I’ve got problems making a GXW4108 gateway with asterisk. I followed all the existing tutorials but still, no success. Need your help!
Here is what I want to do:
Link my GXW4108 gateway with Asterisk, so that all my external calls (incoming and outgoing) are going through my gateway. My FXO lines being connected to it. That’s it. So for example, when I dial 9 9613 3214 on my IP phone, Asterisk takes the call and make it through my GXW4108 FXO lines! Easy?
[outbound_calls]
exten=_9NXXXXXXX,1,Dial(SIP/${EXTEN}@101) (I’ve also tried with (SIP/101/${EXTEN} as well as exten=_9NXXXXXXX,1,Dial(SIP/${EXTEN}@192.168.10.27))
GXW4108 config
Everything is default, except: -Profile 1 :
SIP server to 192.168.10.24 (Asterisk srvr address)
SIP Registration to NO -FXO lines :
Wait for dial tone to YES
Stage method to 1
On my phone
When I dial 9 9645 2145, I get:
“Call failed
Reason code: 503”
Thanks a lot for your help! My hair is turning gray…
It is better to post the output of asterisk , then we can find out why this call failed. for wrong config on asterisk or for wrong on GXW. that is the clue.
The “Prefix to Specify Port(1 stage dialing method)” was set to 99. In my dial plan I use 9 as a prefix for outbound calls, and Hong-Kong mobile phone numbers start with 9 most of the time. Thus, I was dialing numbers starting with 99 and the gateway was using this “Specify port” function instead of making a normal call.
context=outbound_calls
its your context have you created this in the extension.cnf file ??
if yes then what i can understand is that you are dialing from ‘201’ ext to out bound make sure that the Dial plan outbound_calls has been selected for user ‘201’…
Check out the peers is 301 online?? in CLI type commnad Sip Show peers.