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Fri 16 of May, 2008 [17:13 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.
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Siproxd

Siproxd



From the website:

Siprox is an proxy/masquerading daemon for the SIP protocol.
It handles registrations of SIP clients on a private IP network
and performs rewriting of the SIP message bodies to make SIP
connections possible via an masquerading firewall.
It allows SIP clients (like kphone, linphone) to work behind
an IP masquerading firewall or router.

General Overview:
  • SIP (RFC3261) Proxy for SIP based softphones hidden behind a masquerading firewall
  • works with "dial-up" conenctions (dynamic IP addresses)
  • Multiple local users/hosts can be masqueraded simultaneously
  • Access control (IP based) for incoming traffic
  • Proxy Authentication for registration of local clients (User Agents) with individual passwords for each user
  • May be used as pure Outbound proxy (registration of local UAs to a 3rd party registrar)
  • Fli4l OPT_SIP (still experimental) available, check FLI4L OPT Paket
  • supports Linux and FreeBSD (other BSD derivates not yet tested)
  • Full duplex RTP data stream proxy for *incoming* and *outgoing* audio data - no firewall masquerading entries needed
  • Port range to be used for RTP traffic is configurable (-> easy to set up apropriate firewall rules for RTP traffic)
  • RTP proxy can handle multiple RTP streams (eg. audio + video) within a single SIP session.
  • Supports running in a chroot jail and changing user-ID after startup
  • All configuration done via one simple ascii configuration file
  • Logging to syslog in daemon mode
  • RPM package
  • The host part of UA registration entries can be masqueraded (mask_host, masked_host config items). Some Siemens SIP phones seem to need this 'feature'.

Written in C.

Created by jht2, Last modification by posde on Tue 23 of Mar, 2004 [10:25 UTC]

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