![]() ::17:09:25.701 ERROR 1073742997 RTMP 23060 Received Invalid Command on NetConnection:_OnServerHeartBeat, Message Dropped ::17:09:11.794 INFO 1073742997 RTMP 23060 Received onStatus code classType description details ::17:09:11.792 INFO 1073742997 RTMP 23060 Received onStatus code classType description details In SIP traces we can see that AMG is not answering the call which means I am not getting 200-OK after INVITE and 100-TRYING. I can dial to AMG and can listen to the playback so SIP between my phone and AMG is working. Please note that both AMG and AMS are installed on the same server and I am testing it on a local LAN. Can some one please advise what exactly I am missing out here ? I have pasted AMG logs, RTMP capture and SIP capture. Please note that a new stream (from AMG) is created and published in AMS as I can see from the wireshark dump and AMS admin console but some call goes un-answered. Now my AMG leg service is connected with AMS but when I dial AMS extension from my sip phone, AMG doesn't answer or bridge the RTMP/SIP-RTP audio stream. This will remove blank lines only from nf&nf(these are main files where the configuration changes happens repeatedly) Posted in Asterisk, Technical, VOIP Tagged Asterisk, asterisk-gui, blank lines.I trying to integrate AMG with AMS so that a SIP Phone user can join a virtual class in AMS. ![]() Var t = ASTGUI.cliCommand(‘module reload’) ĪSTGUI.systemCmdWithOutput(“/bin/sed ‘/^$/d’ -i /etc/asterisk/nf”,function(a1)) Search for “module reload”.Then put the following lines after that like below. I found a work around for this problem by deleting the blank lines when the config file changes.įor this we have to make the below change in asteisk-gui file “/var/lib/asterisk/static-http/config/js/index.js”. of the blank lines increases with configuration changes and after certain no of changes the the asterisk pbx stop responding and it could not read the config files because of the huge file size. When I installed the Asterisk-GUI., I found that once we make any changes in configuration from i asterisk-gui, it will insert many blank lines along with the changes in the configuration files(/etc/asterisk/*.conf). This is a list of the known SIP status codesĦ05 – Not Acceptable Posted in Technical Tagged Asterisk, Asterisk Dubai UAE, Freepbx VoIP UAE, IP Phones Dubai, SIP, Technical, VOIP, VoIP Dubai Delete Blank lines from Asterisk Config files ![]() , then it will show the SIP profile “Registered”. Public username: your need to put “sip: ” it will pre append automatically) Use this to select your access point SSID which your are connected(Your router’s Wireless connection’s name). settings -> SIP settingsįrom the options select ” New sip profile” On your phone and goto Menu -> Settings->connectivity->Admin. I could successfully configure actionvoip with E6. Here I brifely describes how to configure nokia E6 with a voip provider.
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