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kamailio: how to get a value from SIP INVITE Header -4.15
Kamailio 5.4 Send CDR data to an API endpoint 0.00
how to make kamailio serial forking? 0.00
How to setup italian voice language in freeswitch? 0.00
Getting information from asterisk (freepbx) on call end , and passi... -0.38
Freeswitch users directory using a database +3.85
Freeswitch wrong caller id number after bridge 0.00
Asterisk AGI script falls when caller hangup 0.00
What is the differences between to install docker-compose by binary... 0.00
How do I connect to postgres docker image from my ruby docker appli... 0.00
Run container in host, but with dedicated IP 0.00
pip install not working inside docker container -0.10
Freeswitch support for generating an SRTP offer from an incoming 3P... 0.00
Can't attach 2 networks to container in docker-compose +4.14
Hangup notification sound 0.00
Asterisk - Distribute incoming calls to 2 destinations (load-balanc... 0.00
Does a VoIP APNS Push Payload Contain the SIP Invite? 0.00
Can I hide the clients number from users when using asterisk? -0.20
Spoofing phone calls to test call center queuing logic 0.00
Kamailio as load balancer for multiple asterisk servers 0.00
Route Call in Asterisk Server -4.20
How can I change the channel_variable like destination_number at ru... 0.00
FreeSWITCH: Unable to connect from browser(WebRTC) behind enterpris... -0.04
How to enable http server on asterisk for an ARI application +3.79
Correct way to revert back to originating server? 0.00
Parallel outgoing calls: Freeswitch 0.00
get status of SIP invite of Freeswitch 0.00
Asterisk autodial some external numbers when joining conference 0.00
is it possible to put a SIP proxy server between clients and VOIP p... 0.00
Is there EC2 Elastic IP that is public facing? without NAT? 0.00
What is the difference between the normal INVITE and the INVITE on... 0.00
Setup Voip Call from SIP Account in php -4.12
How to get the phone number(callerID) in asterisk 0.00
Either the outbound or inbound call only work in asterisk setup, no... 0.00
send a specific DTMF phone as soon as hangup the call on asterisk 0.00
What are the deployment differences between Hosted vs Internal PBX? -0.26
Transfer call to custom extension to pause recording and dial exter... 0.00
Freeswitch's `mod_dingaling` Google Voice authentication failure 0.00
SIP Customer Devices under NAT -0.07
forward REGISTER message with kamailio +3.77
How can i read DTMF using asterisk 0.00
Knowing the type of VoIP application carried by H.323 0.00
Asterisk* Originating a playback php script , but getting fwrite()... 0.00
Asterisk and Sipp UAS 0.00
Re-INVITE during call results with 481 Call Does Not Exist +1.91
How to run agi-script from java 0.00
Play say function playback file with G729 Format 0.00
Forking/Multiregistration of same endpoint 0.00
Testing Asterisk SIP and DAHDI local calls 0.00
Asterisk ringgroup get call after registration 0.00