Want to integrate my cellular service into my asterisk dial plan, so this requires SIP capabilities such as one can get with Vitelity vMobile.Does Google Project Fi or Ting or another offer a SIP integration option, but not advertise it?A drawb..
Hellousing Asterisk 220.127.116.11Current music on hold :myserver*CLI> moh show files Class: default File: /var/lib/asterisk/moh/macroform-robot_dity File: /var/lib/asterisk/moh/macroform-cold_day File: /var/lib/asterisk/moh/reno_project-system File: /var/lib/asterisk/moh/manolo_camp-morning_cof..
Apologies if this is considered off-topic; I suspect the information might benefit a portion of the list.Can anyone point me in a direction to start implementation of E-911 services?Is this just something my upstream should supply, or can I connect..
I need to raise my ptime to 60 on my codecs for outbound calls. To that effect, I add on the endpoint disallow=all allow=ulaw:60and also use_avpf : false use_ptime: trueBut the invites always leave with ptime:20. It used to work fine in the old SIP chann..
Is there any easy way to add a custom subscribe handler? I have a set of users with Polycom phones that attempt to Events that Asterisk/PJSIPdoesnt recognize, call-info and as-feature-event. It just generates a warning, but it got me wondering if I co..
fail2ban does not ban offending IP.NOTICE chan_sip.c: Registration fromuser3 failed for offending-IP:53417 – Wrong passwordNOTICE chan_sip.c: Registration fromuser3 failed for $B!F(Boffending-IP:53911 -Wrong password systemctl sta..
From the blog…http://blogs.asterisk.org/2017/03/01/pjproject-2-6/This week, we’re pleased to say that we’ve updated the Asterisk 13, 14 and master branches’ bundled version of pjproject to 2.6.Here’s a short recap of the steps we took to ..
Hi.Im having problems with the Dial() application when I use full SIP account details in it.Im looking at the OReilly book https://www.amazon.co.uk/dp/1449332420 on page 135, where it says The Dial() application also allows you to connect to a rem..
Asterisk 13.3.2I change the allowed codec from ulaw to g729 in sip.conf and enter sip reload on the console, but calls continue to use ulaw until restart.Before reload:lc10*CLI> sip show settings Global Signalling Settings:—————..
Asterisk 13.3.2The console command sip show settings shows the allowed codecs in the Global Signalling Settings but does not include the packetization setting.Similarly, both core show channel and sip show channel will show the codec(s), but not ..