Digium Asterisk 2.4 setup guide - Powered By Kayako eSupport

09 Sep 2010

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	Digium Asterisk 2.4 setup guide

Solution
===== the following setup guide was contributed by one of SIPME customer =====

I used the following configuration to interface Digium Asterisk 2.4 to sipme. This howto assumes you have an operating and functional asterisk 2.4 and asterisk_gui which are both obtained from Digium. This Howto also requires you to have g729 codec or another SIPME compatible codec installed.

Create a new service provider.

1. Click on 'Service Providers'
2. Click 'Add Service Provider'
3. At the new service provider window, select 'Custom Voip'. This will allow you to enter the custom voip parameters for sipme
4. In the comment field place a comment. Something like SIPME will do
5. Select SIP as the protocol
6. In the host field enter proxy.sipme.com.au
7. Enter the username and password provided by sipme.
8. Deselect the Register check box. I found the register feature to not work, it needs to be manually done.
9. Click save.
10. To configure the CODECS select edit from the 'Options' for the service provider you just created. Add g729a as the preferred codec. and save the changes
11. Select 'Advanced'
12. In the fromdomain enter proxy.sipme.com.au
13. In the fromuser field enter YOUR SIPME_USER_NAME

Modify the sip.conf file

1.. edit the sip.conf and add the following register string in the general context. I have included a copy of my sip.conf file below. Replace xxx and SIPME_USER_NAME:PASSWORD with your details provided by SIPME. Save the file.

register=SIPME_USER_NAME:YOUR_PASSWORD@proxy.sipme.com.au/SIPME_USER_NAME

2. If your asterisk server is behind a firewall then you will need to configure your externip and localnet paramteres.

sip.conf

[general]

context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
useragent=portasip
register=SIPME_USER_NAME:YOUR_PASSWORD@proxy.sipme.com.au/SIPME_USER_NAME
externip=xxx.xxx.xxx.xxx
localnet=YOUR_LOCALNET/255.255.255.0

Edit the users.conf file to enable NAT and a few other parameters to make it work. When you create a service provider it creates them using trunk_ as the prefix. If you have multiple installed then locate your SIPME context.

1.. Edit the users.conf and set/modify the following parameters

type=peer
nat=yes
canreinvite=no
qualify=yes
useragent=portasip

users.conf

[trunk_1]
allow=g729,gsm
context=DID_trunk_1
dialformat=${EXTEN:1}
hassip=yes
port=5060
registeriax=no
secret=esb905
trunkname=Custom - SIPME
trunkstyle=customvoip
type=peer
username=1777xxxxxx
host=proxy.sipme.com.au
fromdomain=proxy.sipme.com.au
fromuser=1777xxxxxx
nat=yes
canreinvite=no
qualify=yes
useragent=portasip
canreinvite=no
canredirect=no
dtmfmode=rfc2833

I then created a Calling rule to forward all SIP and PSTN calls to SIPME. The extracxt of my extensions.conf file is below. This output was automatically generated by the Asterisk GUI.

extensions.conf

[numberplan-custom-1]
plancomment=DialPlan1
include=default
include=parkedcalls
exten=_0X!,1,Macro(trunkdial,${trunk_1}/${EXTEN})
comment=_0X!,1,sipme-inout,standard
exten=_1777XXXXXX,1,Macro(trunkdial,${trunk_1}/${EXTEN:0})
comment=_1777XXXXXX,1,sipme - free,standard

Article Details

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Created On:

18 Mar 2008 10:48 AM