Tag : sip
Home » Posts tagged sip" (paged 6)
folks,We have an issue with several Cisco SPA512G phones connected to an Asterisk platform where several users hear loud, random beeps during calls to external recipients. The noises are akin to button press tones, are very loud and a significant annoyan..
Dear asterisk-users,I have listened that a diaplan on Asterisk can extract information from proprietary SIP messages header fields. That is, if Asterisk receives a SIP message with a modified HEADER (containing proprietary fields) , is it possible..
!Very strange… I ran the Asterisk CLI for other tasks, and suddenly I got this message:== Using SIP RTP CoS mark 5– Executing [000972592603325@default:1] Verbose(SIP/192.168.20.120-0000002a, 2,PROXY Call from 0123456 to 000972592603325) in new stac..
Hello!We have asterisk connected over PRI no our phone network, so Im avaya PBX user. Asterisk connects to another avaya system over h323.Connection can be shown asavaya–PRI-asterisk–h323-avayaWhen I do call as avaya user I see name of remote end a..
What is your experience: if you plan to make a test SIP call to check voice quality overa connection what would be the best call duration? The point is that we should have a call long enough to be able to catch/hear impairments that the connection ..
We have a FreePBX-12 / Asterisk-12 setup that supports about 24extensions, most internal Snom870s but six or so external (Jitsi-2.8). we use TLS and SRTP everywhere on our side of the fence.The server host is a dedicated atom(tm) box using the Free..
list,Im hoping that you could read through this mail and give me some tips on how to improve my setup (functionality, security, really anything). Its my first Asterisk installation and meant for simple home use.I installed Asterisk 11 on an OpenWrt Barr..
Hey,I am running default Asterisk 11.16.0 on a FreeBSD-Machine. I need to register to several other SIP-Services (actually 3):short sip.confregister => XX@a register => XX@b register => XX@cIf I remember correctly this worked quite well, but I now chec..
Im testing Asterisk at home, crummy connection.Skype works fine for me, but every SIP client, even without using Asterisk, fails to connect.Thats ok.Is swapping out SIP for Skype a big deal?Heh, well, I guess its dead:http://www.digium.com/en/products/software/skype-for-asteris..
Background: I dabbled with asterisk years ago, and more recently have more-or-less functioning IncrediblePBX systems for experimenting, but I want to understand more so Im now working with distro packages(Ubuntu) and hand edited configurations file..