login | register
Wed 01 of Aug, 2007 [09:18 UTC]

voip-info.org

Navigation
Today's Deals
Recently Visited Pages
Search with Google
Search this site with Google. Results may not include recent changes.

Web www.voip-info.org
Last Changes
Shoutbox
  • 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
Server Stats
  • Execution time: 0.45s
  • Memory usage: 2.28MB
  • Database queries: 35
  • GZIP: Disabled
  • Server load: 3.31

Asterisk Configurations for connecting with VOIP providers

Asterisk provider specific settings

This page is intended to provide a place for people to place excerpts from configuration files for specific providers.

In General

It is considered bad style to use a dialstring with a password as in "user:pass@voipserver.com" since username and password will then show up on the console. Instead, for each provider enter a peer entry in sip.conf or iax.conf and use that peer entry's name in your dialstring as in "user@peername". Username and password pairs in dialstrings should only be used for testing!

Provider Specific Settings


Settings for softswitches


More sample scripts are to be found on the Asterisk tips and tricks page.

Created by jjhall, Last modification by pmj on Wed 09 of May, 2007 [04:42 UTC]

Comments

Comments Filter

Re: Aterisk configuration for Skypho and Voipstunt

by fivos on Wednesday 15 of February, 2006 [12:29:24 UTC]
in outgoing settings :
allow=alaw&ulaw
bindaddr=0.0.0.0
canreinvite=yes
defaultexpirey=330
disallow=all
dtmfmode=inband
fromuser=zzz
host=voip.eutelia.it
insecure=very
nat=1
port=5060
qualify=yes
realm=voip.eutelia.it
secret=yyy
srvlookup=yes
type=friend
useragent=Asterisk_Eut
username=xxx

in incoming settings :

allow=ulaw&alaw
context=from-pstn
secret=yyy
type=user

in registration

zzz:yyyg@voip.eutelia.it/zzz

Aterisk configuration for Skypho and Voipstunt

by longman on Wednesday 15 of February, 2006 [00:03:51 UTC]
Hello every body.
I have been searching for a while now and i never got my hands on Asterisk configuration for any of the VoSP skypho or VoipStunt.
If any one can provide that i'll be glad.

Voicepulse doesn't connect

by omar on Tuesday 08 of November, 2005 [16:21:27 UTC]
I can't place and receive calls from voicepulse, even if my asterisk is registered.
Here is the registration:
  • CLI> sip show registry
Host Username Refresh State
access1.voicepulse.com:5060 s00xxxxxx 105 Registered

And here is the error when I try to place calls:

  • CLI> — Executing Dial("SIP/op01-3a00", "SIP/1770xxxxxxx@voicepulse01") in new stack
   — Called 177xxxxxxx@voicepulse01
   — SIP/voicepulse01-4976 is making progress passing it to SIP/op01-3a00
Nov 8 12:19:16 WARNING21088: chan_sip.c:6890 handle_response: Forbidden - wrong password on authentication for INVITE to '"Operador 01" <sip:s00xxxxxx@access1.voicepulse.com>;tag=as737d5e9e'
   — SIP/voicepulse01-4976 is circuit-busy
 == Everyone is busy/congested at this time

Any one with the same problem?

Asterisk setttings for VoIPtalk

by bence on Wednesday 27 of April, 2005 [19:40:58 UTC]
Asterisk settings for use with VoIPtalk are available at:

http://www.voiptalk.org/products/iaxconfig.html

Please update this page with new information, just login and click on the "Edit" or "Add Comment" button above. Get a free login here: Register Thanks! - support@voip-info.org

Page Changes | Comments

Sponsored by:
Terms of Service Privacy Policy
© 2003-2007 Arte Marketing, Inc.

Powered by bitweaver