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arheops

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Asterisk Conference issue from Cisco Call Manager +0.00
Get state of other channel 0.00
Asterisk dialplan: execute context +0.00
How to Transfer a call over H323 Trunk in Asterisk 0.00
Multiple SendDTMF extension in asterisk 0.00
Select CPU type to install right flavor of FAX for asterisk +0.28
asterisk to opensips conversion. all help appresciated 0.00
Asterisk - Codec preference 0.00
Asterisk variable maximum length 0.00
Using non-standard names for contexts +0.27
Asterisk - Detect answer via sip trunk +0.01
Zoiper add pause between dialing phone numbers? 0.00
UDP port 3526 Starquiz-port 0.00
asterisk confbridge - dynamically add users to conference -0.49
Incoming SIP calls connect but end after being answered 0.00
Mute left /right audio channels 0.00
Asterisk does not start up after Trixbox reboots +0.01
Asterisk not sending media packets using music on hold 0.00
elastix cdr stop working 0.00
bash script linux - process output of ifconfig -a output -0.41
asterisk: symbol lookup error: /usr/lib/libasteriskssl.so.1 After I... 0.00
how to create a conference server for 4 users 0.00
Detect silence while playing sound 0.00
Asterisk Call Manager/1.3 Mssing action in request 0.00
Asterisk Realtime keep updating NAT value as yes -0.01
Asterisk to Asterisk connection via sip details 0.00
Asterisk Call Manager hangup event on Python 0.00
Read dtmf using asterisk-java 0.00
FreePBX - Notify missed calls on queue 0.00
Amazon EC2 and getting a response from my IP address without using... +0.27
How to keep a call alive when a caller is on the car? 0.00
How to check Asterisk SIP registration in realtime? -0.02
Asterisk: Zombie channels when initiating call bridging through AMI 0.00
Integration of AGI and its Working in Asterisk 0.00
How to add IVR audio file in dialpans in Asterisk 0.00
Elastix Call Center Module "How to disable Agent Auto Answer?&... 0.00
How to add and build app_meetme to make Conference Calls (MeetMe) i... 0.00
No application 'MeetMe' for extension error in asterisk 0.00
Kamailio-Asterisk - route "FROMASTERISK" not found 0.00
Does the Asterisk support SIP authentication using TLS authentciati... 0.00
How to Make my asterisk server to make Outbound Calls and Recieve I... 0.00
connect to a softphone from a normal mobile phone 0.00
channel name in asterisk +0.03
continue a call from asterisk by call file 0.00
force_rtp_proxy_body: incorrect port 0 in reply from rtpproxy 0.00
Asterisk - Any way to end/cancel the queue wrap-up time? 0.00
Asterisk 11 cannot make a call after installing MySQL real-time dat... 0.00
Change time in asterisk 11 0.00
how to add one SIP Extension by command line in Asterisk +0.01
How to remove one extension by asterisk CLI +0.02