asterisk-users

2015-04-01 - 2015-05-01 (223 messages)

2015-03-01 - 2015-04-01 (370 messages)

2015-02-01 - 2015-03-01 (195 messages)

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  1. 2015-03-31  [5] [asterisk-users] Update peer IP address                                  asterisk-use Scott Griepentrog 
  2. 2015-03-31  [2] [asterisk-users] How does chan_sip match an ACK?                         asterisk-use Tony Mountifield
  3. 2015-03-31  [1] [asterisk-users] help : annoucement queue                                asterisk-use Anicet LANJANIAINA 
  4. 2015-03-31  [6] [asterisk-users] Call Quality Measuring                                  asterisk-use Patrick Beaumont 
  5. 2015-03-30  [1] [asterisk-users] WaitForSilence NEVER detects silence                    asterisk-use Mike A. Leonetti 
  6. 2015-03-30  [1] [asterisk-users] WaitForSilence NEVER detects silence,,Post              asterisk-use Mike A. Leonetti 
  7. 2015-03-29  [1] [asterisk-users] Iax2 statistics in dialplan                             asterisk-use Ethy H. Brito 
  8. 2015-03-29  [1] [asterisk-users] Mixing HASH() and LOCAL()                               asterisk-use Leandro Dardini 
  9. 2015-03-29  [1] [asterisk-users] Help! How to make Asterisk support ICE in public networ asterisk-use =?utf-8?B?5pu56LS15p6
 10. 2015-03-28  [8] [asterisk-users] Anonymous SIP calls                                     asterisk-use James Cloos 
 11. 2015-03-27  [5] [asterisk-users] call between snom 300 and aastra 6731i                  asterisk-use Gareth Blades 
 12. 2015-03-27  [1] [asterisk-users] What's the best average duration for a SIP test call?   asterisk-use Sevana Oy 
 13. 2015-03-27  [3] [asterisk-users] Gateway Eurotech                                        asterisk-use ricky gutierrez 
 14. 2015-03-27  [1] [asterisk-users] Problems playing audio file over a Page                 asterisk-use Tech Support 
 15. 2015-03-27  [2] [asterisk-users] Auto Answer                                             asterisk-use ricky gutierrez 
 16. 2015-03-26  [2] [asterisk-users] CDR dst value null after attended transfer              asterisk-use Matthew Jordan 
 17. 2015-03-26  [2] [asterisk-users] Dial to PJSIP Channel with Typo "PJSIP//" Causes Asteri asterisk-use Matthew Jordan 
 18. 2015-03-26  [4] [asterisk-users] Determining if a queue member is paused in Dialplan log asterisk-use Dale Noll 
 19. 2015-03-25  [3] [asterisk-users] Asterisk 13. Writing call quality parameters to CDR. Ho asterisk-use Ethy H. Brito 
 20. 2015-03-25  [1] [asterisk-users] PJSIP configuration for Asterisk 13.1.0/SIP trunk outbo asterisk-use Sonny Rajagopalan 
 21. 2015-03-25  [1] Re: [asterisk-users] Call Quality Measuring (Laszlo)                     asterisk-use marlon araujo 
 22. 2015-03-25  [5] [asterisk-users] TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34     asterisk-use Salaheddine Elharit 
 23. 2015-03-25  [1] Re: [asterisk-users] asterisk 11.14 - voicemail incorrect duration       asterisk-use Dominique Haeber 
 24. 2015-03-24  [4] [asterisk-users] RTP handling                                            asterisk-use Richard Mudgett 
 25. 2015-03-24 [11] [asterisk-users] Asterisk 13.1.0/PJSIP outbound calling using SIP trunk: asterisk-use Sonny Rajagopalan 
 26. 2015-03-24  [6] [asterisk-users] outbound calls                                          asterisk-use Salaheddine Elharit 
 27. 2015-03-23  [1] [asterisk-users] trying to connect to asterisk with softphone (logs, etc asterisk-use thufir 
 28. 2015-03-23  [8] Re: [asterisk-users] [OT] switches                                       asterisk-use Lukasz Sokol 
 29. 2015-03-23  [2] [asterisk-users] PJSIP - Video Support for WebRTC                        asterisk-use Matthew Jordan 
 30. 2015-03-23  [1] [asterisk-users] Question about hangup - Asterisk v11.15.0               asterisk-use Administrator TOOTAI 

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