Archives : June-2019
Home » 2019 (paged 18)
I am looking for a consultant that know asterisk in and out including how to troubleshoot sip and rtp. I have a device that this acting very strange and I need t..
i need call time of userB after attended transfer scenario 1) call from Customer to userA 2) userA start consultancy to userB (attended transfer started) 3) userA attended transfer to userB (transfer after consultacy) 4) userA hangup in CEL i have eventt..
–_000_EFCDF2C6785A7B478B3A77A6E7C36369022F3B37E6mailxaccelnet_Content-Type: text/plain; charset=us-asciiContent-Transfer-Encoding: quoted-printableAnyone know how someone can hack an asterisk box and register with every single account on the box. T..
Hi: my ITSP use G.711 with VAD and it can not change the settings. I was using Asterisk 13 but the voice quality is not very good. I dont know if asterisk 16 is good enough to handle this kind of situation? or I still need FreeSWITCH in front of Aster..
allIve got an issue where when I call a number that just plays early media back to me. Instead of hearing the full sequence of tones I hear a short ringing then part of the sequence. What seems odd is that I can see the telephone-event/8000 being pas..
all,We are about to migrate a system that is using AMI at the moment. We want to shift everything from AMI to ARI step by step so both the manager interface and the REST interface have to receive all events.Our idea is to create our own dial app in ..
! I use Asterisk 13.14.1 from Debian repository on a DSL from Deutsche Telekom. Asterisk works well, but I have really often an high delay (I understand it since the other party speak some seconds before he hears my question and answer) and someti..
Dear List Its probably been more than a year now I switched from chan_sip to pjsip. pjsip works much cleaner than chan_sip. But! I have come across a Problem I was not able to solve with Asterisk Dialplan Logic. With pjsip an endpoint can have multi..
This is using Asterisk certified/13.21-cert2, FWIW. I have a hangup handler on an outgoing SIP channel that grabs the SIP status like this: NoOp(keys=${HANGUPCAUSE_KEYS()} sipmsg=${HANGUPCAUSE(${CHANNEL},tech)}) This works fine if the call connects..
We have a need to record audio and allow the user to press any DTMF key to end the recording. Currently were using the AGI command record filewhich does allow us to specify which DTMF keys can end the recording.However we also need to know *which* ..