ww211123456' in chan_dongle, > only 0123456789*#+ABC allowed [2018-04-10 13:23:46] > WARNING[1327][C-00000003]: app_dial.c:2455 dial_exec_full: Unable to create > channel of type 'Dongle' (cause 88 - Incompatible destination) > == Everyone is busy/congested at this time (1:0/0/1) > -- Executing [9211123456 at DefaultPlan:2] Hangup("SIP/500-00000003", > "88") in new stack == Spawn extension (DefaultPlan, 9211123456, 2) exited > non-zero on 'SIP/500-00000003' > Pbx*CLI> > > On Tue, Apr 10, 2018 at 1:30 PM, Doug Lytle <support at drdos.info> wrote: > >> >>> > exten => _9X.,1,Dial(Dongle/dongle800/#31#${EXTEN:1},120,KT) >> >> My suggestion would be to add a pause or two before dialing the phone >> number >> >> exten => _9X.,1,Dial(Dongle/dongle800/#31#ww${EXTEN:1},120,KT) >> >> D(digits): After the called party answers, send digits as a DTMF stream, >> then connect the call to the originating channel (you can also use 'w' to >> produce .5 second pauses). You can also provide digits after a colon - all >> digits before the colon are sent to the called channel, all digits after >> the colon are sent to the calling channel (all digits are sent to the >> called channel if there is no colon present). >> >> Doug >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> Check out the new Asterisk community forum at: >> https://community.asterisk.org/ >> >> New to Asterisk? Start here: >> https://wiki.asterisk.org/wiki/display/AST/Getting+Started >> >> 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/20180410/be052922/attachment.html>