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Sat 17 of May, 2008 [18:28 UTC]

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  • Juan Ortega, Thu 15 of May, 2008 [10:33 UTC]: Hi everybody, I'm Juan, an ITCom student, and I need to know what basic elements I need to create a VoIP network. Can anybody helpme, please?,Thank you very much
  • gineta, Wed 14 of May, 2008 [03:58 UTC]: any here not fine the configuration of firewall juniper -screem for VOIP asterisk????
  • Anoop Prabhakaran, Tue 13 of May, 2008 [12:16 UTC]: I am developing Asterisk IVR, Whenever i make a internation call to the IVR system, the DTMF is not getting detected properly, this happens only for the first time, second call onwards system works fine. why this is happening
  • joe, Mon 12 of May, 2008 [04:27 UTC]: Is there an opensource browser based softphone, or a system like Busta where everything is not manages through their website?
  • Nick Barnes, Fri 09 of May, 2008 [11:36 UTC]: Christopher - yesterday I tried an Asterisk install on a CentOS 5.1 box with stock GUI and it all worked fine. Sorry I can't help.
  • aero, Fri 09 of May, 2008 [08:20 UTC]: can someone help me out on this, i tried to play some sound files on my asterisk box and this is the error message i got. WARNING[4429]: format_wav.c:169 check_header: Unexpected freqency 22050 May 8 11:17:39 WARNING[4433]: codec_gsm.c:194 gsmtolin_fra
  • Christopher Faust, Thu 08 of May, 2008 [14:15 UTC]: I beleive that I may have to change something in the xserver configuration. Please advise
  • Christopher Faust, Thu 08 of May, 2008 [14:14 UTC]: Everything was perfect. In the bios I have increased the memory allocated Still receive input not supported on my display.
  • Christopher Faust, Thu 08 of May, 2008 [14:13 UTC]: This would not be my main box. I am doing some testing to see if I can install zaptel and asterisk 1.4 on a full centos 5.1 box with development software Its bizzare, because before I went through the asterisk and zaptel installation everything was perfe
  • Nick Barnes, Thu 08 of May, 2008 [13:44 UTC]: Christopher - I can't see any way in which an Asterisk installation would muck your GUI, but remember that it is advised not to use a GUI on an Asterisk box anyway.
Server Stats
  • Execution time: 0.36s
  • Memory usage: 2.18MB
  • Database queries: 33
  • GZIP: Disabled
  • Server load: 0.92

PJSUA

PJSUA is a command line SIP user agent (UA) written with PJSIP Open source SIP stack. While it is used mainly as the reference implementation of PJSIP, it is quite useful for testing or troubleshooting SIP installations, because it prints out all SIP messages sent and received by the application to console, so everything is visible by the user.

All SIP features supported by PJSIP should be available from PJSUA. Among other things, it supports:
  • multiple accounts, with each having different registration settings,
  • multiple SIP calls (with or without conferencing),
  • SIMPLE/presence with SUBSCRIBE/NOTIFY,
  • PUBLISH,
  • support for PIDF and X-PIDF
  • instant messaging with MESSAGE request,
  • call transfer,
  • conferencing,
  • RFC 2833,
  • adaptive jitter buffer,
  • AEC, PLC, VAD,
  • STUN,
  • ICE,
  • Speex, iLBC, GSM, G.711, L16 codecs, narrow-band, wideband/16KHz or ultra-wideband/32KHz,
  • looping the media (local or remote),
  • playing WAV files,
  • quality statistics with RTCP
  • etc.


Screenshots of pjsua:


Links


Created by Martin, Last modification by Martin on Sat 21 of Apr, 2007 [02:37 UTC]

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