We are trying to get inbound DID's working on our SER box. Outbound is working perfectly as we are doing pre-paid.
I ran this command [root@voip1 ~]# serctl alias add 6049094251 sip:5002@216.x.xxx.xx domain unknown: use usernames with domain or set default domain in SIP_DOMAIN [root@voip1 ~]#
I get the same result if I put our domain in place of the IP address. We had someone else set this box up for us and I am having problems reaching them.
5002 is is the username of my test ATA logged into the system I can make outgoing calls to the PSTN
What am I doing wrong?? I have been reading Archives allday trying to grasp this. The DID provider is directing that DID at the IP of this SER box
Eric Haskins Lead Software Developer eric@v1010.com
The sip:5002@216.X.XXX.XX is right. What it's asking for is for you to either set the SIP_DOMAIN variable to your domain:
SIP_DOMAIN=my.domain.com export SIP_DOMAIN
OR to put it into the command as:
serctl alias add 6049094251@my.domain.com sip:5002@216.X.XXX.XX
It's odd, but either way I do it, it doesn't actually PUT my domain name into the database (leaves that field blank), but the serctl command asks for it anyway. I haven't gone back to check the shellscript in the code to find out why it doesn't put the domain in there, as I use my own set of commands for user management anyway, so it's never been an issue.
Hope that helps.
N.
On Thu, 27 Oct 2005 17:00:32 -0400, Eric Haskins wrote
We are trying to get inbound DID's working on our SER box. Outbound is working perfectly as we are doing pre-paid. I ran this command [root@voip1 ~]# serctl alias add 6049094251 sip:5002@216.x.xxx.xx domain unknown: use usernames with domain or set default domain in SIP_DOMAIN [root@voip1 ~]# I get the same result if I put our domain in place of the IP address. We had someone else set this box up for us and I am having problems reaching them. 5002 is is the username of my test ATA logged into the system I can make outgoing calls to the PSTN What am I doing wrong?? I have been reading Archives allday trying to grasp this. The DID provider is directing that DID at the IP of this SER box Eric Haskins Lead Software Developer eric@v1010.com
<SNIP> serctl alias add 6049094251@my.domain.com sip:5002@216.X.XXX.XX
It's odd, but either way I do it, it doesn't actually PUT my domain name into the database (leaves that field blank), but the serctl command asks for it anyway. I haven't gone back to check the shellscript in the code to find out why it doesn't put the domain in there, as I use my own set of commands for user management anyway, so it's never been an issue.
</SNIP>
Thx I was able to add the alias and it showed up in the DB. Using this
[root@voip1 ser]# serctl alias add 16049094251@216.x.xxx.x sip:5002@216.x.xxx.x 200 Added to table ('16049094251','sip:5002@216.x.xxx.x') to 'aliases'
When I call the DID it rings and rings then says call couldn't be completed. Here is an ethereal capture
No. Time Source Destination Protocol Info 279 20.881245 69.28.xx.xxx 216.x.xxx.x SIP/SDP Request: INVITEsip:16049094251@216.x.xxx.x:5060
349 25.887031 69.28.xx.xxx 216.x.xxx.x SIP Request:CANCELsip:16049094251@216.x.xxx.x:5060
So the call is being forwarded to us by the provider.Do I have to trust the inbound IP 69.28.xxx.xxx from the provider??? I dont understand why it isnt forwarding the call to user 5002?? I can make outgoing calls fine thru MCI our outbound carrier. Do I need to add an entry to the location DB??
Thx for all your help on this
Eric
Ok I was able to get the inbound call to ring to user 5002 I had to add Provider IP to
# authenticate calls if(!www_authorize("voip.mydomain.com","subscriber")) { # you didn't send me credentials Maybe I trust your IP? if (src_ip=~"216.8.xx.*" || src_ip=~"24.206.xxx.*" || src_ip=~"69.28.xxx.*") { log(1,"Incoming call from trusted IP"); } else { # I don't trust this IP.. ask for credentials www_challenge("voip.mydomain.com", "0"); break; } }
But once I answer 5002 it is dead air on the 5002 and the PSTN phone keeps ringing until call could not be completed.
No. Time Source Destination Protocol Info 31 2.248714 216.8.xxx.xx 65.175.xxx.xxx SIP Status: 200 OK 120 10.499698 69.28.x.xxx 216.8.xxx.xx SIP/SDP Request: INVITE sip:16049094251@216.8.xxx.xx:5060, 125 10.513795 216.8.xxx.xx 69.28.x.xxx SIP Status: 100 trying -- your call is important to us 126 10.513918 216.8.xxx.xx 65.175.xxx.xxx SIP/SDP Request: INVITE sip:5002@65.175.xxx.xxx:5060, 128 10.580401 65.175.xxx.xxx 216.8.xxx.xx SIP Status: 100 Trying 129 10.597017 65.175.xxx.xxx 216.8.xxx.xx SIP Status: 180 Ringing 130 10.597219 216.8.xxx.xx 69.28.x.xxx SIP Status: 180 Ringing 186 15.502397 69.28.x.xxx 216.8.xxx.xx SIP Request: CANCEL sip:16049094251@216.8.xxx.xx:5060 187 15.502684 216.8.xxx.xx 65.175.xxx.xxx SIP Request: CANCEL sip:5002@65.175.xxx.xxx:5060 188 15.502851 216.8.xxx.xx 69.28.x.xxx SIP Status: 200 canceling 190 15.569267 65.175.xxx.xxx 216.8.xxx.xx SIP Status: 487 Request Terminated 191 15.569386 216.8.xxx.xx 65.175.xxx.xxx SIP Request: ACK sip:5002@65.175.xxx.xxx:5060 192 15.569550 216.8.xxx.xx 69.28.x.xxx SIP Status: 487 Request Terminated 193 15.573890 65.175.xxx.xxx 216.8.xxx.xx SIP Status: 200 OK 198 15.929421 216.8.xxx.xx 69.28.x.xxx SIP Status: 487 Request Terminated 214 17.308191 65.175.xxx.xxx 216.8.xxx.xx SIP Request: NOTIFY sip:216.8.xxx.xx 215 17.308387 216.8.xxx.xx 65.175.xxx.xxx SIP Status: 200 OK 223 17.933116 216.8.xxx.xx 69.28.x.xxx SIP Status: 487 Request Terminated 397 32.367404 65.175.xxx.xxx 216.8.xxx.xx SIP Request: NOTIFY sip:216.8.xxx.xx 398 32.367590 216.8.xxx.xx 65.175.xxx.xxx SIP Status: 200 OK
65.175.* IP is my ATA Test Box user 5002 69.28.* is the DID Provider 216.8.* is the SER box
Considering this box was designed orignally for outbound only I am wondering if I have to send a response back to our DID provider to let them know it was answered?? I have searched for example configs utilizing in & outbound and have come up short
Thx again for any help
Eric
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Eric Haskins -
sip