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  • Aykut, Wed 01 of Aug, 2007 [07:53 UTC]: Hi all, does anybody know about Thomson ST2030 SIP phone. I have upgraded it to latest version (1.56) but "Hold" and "Conf" features are not working after the upgrade ?? Do you know any solution or do you have Ver. 1.52 ?? Where can I find it?
  • Edward J Brown, Tue 31 of Jul, 2007 [23:33 UTC]: Has anybody experienced Choppy voice quality when using a Linksys SPA942 in an Asterisk Conference bridge? It works fine with my polycom and Cisco, but sucks with my Linksys.
  • www.astawerks.com, Fri 27 of Jul, 2007 [18:00 UTC]: does anyone use asterisk on top of clark connect? does it work good?
  • simon, Fri 27 of Jul, 2007 [14:16 UTC]: Hi All, Has anyone here managed to get the Cisco79x1 to successfully fail over to the backup proxy. I have 2 asterisk servers , handsets all register and function, except that backup proxy function doesn't work. Any working example would be very apprecia
  • Matthew Richmond, Thu 26 of Jul, 2007 [03:40 UTC]: using the page() application to page across our building...often the meetme conferences don't disconnect after the caller hangs up. Anyone else having this problem. (using Polycom phones)
  • Matthew Richmond, Wed 25 of Jul, 2007 [02:58 UTC]: thanks Nicholas Blasgen! I haven't worked with AGI before, but there's always a first! Thanks again!
  • Nicholas Blasgen, Tue 24 of Jul, 2007 [19:18 UTC]: Matthew Richmond, AGI will handle all that for you.
  • sam, Mon 23 of Jul, 2007 [16:39 UTC]: need help - certain voicemail extension will stop working and recording voicemail on asterisk - anyone know why and how to fix it? Thanks
  • john haji, Mon 23 of Jul, 2007 [14:55 UTC]: free calls to pakistan
  • bong, Sat 21 of Jul, 2007 [19:09 UTC]: hi good day to all can anyone help me how to configured the nortel sip to the signaling server and how to activate in mobile w/ sip compatible without mcs
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P-Asserted-Identity and Remote-Party-ID header

P-Asserted-Identity and Remote Party ID SIP header

Some wholesale providers are now requiring that packets include P-Asserted-Identity or Remote Party ID in the header of SIP packets.

I spent some time messing with this since there is little to no documentation on such things regarding Asterisk or Trixbox. Since I had such a hard time of it, I thought I would write a really short howto so that maybe others will have it bit easier.

Remote Party ID:

This was the easier of the two. In sip.conf you must set the two variables

sendrpid = yes
trustrpid = no



P-Asserted-Identity:

This was more difficult to figure out and had far less documentation. To add headers we use the SipAddHeader() function in the dial plan, extensions.conf, or for trixbox, extensions_custom.conf.

in our trixbox extensions_custom.conf we have set up the following... assuming you dial from the [from-internal] context

include extensions_trixbox.conf
include extensions_hud.conf

[from-internal-custom]
exten => _1NXXNXXXXXX,1,SipAddHeader(P-Asserted-Identity: <sip:${CALLERIDNUM}>)
exten => _NXXNXXXXXX,1,SipAddHeader(P-Asserted-Identity: <sip:${CALLERIDNUM}>)

include => from-internal-trixbox


You may substitute whatever you want in side the < > a provided ID number or whatever... we just set callerid for demonstration purposes.

Obviously you can also change, or add a dial pattern, or specific extension in place of the _NXX for example

exten => 2000,1,SipAddHeader(P-Asserted-Identity: <sip:${CALLERIDNUM}>)

would only add headers when extension 2000 dialed.

Testing:
To test that your headers are being added, start the asterisk console and make a call that would match the extension pattern. You should see some thing like the following on your cli

-- Executing SIPAddHeader("SIP/2001-0857e7b8", "P-Asserted-Identity: <sip:2001>") in new stack

To further test, you can run tshark (the new name for ethereals command line packet capture tethereal) on your asterisk server when you make the call and capture sip packets to a log file

  1. tshark port 5060 -w sip.cap

After you place the call hit ctrl+c to close tshark then open up sip.cap and look for the appropriate header entry in the packet.

Remote-Party-ID: "eric" <sip:2001@64.34.93.136>;privacy=off;screen=no


AND


P-Asserted-Identity: <sip:2001>


Thats it you have both P-asserted-id and Remote-party-id in your sip packet headers!
Created by Eric Tamme, Last modification by the_duke on Fri 29 of Jun, 2007 [15:46 UTC]

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