Archives : June-2020
Home » 2020 (paged 17)
Please see the above footer, as shown on all list emails. Regards, Antony. — Pavlov is in the pub enjoying a pint. The barman rings for last orders, and Pavlov jumps up exclaiming Damn!I forgot to feed the dog! Please reply to the list; please *do..
I upgraded our Asterisk 13 LTS to Asterisk 16 Certified 16.8 yesterday and converted form SIP to PJSIP using the python script as a start and then mofiying from there. I ran into an issue when testing that incoming calls from MagicJack would go sil..
its possibe to dont start music on hold when caller (from sip operator trunk) press HOLD (i.e. on mobile phone)
Asterisk acts on SDP a=sendonly
i want pass trough media from SIP trunk provider
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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..