I am running Asterisk 11.17 with DAHDI 2.9.0 and an OpenVox A800P with 8x analog POTS lines coming into my Asterisk server from the phone company. Internally, I have about 180 SIP clients defined in sip.conf. What appears to be happening is that if exist..
Author : Andrew Martin
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Running Asterisk 11 on CentOS 6.x using the DAHDI module with 8x PSTN analog phone lines for outside connectivity. Internally, I am using several models of Yealink SIP phones (e.g SIP-T32G) on a dedicated VoIP network, 192.1..
I am running Asterisk 11 on CentOS 6.x. I have configured several queues as follows in extensions.conf:exten => s,1,Queue(myqueue,rtnC,18)same => n,Background(user_unavail)same => n,WaitExten(10)exten => 1,1,Voicemail(1111@my-vm,s)This rings the pho..
I am running Asterisk 11 on CentOS 6.4 with SIP clients (Yealink phones). All the SIP clients are on a LAN, so no NAT is involved. I have been experiencing an intermittent problem where a call will be successfully answered, but then dropped by Aster..
I am running Asterisk 11 on CentOS 6.4 with about 150 local SIP clients on a LAN. The SIP clients are a mixture of Yealink phones (e.g SIP-T32G, SIP-T42G, etc). I have configured the system as follows:sip.conf:[169]secret1111dtmfmode=rfc2833directmedia..
I am running Asterisk 11.12.0 on CentOS 6.4. The asterisk server and internal phones are located on the 10.10.32.0/21 LAN subnet. I have many internal SIP phones, which appear to be working correctly. I have a few external phones (Yealink SIP-T32G..