Hi, this is the second time that i post this, may be a wasnt clear the
first time.
Im having problems with an incoming peer after i upgraded asterisk from
1.0 to 1.2.4, in 1.0 i used to configure the incoming peers like this:
register => @prepago-in
[prepago-in]
type=friend
host=192.168.10.102 ; this is the cisco's ip
context = from-external
dtmfmode=rfc2833
insecure=very ; required for incoming FWD calls
in cisco as5400 the dial-peer is configured like this:
dial-peer voice 2662 voip
tone ringback alert-no-PI
description OUTPUT_TO_ASTERISK
translation-profile outgoing remove_#
destination-pattern 22662[0,1,8]T
voice-class codec 5
session protocol sipv2
session target ipv4:192.168.10.103 <--- this is the asterisk's ip
dtmf-relay rtp-nte
Using asterisk v1.0 i can receive calls perfectly, after i upgraded to
asterisk v1.2.4 i receive the following error
Feb 28 16:49:34 WARNING[11142]: chan_sip.c:3207 sip_register: Format for
registration is user[:secret[:authuser]]@host[:port][/contact] at line 154
Of course, i cant receive incoming call anymore, reading the error i
undestand that im missing the username in the register => line in
sip.conf , as you can see, there is not username parameter in the
cisco's dial-peer configuration.
Is the username a required parameter in 1.2.4, if so, why did you do
this change?
any help help will be greatly appreciated
thanks
----
Miguel
