[asterisk-users] Asterisk-beginner : cannot make phonecalls using Asterisk (update)
Miguel Molina
mmolina at millenium.com.co
Mon Apr 13 15:31:47 CDT 2009
jonas kellens escribió:
> Hey there again !
>
Hey, just my two cents:
> I've changed some things now :
>
> 1) IP-phones get there IP from a DHCP
>
> 2) sip-accounts simplified :
>
> /[root at asterisk asterisk]# cat sip.conf/
> /[general]/
> /context=default/
> /port=5060/
> /bindaddr=0.0.0.0/
> /srvlookup=yes/
> /disallow=all/
> /allow=ulaw/
>
> /[210]/
> /type=friend/
> /context=intern/
> /host=dynamic/
>
> /[211]/
> /type=friend/
> /context=intern/
> /host=dynamic/
>
Mi first cent: This is oversimplified. I think you need to put the
username=2XX here too. You can check the configuration with the asterisk
CLI commands "sip show users", "sip show peers", "sip show user XXX" and
"sip show peer XXX"
> 3) dial plan simplified :
>
> /[root at asterisk asterisk]# cat extensions.conf/
> /[globals]/
>
> /[default]/
> /include => intern/
>
> /[intern]/
> /exten => 210,1,Dial(SIP/210)/
> /exten => 211,1,Dial(SIP/211)/
>
> The IP-phones are set as DHCP-client...
>
> I reloaded everything on the Asterisk CLI.
>
> I put off the power of the IP-phones and then put them back on.
>
My second cent: check again if your phones are configured to register,
and recheck your network configuration. Something like 255.255.255.255
on your netmask will make the communication impossible.
> I still see no register-message on the CLI. This really should happen
> now, as they are defined host=dynamic !
>
> What can be going wrong here... Tell me, I'm not writing a wrong
> sip.conf or extensions.conf, do I ?
>
> I will now hang my portable on the switch and monitor the network with
> wireshark to see if the phones send a SIP-register to the
> Asterisk-server...
>
> In the mean time... every feedback on this is very welcome, thanks.
>
You're welcome.
> Jonas.
> ------------------------------------------------------------------------
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center
PBX: (+57 1)6500800 ext. 1201
Fax: (+57 1)6500816
Móvil: (+57)3138873587
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090413/6159be72/attachment.htm
More information about the asterisk-users
mailing list