SIP Phone can't call Analog Phone

Asterisk Asterisk Support
1

Good morning,
I have one problem I have two analog phones connected to a 2 cisco router FXS ports, we also have 4 cisco sccp phones and one asterisk sip server with two sip phone attach to the Cisco switches.
I have the sip phone working with the cisco phones and one analog phone, I can call between all of them except one analog phone, this analog phone can call the sip with no problem, but when I go to call from the SIP phone to that one analog phone I get call failure 488.

First analog phone router address is: 172.16.1.1
Second analog phone router address is 172.16.4.1 (This is the one I can’t call)

My SIP.conf is:

[1101]
type=friend ; Friends place calls and receive calls
host=dynamic ; This peer register with us
context=phones ; Context for incoming calls from this user
defaultip=172.16.1.1

[4404]
type=friend ; Friends place calls and receive calls
host=dynamic ; This peer register with us
context=phones ; Context for incoming calls from this user
defaultip=172.16.4.1

extensions.conf:

exten => 1101,1,Dial(SIP/${EXTEN}@172.16.1.1,60,t)
same => n,hangup()

exten => 4404,1,Dial(SIP/${EXTEN}@172.16.4.1,60,t)
same => n,hangup()

Thank you

2

The argument to dial application is probably the problem.
Remove the ‘@’ sign and the IP address.

3

If this does not help, please enable debug & verbose printing and monitor the message at the Asterisk console.

4

The @ and IP address aren’t necessarily wrong, but cause Asterisk to use the default settings in sip.conf, when most people would want to use a specific entry.

5

GM and thank you for replaying back.
I took the @ symbol out and it doesnt work.
I set the debug on and verbose and when I call 1101 or 4404 I compare both print outs and they look the same with the exception that the ACK from cisco is not accepting the call, I call my cisco admin to see if she could help me in this matter.
Any other help is appreciated.
Thank you again.