Archives : November-2019
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I would like to offer end users in a LAN, asking for this (why ? I dont know) the capability to use a laptop (along or in replacement of hardphones) to emit and receive PSTN calls.PSTN pass through a plain SIP trunk which does not support video (..
Gang Next Problem which occurs. In Switzerland this is the common using form SIP Signaling: P-Asserted-Identity: Contains the provider provided and screened phone number which is the legal origin of the call. The origin which is to be billed for ..
One more Problem I stumbled upon. Using Asterisk in a TSP environement. Incomming IC Calls are e164 and have a NPRN (Routing Number) prefixed. Example: +4198055615995555 +41 country prefix 98055 Routing Prefix 615995555 effective phone number Calls rou..
Following [1], you get precious help for webRTC installation.Something that is missing there, though, is a note expliciting/etc/asterisk/keys files ownerships and modes.As people are either running asterisk as root:root, asterisk:root and others or..
Ive installed a new Asterisk 17.0.0 on a Debian Buster system.This Asterisk instance is run by asterisk user (and group). Ive got:# ls -l /etc/asterisk total 68-rw-r–r– 1 asterisk asterisk501 nov.18 19:12 asterisk.conf-rw-r–r– 1 asterisk asterisk..
Gang Yes, big project on the rise to do things better / more flexible than our existing commercial TSP switch. During call screening process, we would like to allow customers to send the original callingID in a attended call diversion scenario. F..
Gang To increase security against phished passwords and similar attacks, we consider offering customers to define IP ranges (or GeoIP locations) from which their dynamic registrations are being accepted. I can already look at the source IP in the d..
We have a customer using Avaya.Currently, they are using chan_sip.We are working to migrate them to PJSIP. I have not been filled in on the exact scenario.I suspect they have some auto forward feature on the number.Rather than their Avaya transferr..
Hello.I have a problem with the native Android SIP client, not acknowledging the call. Sent a message to the list for some weeks ago containing a sip debug log, but it only got stuck in moderation queue due to too large size (and it said I would ..
were running a small Asterisk appliance on a PCengine APU2C4. Base operating system is FreeBSD 12-STABLE, most recent incarnation as of today.Since update of port net/asterisk16 to the latest bug fix revision 16.6.1, we face a severeslowdown of everyth..