Archives : May-2020
how can I change the color of the asterisk prompt to red ? I read in the wiki that I can use %Cn[;n] https://wiki.asterisk.org/wiki/display/AST/Asterisk+CLI+Configuration But what does this mean ? There is no example how to actually use it. where..
I got the response below from a provider. How do I extract the Identity header and apply it to the next INVITE? Is it possible at all with PJSIP?SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 172.16.7.254:52169;rport=52169;received=XX.205.172.89;branch=z9hG4bK-524287-1—129f4244aaba9f04Call-..
–_000_EFCDF2C6785A7B478B3A77A6E7C36369024F0F2D17mailxaccelnet_Content-Type: text/plain; charset=us-asciiContent-Transfer-Encoding: quoted-printableAnyone know how to set the To:in an invite for PJSIP to custom settings. I got the from to be the ..
I am getting this error on CentOS 8CCdahdi-linux-complete-3.1.0+3.1.0/linux/drivers/dahdi/xpp/xpp.mod.oLD [M] dahdi-linux-complete-3.1.0+3.1.0/linux/drivers/dahdi/xpp/xpp.koCC dahdi-linux-complete-3.1.0+3.1.0/linux/drivers/dahdi/xpp/xpp_usb.mod.oLD ..
https://wiki.asterisk.org/wiki/display/AST/STIR+and+SHAKENThe Wiki above is misleading in what Stir-Shaken means and how it works. End users cannot get a certificate, they cannot self-certify their calls. Somebody completely misunderstood the mod..
Here is some material for you to read. Rest assured that this is real. https://www.fcc.gov/call-auth..
Everybody, Ive had a request from my manager that I figure out how to get our Asterisk 13.x system using chan_sip to be able to display on the Polycom VVX series phone display (firmware 5.9.5), when an extension is called and the person on the ot..
In a few weeks, no SIP call is going to terminate unless they are signed properly, as mandated by law.We are in the business of Stir-Shaken, signing calls, as an FCC-approved provider. A big differentiator between our service and the rest: we are ..
I have an endpoint with multiple phones registered as aor contacts.When I attempt to originate a call it will only ring one of the phones.Is it possible to ring multiple phones as a single endpoint.First phone to answer wins the call and all others terminated?Aga..
all,We would like to pull the RTT of registered endpoints from MySQL for use in a webportal. However it doesnt appear asterisk tables this by default like chan_sip did.Ive found some information [1] that a modification of sorcery.conf can get it writ..