Any ideas why a VPS, cloned from another instance (DigitalOceandroplets if it matters), wont run on the new instance?Everything else is the same; region, memory, disk, hypervisor version etc.And everything else runs, just not Asterisk, unless I d..
I just made a asterisk / dahdi fresh install on Debian 8.4, and ended up with the following packages :$ sudo dpkg -l|grep -Ei dahdi|asterisk|libpriiiasterisk 1:11.13.1~dfsg-2+b1 amd64Open Source Private Branch Exchange (PBX)iiasterisk-config1:11.13.1~dfsg-2allConfigurat..
All,Ive setup a Digium G100 VoIP gateway to replace an internal PCI VoIP card in our Asterisk PBX.When using the VoIP card the callerid entries listed in sip.conf were displayed when calling someone over the PSTN.Now, however, though the gateway it j..
I am turning up a PJSIP Endpoint and am having problems when they send an INVITE to my server. Asterisk is returning a 421 Extenstion Required. Sinceextension means different things in the SIP stack versus Asterisk, Idont know what it is complain..
Hi! I wish you all e Happy New Year first! Allthough, Im relative new to Asterisk, I got our server up and Running, Softphones, ISDN, and a brand new Snom 821 are working flawlessly. 🙂 Platform is Debian 8/Asterisk Packages (11) from Debian Repo. ..
I am trying to set up a default outbound endpoint for my Asterisk 13.6.0PBX, and per https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Configuration_res_pjsip, I do in pjsip.conf:[global]default_outbound_endpoint=SillyEndpoint…[SillyEndpoint]type=endpointetc.Howev..
I have a customer where we are putting an Asterisk PBX in front of a legacy (non-VoIP) PBX. One of the requirements it that the Asterisk PBX have 2 PRI ports (one towards the legacy PBX and one towards the carrier) with the ability to go to pass thro..
, 2 weks ago I asked questions about PJSIP and T.38 but got no replies. I upgraded Asterisk to git as of yesterday (309dd2a), and Im still having the same issues. In the trace below, Im sending a fax from Hylafax server through iaxmodem on Asterisk..
Hi!I am running an Asterisk PBX 11.6-cert10 with about 20 SIP phones and recently one of the phones (Snom 720) always returns 486 Busy Here when calling anonymously. Its only a single phone, the rest works as expected. I checked the phones settings ..
Long story short – I have an ancient Britsh Telecom phone attached to my Asterisk PBX via Dahdi. It works beautifully, receiving calls, and the call quality is excellent. However, dialling out is impossible, as Asterisk consistently mis-reads the num..