Used it with jitsi and linphone softphones, works just OK.Just for testing i did a video-call on the loop-back, great test tool for showing the influence of (limited-) bandwith / latency.Ideal for..
From: firstname.lastname@example.org Reply-to: Asterisk Users Mailing List – Non-Commercial Discussion To: email@example.com Subject: [asterisk-users] НА: asterisk-users Digest, Vol 105, Issue 40 Date: Tue, 07 May 2013 07:53:53 +0600 help —–Origi..
all,I had to re-install a new machine and noticed that by default, ip was only listening on 0.0.0.0, thus ipv4 only. Easily changed.However, when looking at iax.conf, I found here the same, but it looks like iax is still ipv4 only?If i change bindaddr2.168…
all,Finally i got hold of some bt-dongles that seems p[retty stable, the asus-bt211.After installing them, i rebuild 11.3-rc1 added mobile.conf (bt-addres and blackberry address) and mobile show devices is showing me that the BT-link is up, and rema..
all,Im caught up in a struggle between people how can not make up their mind… Half way implementing a asterisk farm and they come up with another feature theyve seen in kamaillo.What he showed me was this: three registered sip users, a) one chan..
all,Is there a simple way of disabling regular expressions in the dialplan?Reason for asking, is that people hate to remember numbers. So i want to use there full smtp address as as their extension.In stead of 12345678 i would like to use firstname.lastname@example.org..
Are there any thoughts about how cpu-expensive motif is?Does it only translate SIPjingle (during call-setup) if so, impact will probably neglectible.or does asterisk remains constantly in between the data-stream?In that case, it might be something..
Perhaps can someone tell me if i had the wrong expectancies….If one sip-clinet only supports GSM-codec, and another only supports g711-U, they still can call each other and asterisk does the transcoding Correct?If i try to do the same with an AV-ca..
all,For one of my inverstigations it looks like im back to square oneIm trying to accept an incoming XMPP call and forward it conditionally to a sip, isdn, or voicemail. No google is involved as i use a local xmpp server (ejabberd)I was experiment..
all,Been reading about chan_motif / chan_xmpp in the wikis for 1.8, 10 and11 version of Asterisk. In each example i got the impression that the asterisk server is registering on a XMPP server as a single user with the credentials as specified in jabber.conf.Inst..