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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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Asterisk Inphonex

This is what i had to do to setup my $7.95 Miami inbound DID with inphonex

1) sign up on their site for pay as you go outbound service $9.95 + $1.00 tax
2) After signing up you will have a virtual number assigned to you by INPHONEX
3) Email sales@inphonex.com that you want to purchase an incoming DID (since you can't do it on the website directly as of yet)
4) They send you an email giving you instructions on where/how to sign up for incoming DID number
5) after choosing and purchasing a DID, log in to inphonex account on their site and click on control panel
6) now under control panel you will see everything needed to setup Asterisk
7) edit sip.conf for Asterisk with these lines


;#####sip.conf###################################
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0

insecure=very ; I had to add this or i would get chan_sip.c:9046 (handle_request_invite: Failed to authenticate user) errors on the caller iD and it would get redirected to inphonex's voicemail.

disallow=all ; Disallow all codecs

allow=ilbc ;inphonex supports only ilbc and G729 I didn't feel like purchasing a G729 license from digium so I use ilbc, sounds fine



register => virtualacctnum:pass:virtualacctnum@sip.inphonex.com:5060/ext

;use the virtual account number not your actual DID in the above register => line
;'ext' at the end of this line can be any extension you choose that will correlate with an extension in extensions.conf

[inphonex]
type=peer
username=virtualacctnumber
fromuser=virtualacctnumber
secret=pass ; password used to login their website (same as in register =>)
host=sip.inphonex.com
nat=yes ; my asterisk is behind nat
canreinvite=yes
qualify=yes
context=inbound-inphonex ; context to be used in extensions.conf for inbound calls from inphonex

8 ) edit extensions.conf
;#########extension.conf#############################

[globals]
OFFICE=SIP/x101 ; my sip phone in the office


;remember replace 'ext' with the number you used at the end of the register => line in sip.conf

[inbound-inphonex]
exten => ext,1,Wait(1) ;Wait a second, and to get the CallerID information.
exten => ext,2,Playback(support) ; play initial greeting/recording
exten => ext,3,Dial(${OFFICE},20,tr) ; dial my sip phone for 20 sec
exten => ext,4,SetCallerID(${IDCALLER}) ; im not there? set call id to pass to my cell phone
exten => ext,5,Dial(IAX2/login@voipjet/cellphone#,10,tr) ; dial my cell phone , and seamlessly fw call to it. I use voipjet
exten => ext,6,Playback(busy-on-call) ;switch to a on-call message if i didnt pick up both cell and sip phone
exten => ext,7,Voicemail(1) ;leave voice mail
exten => ext,8,Hangup() ; hangup the call

9) done




Created by joshbaptiste, Last modification by joshbaptiste on Sun 07 of Aug, 2005 [23:16 UTC]

Comments Filter

by Tara Nesbitt on Monday 11 of June, 2007 [18:42:38 UTC]
Emailing sales is no longer necessary. You can order a DID number here: http://www.inphonex.com/availability/ or thru the control panel if you are an existing customer.

My comments

by joshbaptiste on Sunday 07 of August, 2005 [23:16:31 UTC]
This is what I did to get my office up with Inphonex.com, I chose Inphonex because they are a local Miami,fl company and their servers are also local (15ms ping). I'am by no means an Asterisk expert (yet), So this may have mistakes, but I'am just sharing what I did to get it to work for me.

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