I've done that...I think. :^)
Here's the excerpt from sip.conf:
[tycisco]
type=friend
username=cisco1
secret=***
qualify=200 ; Qualify peer is no more than 200ms away
nat=yes
;insecure=no
host=dynamic; This device registers with us
;defaultip=192.168.0.30
canreinvite=no
context=fullaccess
dtmfmode=inband
mailbox=101
disallow=all
allow=ulaw
allow=alaw
allow=g729
I still get no registration when I do a sip show peers. Am I missing
something simple?
Thanks,
Ty
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of end1r
Sent: Wednesday, April 20, 2005 8:58 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Cisco 7960 SIP registration???
Looks like you have sip.conf set up to expect registrations
for tycisco since it has a D for dynamic.
You can either set up the 7960 to register with asterisk and
use something like this in sip.conf:
[tycisco]
type=friend
username= someusername
secret= somesecret
insecure=no
mailbox=757
host=dynamic
callerid=
or just not have the 7960 register and specify its IP address
using the host= line instead.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
List Receiver
Sent: Wednesday, April 20, 2005 11:19 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Cisco 7960 SIP registration???
So, here's my quandary:
1) Asterisk running CVS HEAD as of a couple days ago
2) Cisco 7960 SIP phones in a different subnet than the
Asterisk server
3) NAT/Firewall device between 7960's and *
I can initiate a call from the 7960's just fine. They can
call out using our Broadvoice account and access any of the
vmail stuff on *.
When calling in from the outside world and dialing one of
their extensions, however, I always get a this user is on
the phone message.
The console spits out this nugget:
== CDR updated on SIP/4252780761-933d
-- Executing Macro(SIP/4252780761-933d,
stdsip|tycisco|101) in new stack
-- Executing Dial(SIP/4252780761-933d, SIP/tycisco)
in new stack Apr 20 08:14:59 NOTICE[32728]: app_dial.c:973
dial_exec_full: Unable to create channel of type 'SIP' (cause 3)
== Everyone is busy/congested at this time (1:0/1/0)
A showing of the sip peers:
sip show peers
Name/username HostDyn Nat ACL Mask
Port Status
rickcisco/cisco2 (Unspecified)D N 255.255.255.255
0UNKNOWN
tycisco/cisco1 (Unspecified)D N 255.255.255.255
0UNKNOWN
sip.broadvoice.com/425278 147.135.4.128 255.255.255.255
5060 OK (127 ms)
3 sip peers [1 online , 2 offline]
I'm sure the reason I can't call to an extension is that they
are appearing offline. How can I remedy this, however?
I'm an * newbie, so go easy on me. :^)
Thanks,
Ty Christensen
MCP, MCSP, MCSB
Master Mind Productions Inc.
www.mastermindpro.com http://www.mastermindpro.com/
(425) 378-7724
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
smime.p7s
Description: S/MIME cryptographic signature
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
