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Some
additional information: It
looks like it’s not even waiting for any additional DTMF signals as it
immediately tries to find the extension as soon as it gets the first digit: Sep
3 22:54:42 VERBOSE[15732] logger.c: -- Playing 'pbx-transfer' (language
'en') Sep
3 22:54:44 DEBUG[15738] chan_sip.c: * Detected inband DTMF '*' Sep
3 22:54:48 VERBOSE[15732] logger.c: -- Unable to find extension '*' in
context 'from-internal' Sep
3 22:54:48 VERBOSE[15732] logger.c: -- Playing 'pbx-invalid' (language
'en') Sep
3 22:54:52 VERBOSE[15732] logger.c: -- Stopped music on hold on
SIP/801-b190 I
was watching this log as it was generated … I dialed *801 but it only got the
first DTMF. This
was tested by picking up my SIP handset, dialing 7777, then dialing 801 which
did an outbound call to my cell phone. When I answered my cell phone, I pressed
# and was issued the prompt “Transfer” … I dialed *801, but as I said above it’s
not getting the additional DTMF tones. If
I dial *in* to the system from my cell phone and dial 801 it will send
me to the SIP extension. From there, I can successfully hit # and transfer the
call to *801 with no problems. It only seems to be when the call is
transferred to an external phone. This
isn’t an issue with the transfer system not finding the extension due to
improper context or something… this is an issue with Asterisk recognizing DTMF
tones during transfer. Any
thoughts? George From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of George A.
Roberts IV This
isn’t [EMAIL PROTECTED] functionality. It’s basic Asterisk functionality. George From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of William
Piper This is not the list for [EMAIL PROTECTED]. For
questions about [EMAIL PROTECTED] functionality, they have their own
mailing list. bp On 9/3/06, George A. Roberts IV
<[EMAIL PROTECTED]>
wrote: No one has any ideas? From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of George
A. Roberts IV Hello all, Using [EMAIL PROTECTED] and are using its Follow Me functionality to bounce
calls to my cell phone when I'm not in the office. Would like to be able
to use the blind transfer functionality from my cell phone when I receive a
call in from Asterisk but am not having much luck getting it to work… I can press ## (that's what it's set to in features.conf) and get the
"Transfer" prompt from Alison and the dialtone. But no matter
what I punch in, it seems that Asterisk is only getting the first digit I
press. -- Unable to find extension '*' in context
'from-internal' -- Playing 'pbx-invalid' (language 'en') -- Stopped music on hold on SIP/801-b190 -- Started music on hold, class 'default', on
SIP/801-b190 -- Playing 'pbx-transfer' (language 'en') -- Unable to find extension '8' in context
'from-internal' -- Playing 'pbx-invalid' (language 'en') The first one there I tried to punch in *801 to transfer to voicemail.
The second one I punched in 802 to transfer to another extension. Each
time, it's only getting the first digit. Anyone seen this before? Any thoughts? Thanks! Regards, George A. Roberts IV President & CEO, Interjuncture Corp.
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