I am looking for CDR reporting solution? Any suggestions? I am using Asterisk 13.13.1I would like a report on number of calls per extension.Tha..
I have a setup which is not working right now:Provider – DSL-Router (192.168.2.1) – Bintec-Router (10.17.46.66) – Asterisk (10.17.46.99)Note: upgrade your network to wireless AC for reliable, high-speed Wi-Fi up to 1200Mbps. Enjoy more HD gaming ..
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..
all,we have a running Asterisk 11.25.1 in a VM (qemu/kvm) OS being Debian 7.11 (wheezy), the host OS being the same.Problem: when we restart the server (eg host + VM), all customers Asterisk connecting without a VPN (doesnt matter which Asterisk versi..
Im currently testing a so-called VQ RTCP-XR feature from a a SIP hardphone.When a phone has enabled this feature, it would send a SIP PUBLISH to its SIP Server letting this server dispatch to whatever is needs to.These messages are sent during ca..
I have a user that prefers Soft SIP phone install on his laptop, for security reasons I have enable TLS on our Asterisk server to support TLSauthentication, It works well with hard phones. Has anybody in this forum use SIP Soft phones with TLS authenticat..
Now that the g729 patents have expired, how do we use g729 in Asterisk?Will Digium be releasing a g729 codec for free use or do we download the free codec off the Internet now that we can use it without moral or legal res..
I recently upgraded from Asterisk 1.8 to Asterisk 13. Now users are starting to complaint about packets loss, conversations are choppy! I dont even know where to start looking! Choppy conversations happened within users. I am using sip.conf type=friendcontext=sip-phonecall-limit=2trustrpid=nocallerid=d..