search for: dialtim

Displaying 5 results from an estimated 5 matches for "dialtim".

Did you mean: dialtime
2010 Nov 04
0
[backport] Allow app_dial to play 'indication tone while ringing' back ported to 1.6.2.X
...ps://reviewboard.asterisk.org/r/448/ We use it to let the caller know if the extension being called is INUSE using the following excerpts: indications.conf: [us] ringinuse = 440+480/400,0/200,440+480/400,0/2000 extensions.conf: [macro-stdexten-v3] . . same => n,ExecIf($["${DIALTIME}" = ""]?Set(DIALTIME=${DEFAULTDIALTIME})) . same => n,ExecIf($["${DIALARGS}" = ""]?Set(DIALARGS=${DEFAULTDIALARGS})) . same => n,Set(DEVICES=${ARG4} . same => n(checkdevicestate),ExecIf($["${DEVICE_STATE(${CUT(DEVICES,&,1)})}" = &quot...
2013 Aug 22
2
How to get the original SIP result code
B.H. Hello, i'm using AMI Originate action (with async=true) to send outgoing calls to a SIP trunk (using asterisk-java library to connect to AMI). The problem is that in case of failed originate, OriginateResponse event is returning only the reason code which is sometimes not sufficient to determine the real cause of failure. Also, there's no way to link between the channel that was
2006 Mar 26
0
RE: Hopefully a Simple Question?
...pts and is transferred to an outgoing channel (B) using the Dial cmd. That part works perfectly! For billing I'd like to be able to charge for the time that the first caller is connected to the callee on channel (B) so I can pass on my own outgoing voip costs. How do I do this? I can get the DIALTIME and END time of the call from the cdr but there doesn't seem to be a way of capturing the ANSWERTIME of channel (B) from the dialplan. Any suggestions would be greatly appreciated. clint_in_sydney ************* Clint, Use the forkcdr command in the extension logic right before you connect...
2009 Mar 24
0
Unrecognized prilocaldialplan error when dialing a SIP call to a PRI trunk
...odifier: a Unrecognized prilocaldialplan TON modifier: b Unrecognized prilocaldialplan TON modifier: c Where abc is the SIP username. Is this a bug where the SIP username is somehow becoming a dialed number prefix, as described here in chan_dahdi.conf ; pridialplan may be also set at dialtime, by prefixing the dialled number with ; one of the following letters: ; U - Unknown ; I - International ; N - National ; L - Local (Net Specific) ; S - Subscriber ; V - Abbreviated ; R - Reserved (should probably never be used but is included for completeness) I...
2005 May 25
1
LiveVoip does not like customers anymore, ....
> You have been replied to - we do not use digital certs, we do not > reply when you have some sort of Spam blocker. This time I am > responding even though that is not policy. > It seems it is their policy not to answer. FYI info I tried to get an account with them a week ago. I did not get any information how to setup, just that they cashed my credit card. Several calls to them