I wasn’t correcting you or even addressing you in the post.
This thread is getting out of hand and the OP is getting confused.
My comment was to simplify things. Registration has nothing to do with authenticating an inbound SIP session.
Asterisk is going to try to match the inbound call against all peers, it’s that simple.
As far as editing posts. Talk to the chief admin.
Only other thing I can suggest is change type=peer to type=friend with the username in the trunk name box as already suggested.
I seem to remember that I read somewhere that the SIP invites from Callcentric come from different IP addresses than the address to which you are registered which might explain why the peer cannot match. With type=friend you are set as both user and peer so it should be possible for the match to occur on the username.
I have always found that trunk setup can be more of an art than a science as every Sip provider seems to do things slightly differently which means sometimes there is quite a bit of trial and error involved.
That most likely means your trunk is not set up properly
I gave that a try. No luck. Didn’t change the behavior at all.
Based on all the responses above, Callcentric support and several other places on the Internet, it seems that accepting anonymous SIP calls is necessary when using Callcentric.
I was intrigued by this problem and so did some research by signing up for a Callcentric free trunk and DID. Using Wireshark I discovered that the sip invites for inbound calls were coming from a range of IP addresses from 204.11.192.160 to 204.11.192.171. Those were the ones I saw but there may be more. The trunk registration from callcentric.com was on IP 204.11.192.161.
This explained why the peers would not match and anonymous sip had to be turned on.
I created a SIP trunk as follows:
[Callcentric161]
type=peer
secret=password
insecure=port,invite
host=204.11.192.161
fromuser=1777XXXXXXX
fromdomain=callcentric.com
dtmf=rfc2833
disallow=all
defaultuser=1777XXXXXXX
context=from-trunk
allow=ulaw
Register String: 1777XXXXXXX:[email protected]/1777XXXXXXX
and then created trunks for the other IP addresses without register strings like this:
[callcentric160]
type=peer
secret=password
insecure=port,invite
host=204.11.192.160
fromuser=1777XXXXXXX
fromdomain=callcentric.com
dtmf=rfc2833
disallow=all
defaultuser=1777XXXXXXX
context=from-trunk
allow=ulaw
and so on for the other IP’s.
Doing this gave me peers that match all of the possible IP addresses that callcentric appear to be sending so I could switch off allow anonymous sip calls.
I do not understand why a user could not be matched using type=friend as I did this some years ago on an old Trixbox (1.4 I think), it seems to me that the ability to do this might have been removed for more modern versions of asterisk. I was using 1.11 for this investigation. I don’t know if anyone can throw some light on this.
It is ungainly but it works and makes your system much more secure. Does anyone have any ideas for an easier way?
type=friend, insecure=port,invite and the from-domain should be enough to allow any calls from that domain. You still need to have non- AAA parameters such as context, DTMF, CODEC etc.
Scott that’s what I expected but it dose not happen when I make type=friend and the trunk name the username. It seems to me the match should occur on the username in that situation.
One other thing if you have more than one DID on the trunk or would prefer the DID number to be the called number make context=from-pstn-toheader instead of from-trunk, then DID’s will appear in the format 1NXXNXXXXXX rather than the username.
Was a good solution found to make a CallCentric trunk work without Allow Anonymous Inbound SIP Calls set to YES?
Thanks
not that I’m aware of
CallCentric has been very helpfull with this issue, I’m not sure if this will work however, anybody want to comment on it before I try it? I’m afraid I’m going to break it, I’m just a noob!
Here is what they gave me to try:
If you would like to keep the Allow Anonymous Inbound SIP Calls setting disabled on your end, you will need to additionally add the following within your sip_custom_post.conf file:
[callcentric1](callcentric);
host=alpha1.callcentric.com
[callcentric2](callcentric);
host=alpha2.callcentric.com
[callcentric3](callcentric);
host=alpha3.callcentric.com
[callcentric4](callcentric);
host=alpha4.callcentric.com
[callcentric5](callcentric);
host=alpha5.callcentric.com
[callcentric6](callcentric);
host=alpha6.callcentric.com
[callcentric7](callcentric);
host=alpha7.callcentric.com
[callcentric8](callcentric);
host=alpha8.callcentric.com
[callcentric9](callcentric);
host=alpha9.callcentric.com
[callcentric10](callcentric);
host=alpha10.callcentric.com
[callcentric10](callcentric);
host=alpha11.callcentric.com
[callcentric11](callcentric);
host=alpha11.callcentric.com
[callcentric12](callcentric);
host=alpha12.callcentric.com
[callcentric13](callcentric);
host=alpha13.callcentric.com
[callcentric14](callcentric);
host=alpha14.callcentric.com
[callcentric15](callcentric);
host=alpha15.callcentric.com
[callcentric16](callcentric);
host=alpha16.callcentric.com
[callcentric17](callcentric);
host=alpha17.callcentric.com
[callcentric18](callcentric);
host=alpha18.callcentric.com
[callcentric19](callcentric);
host=alpha19.callcentric.com
[callcentric20](callcentric);
host=alpha20.callcentric.com
If you have added the lines above into your sip_custom_post.conf file, you will need to restart the entire Asterisk process in order for the changes to take into effect.
If you have any more questions, feel free to ask us.
The more I think about it I wonder if under the settings/asterisk sip settings page in FreeBPX at the bottom you have a spot to add Other SIP Settings, I bet a person could add host=alpha7.callcentric.com, host=alpha8.callcentric.com, etc. I may give that a try first.
In the end I ended up added the above to the sip_custom_post.conf file, all seems to be working now. This seems to be the best option that I can come with.