[asterisk-users] 11.5.0: blindxfer problems
sean darcy
seandarcy2 at gmail.com
Sun Dec 21 10:09:22 CST 2014
On 12/21/2014 04:42 AM, Patrick Beaumont wrote:
> 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.
>>
OK. I'll figure out DTMF logging, but notice asterisk does recognize
both #1 (blindxfer) and *2 (atxfer), so it recognizes DTMF tones.
sean
More information about the asterisk-users
mailing list