First post, please be kind 
Hi guys, thanks to a of your advice, I’ve now installed Freepbx12 Distro (FreePBX-10.13.66-17-i386-Full-1477919214.iso) as I couldn’t get the latest to register my phone. The phone talks happily with the PBX but can’t make a call through the trunk. I have no idea what’s going wrong as I’ve used Orbtalks instructions perfectly ( I think)
When I call *97 I get voicemail but when I dial an outside number I get " All circuits are busy now, please try your call again later"
Any help would be gratefully received.
Which context are you connecting the phone to the server with?
Asterisk usually uses “default”, and we use “from-local” from-internal.
Entries from a call from the file “/var/log/asterisk/full” might help as well.
Lorne is right - from-internal Thanks Lorne
On the trunk?
host=sip.orbtalk.co.uk
username=USER
secret=PASSWORD
type=peer
nat=yes
insecure=very
dtmfmode=RFC2833
allow=alaw&ulaw
port=5060
fromuser=USER
fromdomain=sip.orbtalk.co.uk
context=from-trunk
This what ORbtalk had advised, is it wrong?
I could’ve been clearer I think, the phone connect quite happily to Freepbx12 as it can get VM, it just doesn’t go through the trunk at all. I can’t help but think it’s by noobish configuring guys.
I think I see it, cheers Dave (The Cisco Guru!!)
Any idea what might cause the adress not to resolve though?
18:50:06] ERROR[1778] netsock2.c: getaddrinfo(“sip.orbtalk.co.uk”, “(null)”, …): Temporary failure in name resolution
[2017-04-10 18:50:06] WARNING[1778] acl.c: Unable to lookup ‘sip.orbtalk.co.uk’
[2017-04-10 18:50:06] WARNING[1778] acl.c: Cannot connect
IN your System Admin settings, you can add a couple more DNS servers. Most people like to have 8.8.8.8 as one of the servers (Google) as it works reasonably well everywhere.
You can also add an entry in your “/etc/hosts” file for the server IP address for “sip.orbtalk.co.uk” (217.20.39.101).
System Admin was an easy find in the current version of Freepbx but I don’t see it in this version. I did add the ip (thanks Dave) to the hosts file but still the same error?
If you can’t find System Admin in your current version, you may have to man-handle the networking stuff yourself.
You can Google this stuff for more information and (as if it wasn’t confusing enough) I actually use NetBSD for everything but FreePBX, so here’s what you need to do:
-
Add the DNS servers you want to use into the list of DNS servers for your system. In my systems, that goes into /etc/resolv.conf. I don’t know where it goes exactly in Centos.
-
Add in the DNS Server stuff for a caching DNS. In the FreePBX distro, I think you get one of these for free, but I don’t know if you have one installed.
-
There should also be a file that change where you look for DNS resolution - I don’t remember the name of the file off the top of my head, but you can specify places like ‘file’ (for the hosts file), ‘yp’ (for a Yellow Pages server), and a bunch of other sources.
DNS on Centos expires quickly (like 5 seconds) so if your network isn’t quick, or your local DNS isn’t working well, you’ll run into a lot of this kind of stuff…
Well I added googles DNS in the ifcfg file for Eth0 and I updated the modules on the system and I now have System Admin showing the dns entries that i added but still no change, what am I doing wrong?
Aha! A reboot and I now get this:
app_dial.c: – Called SIP/Orbtalk/07*********0
[2017-04-10 21:20:01] WARNING[1784][C-00000000] chan_sip.c: Received response: “Forbidden” from ‘sip:1****01%20fromdomain=sip.orbtalk.co.uk%[email protected];tag=as612c1955’
[2017-04-10 21:20:01] VERBOSE[2473][C-00000000] app_dial.c: == Everyone is busy/congested at this time (1:0/0/1)
So thank you Dave, you’ve got me out of this prison but I’m unsure where the forbiden has come from as everything looks ok.
Right guys, my wife is threatening to divorce me if I don’t out this down now, so I’ll check back tomorrow if any of you bright sparks know what I’ve done wrong?
Thanks in advance and thanks again Dave!
Executing [s@macro-dialout-trunk:22] Dial(“SIP/001-00000000”, “SIP/Orbtalk/07744123456,300,Tt”) in new stack
[2017-04-10 21:20:01] VERBOSE[2473][C-00000000] netsock2.c: == Using SIP RTP TOS bits 184
[2017-04-10 21:20:01] VERBOSE[2473][C-00000000] netsock2.c: == Using SIP RTP CoS mark 5
[2017-04-10 21:20:01] VERBOSE[2473][C-00000000] app_dial.c: – Called SIP/Orbtalk/07744123456
[2017-04-10 21:20:01] WARNING[1784][C-00000000] chan_sip.c: Received response: “Forbidden” from ‘sip:**username**%20fromdomain=sip.orbtalk.co.uk%[email protected];tag=as612c1955’
[2017-04-10 21:20:01] VERBOSE[2473][C-00000000] app_dial.c: == Everyone is busy/congested at this time (1:0/0/1)
[2017-04-10 21:20:01] VERBOSE[2473][C-00000000] pbx.c: – Executing [s@macro-dialout-trunk:23] NoOp(“SIP/001-00000000”, “Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 21”) in new stack
[2017-04-10 21:20:01] VERBOSE[2473][C-00000000] pbx.c: – Executing [s@macro-dialout-trunk:24] GotoIf(“SIP/001-00000000”, “0?continue,1:s-CHANUNAVAIL,1”) in new stack
[2017-04-10 21:20:01] VERBOSE[2473][C-00000000] pbx.c: – Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
[2017-04-10 21:20:01] VERBOSE[2473][C-00000000] pbx.c: – Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set(“SIP/001-00000000”, “RC=21”) in new stack
[2017-04-10 21:20:01] VERBOSE[2473][C-00000000] pbx.c: – Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto(“SIP/001-00000000”, “21,1”) in new stack
[2017-04-10 21:20:01] VERBOSE[2473][C-00000000] pbx.c: – Goto (macro-dialout-trunk,21,1)
[2017-04-10 21:20:01] VERBOSE[2473][C-00000000] pbx.c: – Executing [21@macro-dialout-trunk:1] Goto(“SIP/001-00000000”, “continue,1”) in new stack
[2017-04-10 21:20:01] VERBOSE[2473][C-00000000] pbx.c: – Goto (macro-dialout-trunk,continue,1)
[2017-04-10 21:20:01] VERBOSE[2473][C-00000000] pbx.c: – Executing [continue@macro-dialout-trunk:1] NoOp(“SIP/001-00000000”, “TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 21 - failing through to other trunks”) in new stack
[2017-04-10 21:20:01] VERBOSE[2473][C-00000000] pbx.c: – Executing [continue@macro-dialout-trunk:2] Set(“SIP/001-00000000”, “CALLERID(number)=001”) in new stack
[2017-04-10 21:20:01] VERBOSE[2473][C-00000000] pbx.c: – Executing [07744123456@from-internal:8] Macro(“SIP/001-00000000”, “outisbusy,”) in new stack
[2017-04-10 21:20:01] VERBOSE[2473][C-00000000] pbx.c: – Executing [s@macro-outisbusy:1] Progress(“SIP/001-00000000”, “”) in new stack
[2017-04-10 21:20:01] VERBOSE[2473][C-00000000] pbx.c: – Executing [s@macro-outisbusy:2] GotoIf(“SIP/001-00000000”, “0?emergency,1”) in new stack
[2017-04-10 21:20:01] VERBOSE[2473][C-00000000] pbx.c: – Executing [s@macro-outisbusy:3] GotoIf(“SIP/001-00000000”, “0?intracompany,1”) in new stack
[2017-04-10 21:20:01] VERBOSE[2473][C-00000000] pbx.c: – Executing [s@macro-outisbusy:4] Playback(“SIP/001-00000000”, “all-circuits-busy-now&pls-try-call-later, noanswer”) in new stack
[2017-04-10 21:20:01] VERBOSE[2473][C-00000000] file.c: – <SIP/001-00000000> Playing ‘all-circuits-busy-now.ulaw’ (language ‘en’)
[2017-04-10 21:20:03] VERBOSE[2473][C-00000000] file.c: – <SIP/001-00000000> Playing ‘pls-try-call-later.ulaw’ (language ‘en’)
[2017-04-10 21:20:04] VERBOSE[2473][C-00000000] app_macro.c: == Spawn extension (macro-outisbusy, s, 4) exited non-zero on ‘SIP/001-00000000’ in macro ‘outisbusy’
[2017-04-10 21:20:04] VERBOSE[2473][C-00000000] pbx.c: == Spawn extension (from-internal, 07744123456, 8) exited non-zero on ‘SIP/001-00000000’
[2017-04-10 21:20:04] VERBOSE[2473][C-00000000] pbx.c: – Executing [h@from-internal:1] Hangup(“SIP/001-00000000”, “”) in new stack
[2017-04-10 21:20:04] VERBOSE[2473][C-00000000] pbx.c: == Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/001-00000000’
[2017-04-10 21:20:09] NOTICE[1784] chan_sip.c: – Registration for ‘username@sip.orbtalk.co.uk’ timed out, trying again (Attempt #9)
[2017-04-10 21:20:09] NOTICE[1784] chan_sip.c: Failed to authenticate on REGISTER to ‘username@sip.orbtalk.co.uk’ (Tries 3)
[2017-04-10 21:20:29] NOTICE[1784] chan_sip.c: – Registration for ‘username@sip.orbtalk.co.uk’ timed out, trying again (Attempt #13)
[2017-04-10 21:20:29] NOTICE[1784] chan_sip.c: Failed to authenticate on REGISTER to ‘username@sip.orbtalk.co.uk’ (Tries 3)
It’s not the %20, daft mistake which I corrected and It’s still not working 
“Forbidden” is easy. Either your username is spelled wrong, your password is spelled wrong, or they aren’t expecting this user/pass combo to come from this IP address.
If it was me, I’d get on the phone with my ITSP and say “Hey, guys, I’m getting a ‘Forbidden’ error when I try to connect. Can you tell me which piece I’ve cobbled up?”
Doh! Noob alert! Your right Dave, an erroneous double character in my password has at least allowed the call to Orbtalk but still not out.
The call waits for a time in silence then anounces that the other party did not answer, even though the other phone did not ring.
[2017-04-11 11:18:31] WARNING[1784] chan_sip.c: Retransmission timeout reached on transmission [email protected] for seqno 102 (Critical Request) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
[2017-04-11 11:18:31] WARNING[1784] chan_sip.c: Hanging up call [email protected] - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
[2017-04-11 11:18:31] VERBOSE[26189][C-000000ee] app_dial.c: == Everyone is busy/congested at this time (1:0/0/1)
[2017-04-11 11:18:31] VERBOSE[26189][C-000000ee] pbx.c: – Executing [s@macro-dialout-trunk:23] NoOp(“SIP/001-000000f5”, “Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 18”) in new stack
[2017-04-11 11:18:31] VERBOSE[26189][C-000000ee] pbx.c: – Executing [s@macro-dialout-trunk:24] GotoIf(“SIP/001-000000f5”, “0?continue,1:s-CHANUNAVAIL,1”) in new stack
[2017-04-11 11:18:31] VERBOSE[26189][C-000000ee] pbx.c: – Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
[2017-04-11 11:18:31] VERBOSE[26189][C-000000ee] pbx.c: – Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set(“SIP/001-000000f5”, “RC=18”) in new stack
[2017-04-11 11:18:31] VERBOSE[26189][C-000000ee] pbx.c: – Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto(“SIP/001-000000f5”, “18,1”) in new stack
[2017-04-11 11:18:31] VERBOSE[26189][C-000000ee] pbx.c: – Goto (macro-dialout-trunk,18,1)
[2017-04-11 11:18:31] VERBOSE[26189][C-000000ee] pbx.c: – Executing [18@macro-dialout-trunk:1] Goto(“SIP/001-000000f5”, “s-NOANSWER,1”) in new stack
[2017-04-11 11:18:31] VERBOSE[26189][C-000000ee] pbx.c: – Goto (macro-dialout-trunk,s-NOANSWER,1)
[2017-04-11 11:18:31] VERBOSE[26189][C-000000ee] pbx.c: – Executing [s-NOANSWER@macro-dialout-trunk:1] NoOp(“SIP/001-000000f5”, “Dial failed due to trunk reporting NOANSWER - giving up”) in new stack
[2017-04-11 11:18:31] VERBOSE[26189][C-000000ee] pbx.c: – Executing [s-NOANSWER@macro-dialout-trunk:2] Progress(“SIP/001-000000f5”, “”) in new stack
[2017-04-11 11:18:31] VERBOSE[26189][C-000000ee] pbx.c: – Executing [s-NOANSWER@macro-dialout-trunk:3] Playback(“SIP/001-000000f5”, “number-not-answering,noanswer”) in new stack
[2017-04-11 11:18:31] VERBOSE[26189][C-000000ee] file.c: – <SIP/001-000000f5> Playing ‘number-not-answering.ulaw’ (language ‘en’)
[2017-04-11 11:18:33] VERBOSE[26189][C-000000ee] pbx.c: – Executing [s-NOANSWER@macro-dialout-trunk:4] Congestion(“SIP/001-000000f5”, “20”) in new stack
[2017-04-11 11:18:33] WARNING[26189][C-000000ee] channel.c: Prodding channel ‘SIP/001-000000f5’ failed
[2017-04-11 11:18:33] VERBOSE[26189][C-000000ee] app_macro.c: == Spawn extension (macro-dialout-trunk, s-NOANSWER, 4) exited non-zero on ‘SIP/001-000000f5’ in macro ‘dialout-trunk’
[2017-04-11 11:18:33] VERBOSE[26189][C-000000ee] pbx.c: == Spawn extension (from-internal, 07711331000, 7) exited non-zero on ‘SIP/001-000000f5’
[2017-04-11 11:18:33] VERBOSE[26189][C-000000ee] pbx.c: – Executing [h@from-internal:1] Hangup(“SIP/001-000000f5”, “”) in new stack
[2017-04-11 11:18:33] VERBOSE[26189][C-000000ee] pbx.c: == Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/001-000000f5’
See Dave, you’re making me look 
Checked my Asterisk full and found NAT=Yes to be depreciated so using force_rport,comedia and guess what? I made my first call out woooohooooooo!
Now I need to see why I can’t get calls in lol
WARNING,“Rejecting unknown SIP connection from 89.163.144.171"”) in new stack
This seems to be my issue, any ideas?
Ok, kept the DID in my Inbound route and removed the CID and now everything is working.
Thanks everyone for their contributions… oh wait!! That’s just Dave!!!
THANK YOU MR BURGESS, I owe you a pint if you’re ever in Glasgow!!
I did see you mention Lorne for the from-internal but I hadn’t seen their post, Have I broken the forum too??
In any case, thank you the mysterious Lorne