Complete noob struggling with setup

Hello all
sorry to just dive right in like this but i’m haveing a difficult time with getting asterisk setup. i have exhausted myself for a week with all the info i could find around the net. i did have a recent setup of trixbox that would make outbound calls but could not recieve calls “mainly just test driving distros” and it was pieceful around here without the phone ringing every couple minutes. i have a small buisness home office with the wife and outsourced phones before but now i would like to save a few bucks and host my own with an old server i have in the rack . i have some skills with linux and my networking skills are great everything is in place just need to make some calls. i have the system installed and i dropped the sip conf i was given from my provider Viatalk in the sip.conf it shows in asterisk log that it does connect with my provider but when i call from my cell i get a log entry (Failed to authenticate device "xxxxx xxxxx ";tag=as3a16bfba) so it obviously see’s the call but doesnt like it. Now as for making calls from simple x-lite4 setup as my extension i get weird tone after dialing and the call fails. i will worry about that in a minute but as for now i need to except some calls i have attached my sip.conf incase it might help and thanks to everyone for the help.


register => my number

authuser=my number
fromuser=my number
nat=no ; switch to yes if behind nat (try to avoid it if at all possible)

nat=yes ; allows you to use a softphone/adapter behind nat
secret=password ; this can be anything you want

Start with a CLI trace at at least verbosity 3. As you have record history set, the output from that, as well.

ok i did what you asked “i think” when i call the line from my cell output is as follows

== Using SIP RTP CoS mark 5
[Feb 28 21:56:01] WARNING[2982]: chan_sip.c:12738 check_auth: usernamemismatch, have , digest has
[Feb 28 21:56:01] NOTICE[2982]: chan_sip.c:20082 handle_request_invite: Failed to authenticate device "my name " sip:mycellphone#@;tag=as 6847a7d4

and when i try to make a call from my extension 6000 that i setup on sip phone i get the following

== Using SIP RTP CoS mark 5
– Executing [7034703185@DLPN_DialPlan1:1] Macro(“SIP/6000-00000004”, “trunkdial-failover-0.3,SIP/viatalk/,viatalk,”) in new stack
– Executing [s@macro-trunkdial-failover-0.3:1] GotoIf(“SIP/6000-00000004”, “0?1-fmsetcid,1”) in new stack
– Executing [s@macro-trunkdial-failover-0.3:2] GotoIf(“SIP/6000-00000004”, “0?1-setgbobname,1”) in new stack
– Executing [s@macro-trunkdial-failover-0.3:3] Set(“SIP/6000-00000004”, “CALLERID(num)=”) in new stack
– Executing [s@macro-trunkdial-failover-0.3:4] GotoIf(“SIP/6000-00000004”, “0?1-dial,1”) in new stack
– Executing [s@macro-trunkdial-failover-0.3:5] Set(“SIP/6000-00000004”, “CALLERID(all)=”) in new stack
– Executing [s@macro-trunkdial-failover-0.3:6] Goto(“SIP/6000-00000004”, “1-dial,1”) in new stack
– Goto (macro-trunkdial-failover-0.3,1-dial,1)
– Executing [1-dial@macro-trunkdial-failover-0.3:1] Dial(“SIP/6000-00000004”, “SIP/viatalk/”) in new stack
== Using SIP RTP CoS mark 5
– Called viatalk/
– SIP/viatalk-00000005 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
– Executing [1-dial@macro-trunkdial-failover-0.3:2] GotoIf(“SIP/6000-00000004”, “0 > 0 ?1-CONGESTION,1:1-out,1”) in new stack
– Goto (macro-trunkdial-failover-0.3,1-out,1)
– Executing [1-out@macro-trunkdial-failover-0.3:1] Hangup(“SIP/6000-00000004”, “”) in new stack
== Spawn extension (macro-trunkdial-failover-0.3, 1-out, 1) exited non-zero on ‘SIP/6000-00000004’ in macro ‘trunkdial-failover-0.3’
== Spawn extension (DLPN_DialPlan1, 7034703185, 1) exited non-zero on ‘SIP/6000-00000004’

i recieved my phones today panasonic kx-tgp550t04 and i am able to get it to connect to asterisk server using extension 6000. i havent ported the buisness phone number yet so this isn’t killing me but it is nerve racking.
thanks for trying to help me out i feel like i may be in way over my head on this one.

insecure=very is no longer recognized. It has been replaced by insecure=port,invite.

However, it was very rarely an appropriate relaxation in authentication. Most times when something is needed there, insecure=invite is what is actually sufficient.

Thanks david for having a look and tring to help me out i am new to this pbx telephony stuff but some peices are starting to come together. now back to the problem I did see in the logs a issue with insecure=very several times and after setting it to “no” the logs shutup about it but i had changed it back due to the sip providers sip.conf that they recomend. all that said i just tried both “invite” and “port,invite” and the only change was the trunks phone number is in place of what used to be the number of where i was calling from ex. my cell phone i called it from.output below.

== Using SIP RTP CoS mark 5
[Feb 29 15:58:21] NOTICE[2985]: chan_sip.c:20161 handle_request_invite: Call from ‘my trunk phone #’ to extension ‘s’ rejected because extension not found in context ‘DID_viatalk’.

i currently have the insecure= set to port,invite. i also see the log about “context=default” and something about degrading it i would post it but cant seem to get that far back to it. also still getting alot of time out logs for trying to register to my provider and the reg string appears wrong though it is correct in the sip.conf but on a reboot it logs that it is now reachable and withen seconds it starts to time out eventualy maximum retries exceeded. output below.

chan_sip.c: Peer ‘viatalk’ is now Reachable. (3057ms / 3600ms

chan_sip.c: – Registration for ‘my phone’ timed out, trying again (Attempt #1)

now as for trying to make a outgoing call from my desk phone ext.6000 i have a dial tone but i get the buisy signal after dialing the output is the same as before. once again thanks david for dropping in on my thread here and i look forward to making this setup work eventualy.

UPDATE i have stopped trouble shooting this current issue. i setup trixbox ce again and went thru the entire setup including all extensions and in the end it all works without one issue and it was all done thru the GUI with the exception of password changes. i think the asterisk GUI has alot of bugs i had to constantly edit the config to get the correct info there the file editor was making duplicates of everything i used it for. i want to thank you david for taking the time to drop in and read my post it was nice to see someone was going to try and help.