Could you help me to summarize what is SIP Early Media useful for ?I was thinking of:- Passing error messages to caller,- Custom ringing tones to caller.Did I miss something ?Be..
Author : Olivier
Home » Articles posted by Olivier (paged 15)
Im trunking with an ITSP that, when treating an outbound to an unknown destination, either:- send a SIP error code (I cant be more explicit, at the moment),- or cast a pre-recorded audio message using Early Media.At the same time, Im also trunking w..
Im giving a try to Resource List Subscriptions feature also called BLFList in several phone vendors documentation (see [1]).I could successfully configured this feature woth Yealink phones but Ive got some issues with Aastra phones (6757i with 3…
How should I configure Asterisk (13.7.0) to get persistently PJSIP SIPmessages in a log file and not in console ?I would expect adding debug=yes in pjsip.conf to produce the same output as pjsip set logger on. Am I understanding correctly ?Be..
Im setting up an active-passive Asterisk solution on Debian Jessie platforms. Im using heartbeat package.As Im not yet familiar with either systemd or heartbeat, Ive got a couple of questions:1. At the moment, Im using /usr/share/heartbeat/hb_stan..
On a new Raspberry2 with Asterisk 11.13.1 binary-installed, ReceiveFAXexits with FAILED status.Each attempt is made with something like:channel originate Local/12345@faxgateway application SendFAX myfile.tifIm not sure if failure mostly comes from Send..
Ive got a system on which Asterisk 11 was first installed. Then Asterisk 13 was tentatively installed. Then Im trying to re-install Asterisk 11.Every install was made from Asterisk source.Now Asterisk 11 seems correctly installed but basic dialplan reloadcomm..
Im currently studying what is needed to implement IMAP Voicemail with Asterisk 11 and up.More precisely, I would like to let users check voicemail with their smartphone from outside (ie not connected to LAN).My first questions are:1. What happens..
2015-04-30 17:45 GMT+02:00 Richard Mudgett :Simply adding a CONNECTEDLINE(num) statement made it !Thank you very much for..
I recently gave CONNECTEDLINE function a try on an Asterisk 11 setup with a couple of SIP phones.When a SIP phone dials an other one, with a CONNECTEDLINE statement in its dialplan, I noticed that Asterisk update callers information using a Remote-Party..