Connect freePBX 2.10.1.2+asterisk 1.6 to cisco trank

General Help
1

I need to connect Asterisk to a server-based telephony cisco. From cisco i have this config:

https://docs.google.com/open?id=0BwMWDgrvIoiWbHB4RHlUZWxMTTA
https://docs.google.com/open?id=0BwMWDgrvIoiWYlNfbDZvQXNxd00

from freePBX i have trank

disallow=all
context=incoming
port=5080
host=172.21.100.10
type=friend
qualify=yes
allow=g711alaw
nat=no
canreinvite=yes

Cisco servers are 172.21.100.10 and 172.21.100.11

The logs do not have anything useful:

Reliably Transmitting (no NAT) to 172.21.100.10:5080:
OPTIONS sip:172.21.100.10 SIP/2.0
Via: SIP/2.0/UDP 172.18.9.9:5060;branch=z9hG4bK0368f097
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as7359af00
To: sip:172.21.100.10
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.10.1(1.8.9.2)
Date: Tue, 18 Dec 2012 15:14:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS

Men, in fact I need help.

2

Several issues.

Asterisk syntax for g711 CODEC is ‘ulaw’ not ‘g711ulaw’ I would also check the CUCM syntax.

Why the notstandard port? You have to change Asterisk bindport and that is global. You can’t change the port for one host. The port directive sets the outbound port.

Lastly, FreePBX does not have an incoming context. The most popular context’s are:

from-pstn - This context sends the call to inbound route processing

from-internal - Complete access to the inside dial plan.

For reference all SIP variables are explained in the docs:

http://svnview.digium.com/svn/asterisk/trunk/configs/sip.conf.sample