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Asterisk Configurations for connecting with VOIP providers

Created by: jjhall,Last modification on Wed 30 of Jul, 2008 [11:16 UTC] by fmarier

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.


Comments

Comments Filter
222

333Re: Aterisk configuration for Skypho and Voipstunt

by fivos, 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
222

333Aterisk configuration for Skypho and Voipstunt

by longman, 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.
222

333Voicepulse doesn't connect

by omarn, 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?
222

333Asterisk setttings for VoIPtalk

by bence, 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