asterisk users - Sep 2006 - using SIP to connect remote other VoIP server

How to using SIP to connect remote other VoIP server? is it only running one
line voice if I registered a one SIP account?
   anybody can give me some sample configuration files? thanks a lot!
 
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Could you be more specific?  Do you want to set up linking between two 
asterisk servers?  Is this to a service provider?  

A single SIP registration and peer entry will handle multiple channels, and 
can also handle different numbers at the destinations.  Try to get away from 
thinking of things in terms of "lines"  PRI and VoIP use channels and
routing
instead.  A SIP registration and peer statement is used to tell a servers 
where to find each other.  You could have multiple calls going to different 
extensions using only one entry.  It's all about how you set up your
routing.

What is it that you want to do?

-Tim

On September 6, 2006 21:27, tengulre wrote:
>     How to using SIP to connect remote other VoIP server? is it only
> running one line voice if I registered a one SIP account? anybody can give
> me some sample configuration files? thanks a lot!
-- 
Tim St. Pierre

IP telephony specialist
sip://5101@communicatefreely.net
Toronto: 647 722 6930
Toll-Free 1 888 488 6940
tim@communicatefreely.net
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All,
   Could any one help me in configuring the feature codes for Call
forward feature in asterisk..?

How  to configure the feature code *XX for activation /deactivation of
call forward for SIP users ?  

Would appreciate , if somebody can help me more in detail .



Thanks & Regards,
-Sathish

Hi,
   
  This is a sample file I am currently using on my server.
  My server has a public IP address and an internal IP address (duan NIC).
  It runs Fedora Core 3 running iptables firewall already configured with ports 
  4569, 5060, 10000-20000 open (udp and tcp)
   
  [general]
context=default
allowguest=no
realm=your.hostname.ext
bindaddr=0.0.0.0
bindport=5060
externip=your.server.ip.address
srvlookup=no
maxexpirey=3600
disallow=all
allow=ulaw
allow=ilbc
allow=gsm
musicclass=default
language=es
rtptimeout=120
rtpholdtimeout=300
useragent=asterisk
localnet=10.10.10.0/255.255.255.0
rtcachefriends=no
qualify=yes
   
  [311]
type=friend
regexten=311
username=311
secret=311
callerid="User on extension 311" <311>
host=dynamic
nat=yes
canreinvite=no
   
  [312] 
type=friend
regexten=312
username=312
secret=312
callerid="User on extension 312" <312>
host=dynamic
nat=yes
canreinvite=no
  

tengulre <tengulre@megamail.com.cn> wrote:
              How to using SIP to connect remote other VoIP server? is it only
running one line voice if I registered a one SIP account?
     anybody can give me some sample configuration files? thanks a lot!
   
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This is what I do:

[cf]
exten =>
_*72XXXXXXX,1,DBput(CF/${CALLERIDNUM}=${CALLERIDNUM:-10:3}${EXTEN:3})
exten => _*72XXXXXXX,2,Answer
exten => _*72XXXXXXX,3,Playback(call-fwd-unconditional)
exten => _*72XXXXXXX,4,Playback(is-set-to)
exten => _*72XXXXXXX,5,SayDigits(${EXTEN:3})
exten => _*72XXXXXXX,6,Playback(vm-goodbye)
exten => _*72XXXXXXX,7,hangup
exten => _*72XXXXXXXXX.,1,DBput(CF/${CALLERIDNUM}=${EXTEN:3})
exten => _*72XXXXXXXXX.,2,Answer
exten => _*72XXXXXXXXX.,3,Playback(call-fwd-unconditional)
exten => _*72XXXXXXXXX.,4,Playback(is-set-to)
exten => _*72XXXXXXXXX.,5,SayDigits(${EXTEN:3})
exten => _*72XXXXXXXXX.,6,Playback(vm-goodbye)
exten => _*72XXXXXXXXX.,7,hangup
exten => *73,1,DBdel(CF/${CALLERIDNUM})
exten => *73,2,Answer
exten => *73,3,Playback(call-fwd-cancelled)
exten => *73,4,wait(.5)
exten => *73,5,playback(vm-goodbye)
exten => *73,6,hangup

*72+ number will activate call forwarding... *73 will deactivate call
forwarding.
You then just add a DBget in your inbound dialplan to see if CF key exists
in the database.



On 9/7/06, A C Sathish-a22713 <sathishac@motorola.com>
wrote:
>
> All,
>   Could any one help me in configuring the feature codes for Call
> forward feature in asterisk..?
>
> How  to configure the feature code *XX for activation /deactivation of
> call forward for SIP users ?
>
> Would appreciate , if somebody can help me more in detail .
>
>
>
> Thanks & Regards,
> -Sathish
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
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