I have been fighting all night with version 1.8 and have not found a
way to do this with any command or Perl AGI->command. I need to play a
file and wait until the customer presses at least $maxdigits to
return, BUT, the file must continue playing until $maxdigits is
received or $timeout has expired. So far I found impossible to achieve
this functionality. Am I missing something?
Hello, voicemail password is not getting changed through phone handset while IVR indicates that password has been changed. During google I found that uniqueid column must not be changed so it is not changed. Please guide on this. During debug log I found below but in mysql db new password is not getting updated, [Jun 15 13:54:07] VERBOSE file.c: --
[Jun 15 13:54:10] DEBUG channel.c: Set channel SIP/123-00000005 to write format ulaw[Jun 15 13:54:15] VERBOSE file.c: --
I want to use rtptimeout function on asterisk 1.4 but any docs I read,
it is said that I need to configure it in sip.conf file,
But can I use rtptimeout in users.conf file or do I need to configure
all the SIP accounts on sip.conf file before I can use rtptimeout ? thanks.
I am trying to achieve call record file and play the file to the user using
phpagi script. I am using the following code $rand="/var/sounds/145712900"; $agi->record_file($rand, "gsm", "0123456789*#", -1, 0, true, 0); also tried $agi->record_file($rand, "wav", "0123456789*#", -1, 0, true, 0); the file is created in the directory but when I try to play the sound I get
the following error incase of gsm
[May 24 14:00:07] WARNING[C-0000019c]: file.c:492 filehelper: File
/var/sounds/145712900.gsm detected to have zero size. and the following error incase of wav file…
I am interested to know if there is any application or way to help me for
this Scenario: When we put Callers in Q, the MOH will stop for announcements!
how if we able to increase the MOH (RT/TX) and then play any announce with
greater RX/TX ( and there so louder ) on the channel! without stoping MOH?
I'm using asterisk 18.104.22.168 and fax for asterisk
I'm receiving a multi page fax using the ReceiveFax application with the
d and f options. The fax is being written as a tiff file but only the first page of the
fax is in the tiff file The asterisk server knows there are multiple files in the fax as can be
seen by the success message FAX session '8' is complete, result: 'SUCCESS' (FAX_SUCCESS), error:
'NO_ERROR', pages: 2, resolution: '204x98', transfer rate: '14400',
remoteSID: 'XXXXXXXXXXX' Here's the part of the dial plan that receives the fax…
I would like capture SS7 Error Code in CDRs, Specifically of outbound call
from the asterisk. calls generated using .call file. In extension.conf extens gets excuted on successful call only , So that on
h extension reason of hangup is captured. But i am not aware of any
provision that capture on Unsuccessful call. please guide on this or suggest any patch. Thanks
Tough Freepbx is not the main focus of this list, may I ask if Freepbx
and its End Point Manager module can work in an environment with an
HTTP proxy ? In my testing, everything works OK but one thing: I can't upload End
Point product list : in End Point Configuration tab, when I click over Check for Updates
button, I get this:
Not able to connect to repository. Using local master file instead.
Aborting Brand Downloads. Can't Get Master File, Assuming Timeout Issues!
Learn how to manually upload packages here…
I have a user that has reported that his HT286 doesn't have a ring tone; it
just buzzes. Also, the ring doesn't sound right. I can see in the provisioning file where this is configured, but I'm not a
musician so I don't know what to put in for a value. Does anyone have suggestions for rational sounding dialtone and ring
indication for North America? TIA,