[Asterisk-Users] Incoming SIP from Cisco 7206

B. J. Bomar bbomar at fngi.net
Thu Aug 4 10:47:30 MST 2005


Here is my entry in sip.conf that works for 7200's, 3600's, and 2600's.
 
[gateway]
type=friend
host=192.168.1.61
canreinvite=yes
context=gw-inbound
qualify=no
dtmfmode=rfc2833
insecure=yes
disallow=all
allow=ulaw
allow=alaw

Hope that helps.
 
B. J.
 
 
 

  _____  

From: Scott Miller [mailto:scoscmil at imap.iu.edu] 
Sent: Wednesday, August 03, 2005 16:09
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Incoming SIP from Cisco 7206



I am running an Asterisk server through a Cisco 7206 PSTN gateway.  I am
able to make outgoing SIP calls without a problem, though incoming calls
have been somewhat of a problem.  I am not sure exactly how sip.conf should
look in such a scenario.  

 

I believe most Cisco gateways are just managed through ACL's, with no
authentication, so I think I have the outgoing "peer" statement right, but I
have no idea where to start on the incoming "user" statement.  Here's my
sip.conf (configured through AMP).

 

[gk02-inbound]     

type=user 

host=10.0.106.10

context=from-pstn

 

[gk01]

type=peer

host=10.0.50.10

 

 

When a call comes it, about every second I get this..

 

Aug  1 11:53:49 DEBUG[4076]: Stopping retransmission on
'495FF45C-1DB11DA-8E67BEF0-CE47D9F7 at 10.0.106.10' of Response 101: Found

Aug  1 11:53:49 DEBUG[4076]: Check for res for 

Aug  1 11:53:49 DEBUG[4076]:  is not a local user

Aug  1 11:53:49 DEBUG[4076]:  is not a local user

 

Any help would be appreciated..

 

Thanks,

 

 

 

Scott Allen Miller

Research Assistant

 

Telecommunications

University Information Technology Services

Indiana University

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050804/85825f05/attachment.htm


More information about the asterisk-users mailing list