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Sat 17 of May, 2008 [04:00 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: 0.38s
  • Memory usage: 2.18MB
  • Database queries: 33
  • GZIP: Disabled
  • Server load: 0.43

SIP Trunking

From the SIP RFC 4904:


A Session Initiation Protocol (SIP) to PSTN gateway may have trunks that are connected to different carriers. It is entirely reasonable for a SIP proxy to choose — based on factors not enumerated in this document — which carrier a call is sent to when it proxies a session setup request to the gateway. Since multiple carriers can transport a call to a particular phone number, the phone number itself is not sufficient to identify the carrier at the gateway. An additional piece of information in the form of a trunk group can be used to further pare down the choices at the gateway. As used in this document, trunks are necessarily tied to gateways, and a proxy that uses trunk groups during routing of the request to a particular gateway knows and controls which gateway the call will be routed to, and knows what trunking resources are present on that gateway.


In an architecture where calls can be terminated on multiple gateways it is wise to consider routing the call to a destination based on some significant criteria such as cost, quality or proximity. Where a proxy has the ability to evaluate a call based on one or more of these criteria, as well as knowledge of the TDM trunk resources available, the proxy can "tag" the call using the tgrp and trunk-context values in the SIP Contact field of the INVITE. It is important to note that the tgrp and trunk-context values can only be used with a TEL URI, not with a SIP URI.

To be added...

  • Least Cost Routing (LCR)
  • PSTN-to-PSTN calls traversing SIP networks

See Also:

SIP | RFC4904
Created by Samuel Rausch, Last modification by Samuel Rausch on Tue 29 of Apr, 2008 [17:49 UTC]

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