Archives : December-2014
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We are seeing something weird we havent come across before. It seems they are sending us a different IP in the SIP from URI, than the IP they are actually sending us the traffic from.Basically, the traffic is coming from 65.211.180.237 but the hea..
all,On an Elastix server with asterisk 11.13.0 I have no verbose logs despite the fact that its OK in CLI, eg verbose set to 3 in my caseLogger.conf[logfiles];; Format is filename and then levels of debugging to be included:;debug;notice;warning;error;verbos..
Is there still an LDAP driver as do not see it in the CentOS 6 re..
I have successfully configured Asterisk Real time to store my configuration in DB but now the FreePBX web interface became useless because it can not be used to manage the real time DB by adding sip users or so. I need any web interface that gives..
On Fedora 20, every time the kernel updates, /var/run/asterisk owner is set to root.root.Im running asterisk under user asterisk.Is there any way to keep /var/run/asterisk as asterisk.asterisk. Or do I find a new place to put asteris..
team, I had implementation complete customized IPPBX solution with the help on Asterisk , ARA and a2billing for billing purpose. Now only issue I come is if a customer A and B want to used similar extension rang then its only possible with adding account-c..
,I have the following situation:Local T.38 endpointASTERISKSIP provider (with T.38 support)I am trying to send a fax from my local T.38 endpoint to arbitrary external fax numbers (which I am not in control of, so I dont know if the other end suppo..
Starting with asterisk 1.8, when you dial multiple channels at once and one of them is answered, all other channels were canceled with the cause 200 -Call completed elsewhere, so modern phones dont display the call asmissed.Do you know a way to trans..
all.Ive got an odd situation with my RT asterisk server.Ive got a number of users who are reporting that their voicemail greeting isnt being played anymore.This used to work before a recent asterisk restart.The dialplan is in AGI, so it wasnt changed..
I am on asterisk 12.6.0. Previously I was using 10.0.1 and for Gtalk I was using chan_gtalk and jabber configuration. But on 12.6.0 I tried to use chan_motif, asterisk starts consuming 100%cpu. From pstack trace I got it is because of ICE and to ..