[Asterisk-Users] problem with incoming peer (cisco as5400)

Miguel mmiranda at 123.com.sv
Thu Mar 2 09:02:27 MST 2006


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







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