I use a simple scheme:SIP video phone A (h264/Asterisk 1.8.11)SIP video phone B (h264/Asterisk 11.7.0)When calls from A to B and vice versa drop on pickup.On B side:[Oct 24 16:33:49] DEBUG[C-00000012] res_rtp_asterisk.c: Setting the marker ..
I am having an issue that prevents WebSockets over SSL/TLS (or any kind of encrypted HTTP traffic to Asterisk) from working after an openssl library update.My setup is CentOS 6 x86_64, and initially, with openssl[-devel]-1.0.0-20.el6_2.5.x86_64 . W..
HiIm trying to get the rebuilt parking functionality to work in Asterisk 12.0.0.In Asterisk 11.6.0 I managed to get a call to get parked by adding a dynamic feature in features.conf for the DMTF sequence *# which called a macro in extensions.conf, wh..
HiI am experiencing Asterisk Crash. Log got stopped when asterisk crashed. Please help me to identify the reason and fix this issue.Asterisk: 1.8.20I am using AMI and fastAGI to control the call. Some part of dial plan is also defined in extensions.co..
all,I have console debugging enabled in logger.conf:console => notice,warning,error,debugThen a issue de command:core set debug 100 manager.cTo see only debugging messages from AMI.But It shows nothing!!!And then if I do:core set debug 1Then I can ..
everybody,I am trying to find an intermittent SIP error with one provider and thought the best first step would be to have sip set DEBUG on for some days and check the logs.Everything gets logged nicely, but the SIP log clutters up the console qu..
Continuing with the saga of Digium vs MTNL Mumbai, looking for suggestions on handling incoming Caller-ID issues.The card manages to grab a couple of (random) digits of the incoming CID, but theyre more or less useless.Is there any way to fix this?Aster..
Asterisk 1.8.13 on Debian Testing (Wheezy), MTNL Mumbai. Digium, Inc. Wildcard AEX410 4-port analog card (PCI-Express)When Asterisk executes HangUp() on an incoming call, the line remains connected for the caller.Zone = in, opermode = INDIA.Line ..
Im trying to set up a way that our users can send an XMPP message to Asterisk (unsolicited) to request information, such as voicemail status or the like.No matter what I set for the dialplan, Im only seeing Asterisk execute the s,1 priority in the cont..
I am encountering problem making concurrent calls using A sangoma card, It seems that the 2nd call get a congested or buzy,I connect via sip–>asterisk–>dahdi attached is the PRI debug ..