On 12/19/2014 09:42 AM, Rusty Newton wrote:> On Wed, Dec 17, 2014 at 1:09 PM, sean darcy <seandarcy2 at gmail.com> wrote: >> I've got a confbridge set up which works if dialed locally: >> >> -- Executing [266 at internal:1] Answer("DAHDI/1-1", "") in new stack >> -- Executing [266 at internal:2] SendDTMF("DAHDI/1-1", "1") in new stack >> -- Executing [266 at internal:3] ConfBridge("DAHDI/1-1", "1") in new stack >> -- <DAHDI/1-1> Playing 'conf-onlyperson.ulaw' (language 'en') >> ....... >> >> >> extensions.conf: >> >> [globals] >> ....... >> GOTO_ON_BLINDXFR="internal,266,1" >> >> features.conf: >> >> [featuremap] >> blindxfer => #1 >> >> But: >> >> -- Executing [s at DialOut:14] Dial("DAHDI/1-1", >> "motif/xxxx/+1234567890a at voice.google.com,,rTt") in new stack >> -- Called motif/xxxx/+1234567890a at voice.google.com >> -- Motif/+1234567890a at voice.google.com-688c is proceeding passing it to >> DAHDI/1-1 >> -- Motif/+1234567890a at voice.google.com-688c answered DAHDI/1-1 >> -- Started music on hold, class 'default', on >> Motif/+123456789a at voice.google.com-688c >> -- <DAHDI/1-1> Playing 'pbx-transfer.ulaw' (language 'en') >> [Dec 17 09:46:59] WARNING[19083][C-000000be]: features.c:2550 >> builtin_blindtransfer: No digits dialed. >> -- <DAHDI/1-1> Playing 'pbx-invalid.ulaw' (language 'en') >> >> I'm expecting the blind transfer to GOTO internal,266,1. >> >> If I input 266 at the transfer dial tone, the blind transfer occurs. >> >> Do I have this set up incorrectly? > > https://wiki.asterisk.org/wiki/display/AST/Asterisk+Standard+Channel+Variables > > "${GOTO_ON_BLINDXFR} - Transfer to the specified > context/extension/priority after a blind transfer (use ^ characters in > place of | to separate context/extension/priority when setting this > variable from the dialplan)" > > Try using ^ characters as it mentions there. >Thanks for the response, but no joy: == Setting global variable 'GOTO_ON_BLINDXFER' to 'internal^266^1' <DAHDI/1-1> Playing 'pbx-transfer.ulaw' (language 'en') [Dec 20 15:12:03] WARNING[12336][C-00000012]: features.c:2550 builtin_blindtransfer: No digits dialed. sean
On 12/20/2014 03:22 PM, sean darcy wrote:> On 12/19/2014 09:42 AM, Rusty Newton wrote: >> On Wed, Dec 17, 2014 at 1:09 PM, sean darcy <seandarcy2 at gmail.com> wrote: >>> I've got a confbridge set up which works if dialed locally: >>> >>> -- Executing [266 at internal:1] Answer("DAHDI/1-1", "") in new stack >>> -- Executing [266 at internal:2] SendDTMF("DAHDI/1-1", "1") in new >>> stack >>> -- Executing [266 at internal:3] ConfBridge("DAHDI/1-1", "1") in >>> new stack >>> -- <DAHDI/1-1> Playing 'conf-onlyperson.ulaw' (language 'en') >>> ....... >>> >>> >>> extensions.conf: >>> >>> [globals] >>> ....... >>> GOTO_ON_BLINDXFR="internal,266,1" >>> >>> features.conf: >>> >>> [featuremap] >>> blindxfer => #1 >>> >>> But: >>> >>> -- Executing [s at DialOut:14] Dial("DAHDI/1-1", >>> "motif/xxxx/+1234567890a at voice.google.com,,rTt") in new stack >>> -- Called motif/xxxx/+1234567890a at voice.google.com >>> -- Motif/+1234567890a at voice.google.com-688c is proceeding >>> passing it to >>> DAHDI/1-1 >>> -- Motif/+1234567890a at voice.google.com-688c answered DAHDI/1-1 >>> -- Started music on hold, class 'default', on >>> Motif/+123456789a at voice.google.com-688c >>> -- <DAHDI/1-1> Playing 'pbx-transfer.ulaw' (language 'en') >>> [Dec 17 09:46:59] WARNING[19083][C-000000be]: features.c:2550 >>> builtin_blindtransfer: No digits dialed. >>> -- <DAHDI/1-1> Playing 'pbx-invalid.ulaw' (language 'en') >>> >>> I'm expecting the blind transfer to GOTO internal,266,1. >>> >>> If I input 266 at the transfer dial tone, the blind transfer occurs. >>> >>> Do I have this set up incorrectly? >> >> https://wiki.asterisk.org/wiki/display/AST/Asterisk+Standard+Channel+Variables >> >> >> "${GOTO_ON_BLINDXFR} - Transfer to the specified >> context/extension/priority after a blind transfer (use ^ characters in >> place of | to separate context/extension/priority when setting this >> variable from the dialplan)" >> >> Try using ^ characters as it mentions there. >> > > Thanks for the response, but no joy: > > > == Setting global variable 'GOTO_ON_BLINDXFER' to 'internal^266^1' > > <DAHDI/1-1> Playing 'pbx-transfer.ulaw' (language 'en') > [Dec 20 15:12:03] WARNING[12336][C-00000012]: features.c:2550 > builtin_blindtransfer: No digits dialed. > > > sean > >I also tried setting up a transfer as an applicationmap. conference => *7,peer/both,ConfBridge,1 Seems to load: features reload == Parsing '/etc/asterisk/features.conf': Found == Registered Feature 'conference' == Mapping Feature 'conference' to app 'ConfBridge(1)' with code '*7' but when the caller dials *7, there's no action, Nothing in the cli. The dtmf is just sent to the callee. Also tried having the callee dial *7, same result. Any help appreciated.
Patrick Beaumont
2014-Dec-21 09:42 UTC
[asterisk-users] 11.5.0: blindxfer problems [Spam score:10%]
Have you enabled DTMF logging and seen the DTMF codes being recognised by Asterisk? I had a bunch of soft phones that I had to change to using ?sip info? for the DTMF signalling as the RFC signalling was not always being recognised. This would cause transfers to appear as if the user had not dialled any digits. On 20/12/2014 20:52, "sean darcy" <seandarcy2 at gmail.com> wrote:>On 12/20/2014 03:22 PM, sean darcy wrote: >> On 12/19/2014 09:42 AM, Rusty Newton wrote: >>> On Wed, Dec 17, 2014 at 1:09 PM, sean darcy <seandarcy2 at gmail.com> >>>wrote: >>>> I've got a confbridge set up which works if dialed locally: >>>> >>>> -- Executing [266 at internal:1] Answer("DAHDI/1-1", "") in new stack >>>> -- Executing [266 at internal:2] SendDTMF("DAHDI/1-1", "1") in new >>>> stack >>>> -- Executing [266 at internal:3] ConfBridge("DAHDI/1-1", "1") in >>>> new stack >>>> -- <DAHDI/1-1> Playing 'conf-onlyperson.ulaw' (language 'en') >>>> ....... >>>> >>>> >>>> extensions.conf: >>>> >>>> [globals] >>>> ....... >>>> GOTO_ON_BLINDXFR="internal,266,1" >>>> >>>> features.conf: >>>> >>>> [featuremap] >>>> blindxfer => #1 >>>> >>>> But: >>>> >>>> -- Executing [s at DialOut:14] Dial("DAHDI/1-1", >>>> "motif/xxxx/+1234567890a at voice.google.com,,rTt") in new stack >>>> -- Called motif/xxxx/+1234567890a at voice.google.com >>>> -- Motif/+1234567890a at voice.google.com-688c is proceeding >>>> passing it to >>>> DAHDI/1-1 >>>> -- Motif/+1234567890a at voice.google.com-688c answered DAHDI/1-1 >>>> -- Started music on hold, class 'default', on >>>> Motif/+123456789a at voice.google.com-688c >>>> -- <DAHDI/1-1> Playing 'pbx-transfer.ulaw' (language 'en') >>>> [Dec 17 09:46:59] WARNING[19083][C-000000be]: features.c:2550 >>>> builtin_blindtransfer: No digits dialed. >>>> -- <DAHDI/1-1> Playing 'pbx-invalid.ulaw' (language 'en') >>>> >>>> I'm expecting the blind transfer to GOTO internal,266,1. >>>> >>>> If I input 266 at the transfer dial tone, the blind transfer occurs. >>>> >>>> Do I have this set up incorrectly? >>> >>> >>>https://wiki.asterisk.org/wiki/display/AST/Asterisk+Standard+Channel+Var >>>iables >>> >>> >>> "${GOTO_ON_BLINDXFR} - Transfer to the specified >>> context/extension/priority after a blind transfer (use ^ characters in >>> place of | to separate context/extension/priority when setting this >>> variable from the dialplan)" >>> >>> Try using ^ characters as it mentions there. >>> >> >> Thanks for the response, but no joy: >> >> >> == Setting global variable 'GOTO_ON_BLINDXFER' to 'internal^266^1' >> >> <DAHDI/1-1> Playing 'pbx-transfer.ulaw' (language 'en') >> [Dec 20 15:12:03] WARNING[12336][C-00000012]: features.c:2550 >> builtin_blindtransfer: No digits dialed. >> >> >> sean >> >> > >I also tried setting up a transfer as an applicationmap. > >conference => *7,peer/both,ConfBridge,1 > >Seems to load: > >features reload > == Parsing '/etc/asterisk/features.conf': Found > == Registered Feature 'conference' > == Mapping Feature 'conference' to app 'ConfBridge(1)' with code '*7' > >but when the caller dials *7, there's no action, Nothing in the cli. The >dtmf is just sent to the callee. > >Also tried having the callee dial *7, same result. > >Any help appreciated. > > >-- >_____________________________________________________________________ >-- 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