Route calls out of FreePBX trunk

[Ace2020]

Hello, new to the forum and don't quite know the rules.

I have a Vicidial system already setup. I have a FreePBX system already setup. Also have a IAX2 truck setup between the severs. Now, how do I edit the dial plan in order for my ROBO calls to be routed out of the FreePBX trunk? Any help would be greatly appreciated. TOTAL NOOB TO VICIDIAL

[williamconley]

1) Welcome to the Party!

2) As you are obviously new here, I have some suggestions to help us all help you:

When you post, please post your entire configuration including (but not limited to) your installation method (7.X.X?) and vicidial version with build (VERSION: 2.X-XXXx ... BUILD: #####-####).

This IS a requirement for posting along with reading the stickies (at the top of each forum) and the manager's manual (available on EFLO.net, both free and paid versions)

You should also post: Asterisk version, telephony hardware (model number is helpful here), cluster information if you have one, and whether any other software is installed in the box. If your installation method is "manual/from scratch" you must post your operating system with version (and the .iso version from which you installed your original operating system) plus a link to the installation instructions you used. If your installation is "Hosted" list the site name of the host.

If this is a "Cloud" or "Virtual" server, please note the technology involved along with the version of that techology (ie: VMware Server Version 2.0.2). If it is not, merely stating the Motherboard model # and CPU would be helpful.

Similar to This:

Vicibox X.X from .iso | Vicidial X.X.X-XXX Build XXXXXX-XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation | Intel DG35EC | Core2Quad Q6600

3) Method One (NOT Recommended, but definitely works, there are other similar methods): FreePBX considers Vicidial an "Extension" (which, BTW, it is, just like any other soft phone or IP phone, technically there is no difference to FreePBX if you connect and dial via Zoiper or Vicidial). Vicidial considers FreePBX a "Carrier". That's because when you connect Vicidial to an extension in FreePBX, FreePBX *IS* the carrier. No different than any other carrier. So now you need to follow normal instructions to allow a number dialed by Vicidial to the FreePBX system to flow out to the FreePBX trunk and you're done. WARNING: Vicidial will blow away a FreePBX server pretty quickly if you actually load it up.

Method Two (much better): Use the Trunk you have connected to FreePBX as a Carrier in Vicidial and skip FreePBX entirely! They are both asterisk-based PBXs and redundant. Kinda like putting your Harley-Davidson in the back seat of your convertible. Definitely turn some heads, especially if you finely craft an most-excellent mount to "showcase" the Harley. And you'll probably be able to get it up to 60MPH. But ... I don't recommend sudden stops or higher speeds. LOL.

Option: If you wanted to be fun, you could use the IAX link to allow agents to use their FreePBX extensions as external phones in Vicidial, thus allowing a Login for Vicidial to ring the FreePBX phone and removing the need for the agents to connect directly to Vicidial. The advantage of that is that you gain the features (FOP especially) of FreePBX for administration of the office. Great if you're integrating a Dialer into an already-running sales office with a lot of non-sales/non-callcenter personnel already in FreePBX.

However: If this is a fresh setup and you don't have "admin personnel", skip FreePBX entirely. It's pointless in that environment.

Happy Hunting

[Ace2020]

ok, thank you for the speedy reply. Where can I locate the system information for VICIDial so I can include it into the post? And the goal of this entire project is to test our ROBO calling. Only using the FreePBX server because I have configured a Google Voice trunk on it for FREE calls. As I stated earlier, I'm a complete noob here. Could you provide some links or a walk threw to get me setup please?

[Ace2020]

And after its configured properly, where is the option to play a recording once the call has been answered? To actually make a ROBO call?

[williamconley]

ok, thank you for the speedy reply. Where can I locate the system information for VICIDial so I can include it into the post? And the goal of this entire project is to test our ROBO calling. Only using the FreePBX server because I have configured a Google Voice trunk on it for FREE calls. As I stated earlier, I'm a complete noob here. Could you provide some links or a walk threw to get me setup please?

Vicibox X.X from .iso | Available usually from the Splash / MOTD screen when you access the server's console or via SSH/PuTTY
Vicidial X.X.X-XXX Build XXXXXX-XXXX | Bottom left corner of almost every admin screen.
Asterisk X.X.X | "asterisk -V"
Single Server | I hope you know this.
No Digium/Sangoma Hardware | If you purchased any of these, you should know (include the model number)
No Extra Software After Installation | Avoid this until your system is operational if you can.
Intel DG35EC | Motherboard Model Number. Often available via "dmidecode | grep Product"
Core2Quad Q6600 | CPU often available via "cat /proc/cpuinfo | grep 'model name' " (listing Core Count, model, and speed is overkill but can be helpful)

ONE line for robodialing is only useful for testing, of course. No overload will occur. But when you get a real carrier ... be prepared to remove FreePBX from the pathway. If you intend to leave it at one (or a couple) channels forever, no problem. But note that Google will cut you off. Inevitable.

1) Install Vicidial using the Vicibox.com .iso. Use the PDF on the Vicibox.com site for installation instructions.

2) When installation is complete, switch to the Vicidial Manager's Manual for configuration instructions. Start at page one. Don't skip anything. Get it up and running and online exactly as the manual says to do it all the way through without any changes before you try to personalize the experience. If you're tempted to skip something because you don't need it, slap yourself. Do it anyway. You are using $250k of software without paying a dime. Put in the time and it'll work out. Trust me.

When you decide to try to outsmart the world: Vicidial documentation can also be found at /usr/src/astguiclient/trunk/docs

Very valuable information can be gathered by running all the perl scripts in /usr/share/astguiclient with "perl SCRIPTNAME.pl --help" Many of these scripts also have notes in the beginning of the script, so using "cat" or "nano" to read those notes can be very useful.

The same can be said for all the scripts in /srv/www/htdocs/agc and ../htdocs/vicidial except they are php scripts and accessed via web instead of using perl at the command line.

And read the manual. Seriously consider purchasing the paid version, it has better graphics and a couple hundred more pages. However: The paid version is not needed to get the server up and dialing (robo, auto, predictive, survey, broadcast ... no need to spend money until you want to "go deep").

Happy Hunting.

[williamconley]

And after its configured properly, where is the option to play a recording once the call has been answered? To actually make a ROBO call?

One step at a time in the manual. You'll get there eventually. Baby steps my friend, baby steps.

PS: That's called a "Survey". Available under "Campaigns" when modifying any campaign, switch to detail view and there will now be a Survey top menu option. The settings in the Survey menu are only used when the Campaign Routing Extension is set to the proper value. For more information (ie: Which extension to set ...) click on the ? next to the Campaign Routing Extension in the Detail View of any campaign.

word of advice: Don't change anything in the Survey at first. Let it run with the default settings. Prove it works (it'll ask democrat vs republican! then transfer you to an agent.). After it works, change ONE thing and test again. Every time you change something, test. If it stops working ... change it back! Then figure out what went wrong (but first test to be sure your "undo" was successful, especially if you've had a lot of caffeine or it's after 4AM).

[Ace2020]

Thank you so much for the taking the time out to respond. I'm about to give it a shot right now. Think I'll reinstall it and start from scratch. Wish me luck...I'll keep you posted.

[williamconley]

Thank you so much for the taking the time out to respond. I'm about to give it a shot right now. Think I'll reinstall it and start from scratch. Wish me luck...I'll keep you posted.

A very wise choice. In fact, one of the things I make new technicians do several times. NEVER fear the reinstall. If you get it right on the 8th time, you also realize at that point how easy it is to start over. No fear. Especially once you learn about the easy backup methods.

Also have a look at updating (don't do it unless you have a reason, though, like "server's not online yet, wonder what happens if I update it?" just to see ... but never on a live box without reason). OH: and by updating I refer to Vicidial by using the /usr/src/astguiclient/trunk/UPDATE instructions. Not the OS. Never update/upgrade the OS unless you have an overwhelming reason to do so.

And whitelist lock your iptables firewall on the server. Nobody should even see that your server exists if they aren't an Authorized IP address. Look up Dynamic Good Guys if you're not sure.

[Ace2020]

going through the manual trying to get things up and running, buy what Trunk am I NOT dialing out of? Everything setup wise is working, just disconnects and gives me an error message once it tries to dial out. Made is all the way to pg 19. Is this where I have to enter my own TRUNK in order to place test calls?

[williamconley]

... buy what Trunk am I NOT dialing out of?

I don't understand this question.

Everything setup wise is working, just disconnects and gives me an error message once it tries to dial out.

So ... to be clear ... it works but it doesn't work and there's an error message? Well, that should be easy to resolve. Of course, having the error message would be good. Knowing where the error message is viewed may also be handy.

Posting the relevent settings for this step (whatever this step is ...) along with what you believed should happen and what actually did happen would be the proper method to ask for technical assistance.

I realize you're in unfamiliar territory and have likely been staring at this for a long time. We don't know what you're staring at, and if you don't bring us up to speed we never will.

Made is all the way to pg 19. Is this where I have to enter my own TRUNK in order to place test calls?

? I'm lost. You're on "page 19", and looking at it, but you're asking US what's on page 19?

[Ace2020]

Vicibox 8 from .iso | VERSION: 2.14-650a BUILD: 180111-1544 | Asterisk 11.25.3 | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation | OS X 13.2 | Oracle VM

Sorry for not being clear, I have gotten the system up and running all the way up to the point it starts dialing out of a trunk. Hence the bad joke about what Trunk wasn't I dialing out of. That's my problem, I'm trying to dial out of my IAX2 trunk or a SIP trunk. Whichever is easier. Nothing I have found in the manual actually addresses this issue. It assumes that you already have a working trunk to dial from before starting the manual. So I am assuming this is just a dial plan issue? Or can I setup some sort of remote extension from my FreePBX machine that I can dial from with no problem? Is it just a simple dial plan thing? Like I said, I'm trying something new here and any and all help would be appreciated. I'll also include my settings if that helps any.

Vicibox IP: 192.168.0.33 FreePBX IP: 192.168.0.18
trying to dial from the 192.168.0.33 server out of trunks already setup on 192.168.0.18. Thats all. Someone please help

[img]https://ibb.co/fRTqP6/img]

[williamconley]

You're getting closer ace, but you left out any details. Nice brochure, don't get me wrong, but posting the Carrier settings would be a start. Posting the asterisk CLI output from a single attempt would also be useful.

Advice that's free requires details. Of course, there are hundreds (thousands?) of previous Carrier resolution posts, any number of which may contain the solution to your problem.

Normally, I advise that you share the page/line/version of the manual where you hit your snag, along with the pertinent configuration settings (related to the issue) and "what happened" (detailed logs of what happened, not a statement like "it didn't work") and what you expected to happen differently (that could be a statement, of course, like "a completed call instead of this terminated without dialing call ...).

[Ace2020]

I thought I included the carrier setting. "https://ibb.co/fRTqP6/img".
You stated in your first message that I could do,
"Method One (NOT Recommended, but definitely works, there are other similar methods): FreePBX considers Vicidial an "Extension" (which, BTW, it is, just like any other soft phone or IP phone, technically there is no difference to FreePBX if you connect and dial via Zoiper or Vicidial). Vicidial considers FreePBX a "Carrier". That's because when you connect Vicidial to an extension in FreePBX, FreePBX *IS* the carrier. No different than any other carrier. So now you need to follow normal instructions to allow a number dialed by Vicidial to the FreePBX system to flow out to the FreePBX trunk and you're done. WARNING: Vicidial will blow away a FreePBX server pretty quickly if you actually load it up."
Where can I find information on setting this up? But how much would help for an issue like this cost if all else fails?

[williamconley]

Progress!

1) Globals String has not been filled in. Should be "TRUNK1 = IAX/Free_Trunk_1". Note: This defines a VARIABLE which converts to the full string to dial instead of putting the Account Entry directly in the Dial command (allowing easy changing of multiple carrier routes at once by altering a single variable).
2) exten => 201 should be exten => NXXNXXXXXX on all three lines. 201 is a literal and will ONLY match "201" whereas NXXNXXXXXX will match any US 10 digit telephone number. You can also create a 2nd and 3rd set of extensions with exten => 1NXXNXXXXXX and exten => 91NXXNXXXXXX to allow "1+10 digit" (long distance, we're all used to that) dialing and Dial Prefix 9 + (1+10 digit) dialing. This is a "choose your carrier via Dial Prefix" method which allows multiple simultaneous live carriers and the ability to choose among them via the "Dial Prefix" field in the campaign or by "dialing 9" before the number directly from a phone (holdover from the old days of "dial 9 to get an outside line, for us old people).
3) $ {Free_Trunk_1} is an attempt to use a variable that does not exist. After creating the TRUNK1 variable in 1) above, use instead ${TRUNK1}, which will expand to IAX/Free_Trunk_1 (with no dollar sign or brackets) which is the literal you could use instead of a variable if you want, but you can't use the $ or { because it's not a variable.
4) ${EXTEN:2} will remove the first two digits of the number dialed. (:2). Probably not what you want. Lose the :2, except in the 91NXXNXXXXXX extension set which will get a :1 to remove the 9.

Code: Select all

; Added in "1" after the "/" so the actual dialed number becomes "1+10 digit" standard US domestic LD.
exten => _NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _NXXNXXXXXX,2,Dial(${TRUNK1}/1${EXTEN},,To)
exten => _NXXNXXXXXX,3,Hangup

exten => _1NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _1NXXNXXXXXX,2,Dial(${TRUNK1}/${EXTEN},,To)
exten => _1NXXNXXXXXX,3,Hangup

; Added in ":1" after the "EXTEN" to remove the dial prefix (9).
exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(${TRUNK1}/${EXTEN:1},,To)
exten => _91NXXNXXXXXX,3,Hangup


5) No debug or command line output from Asterisk? Now I'm disappointed. You were doing so well, to.

[Ace2020]

Did exactly what you just said it this is the output once I try to place a call from the list I setup and imported.
VICIdial error was:
DIAL ALERT:

Call Rejected: CHANUNAVAIL
Cause: 66 - Channel type not implemented.


Close Message

5 iax2 peers [4 online, 1 offline, 0 unmonitored]
[Jan 18 16:02:37] -- Registered SIP '201' at 192.168.0.11:44699
[Jan 18 16:02:50] == Using SIP RTP CoS mark 5
[Jan 18 16:02:50] > 0x7f6910033b20 -- Strict RTP learning after remote address set to: 97.70.178.195:57420
[Jan 18 16:02:50] NOTICE[3084][C-00000005]: chan_sip.c:26002 handle_request_invite: Call from '201' (192.168.0.11:44699) to extension '13136628934' rejected because extension not found in context 'default'.
[Jan 18 16:03:02] == Manager 'sendcron' logged on from 127.0.0.1
[Jan 18 16:03:02] == Manager 'sendcron' logged on from 127.0.0.1
[Jan 18 16:03:02] == Manager 'sendcron' logged off from 127.0.0.1
[Jan 18 16:03:02] == Manager 'sendcron' logged off from 127.0.0.1
[Jan 18 16:03:07] == Manager 'sendcron' logged on from 127.0.0.1
[Jan 18 16:03:07] == Manager 'sendcron' logged off from 127.0.0.1
[Jan 18 16:04:01] == Manager 'sendcron' logged on from 127.0.0.1
[Jan 18 16:04:01] == Manager 'sendcron' logged on from 127.0.0.1
[Jan 18 16:04:01] == Manager 'sendcron' logged off from 127.0.0.1
[Jan 18 16:04:02] == Manager 'sendcron' logged off from 127.0.0.1
[Jan 18 16:04:02] == Parsing '/etc/asterisk/iax.conf': Found
[Jan 18 16:04:02] == Parsing '/etc/asterisk/iax-vicidial.conf': Found
[Jan 18 16:04:02] == Parsing '/etc/asterisk/users.conf': Found
[Jan 18 16:04:02] NOTICE[2937]: chan_iax2.c:13567 set_config: Ignoring bindport on reload
[Jan 18 16:04:02] NOTICE[2937]: chan_iax2.c:13628 set_config: Ignoring bindaddr on reload
[Jan 18 16:04:02] -- Seeding 'ASTloop' at 127.0.0.1:57046 for 60
[Jan 18 16:04:02] -- Seeding 'ASTblind' at 127.0.0.1:8225 for 60
[Jan 18 16:04:02] -- Seeding 'ASTplay' at 127.0.0.1:21899 for 60
[Jan 18 16:04:02] NOTICE[2937]: chan_iax2.c:12435 iax2_poke_peer: Still have a callno...
[Jan 18 16:04:02] NOTICE[2937]: chan_iax2.c:12435 iax2_poke_peer: Still have a callno...
[Jan 18 16:04:02] NOTICE[2937]: chan_iax2.c:12435 iax2_poke_peer: Still have a callno...
[Jan 18 16:04:02] -- Registered IAX2 to '127.0.0.1', who sees us as 127.0.0.1:21899 with no messages waiting
[Jan 18 16:04:02] -- Registered IAX2 to '127.0.0.1', who sees us as 127.0.0.1:57046 with no messages waiting
[Jan 18 16:04:02] -- Registered IAX2 to '127.0.0.1', who sees us as 127.0.0.1:8225 with no messages waiting
[Jan 18 16:04:06] == Manager 'sendcron' logged on from 127.0.0.1
[Jan 18 16:04:06] == Manager 'sendcron' logged off from 127.0.0.1
[Jan 18 16:05:02] == Manager 'sendcron' logged on from 127.0.0.1
[Jan 18 16:05:02] == Manager 'sendcron' logged off from 127.0.0.1
[Jan 18 16:05:02] == Manager 'sendcron' logged on from 127.0.0.1
[Jan 18 16:05:02] == Manager 'sendcron' logged off from 127.0.0.1
[Jan 18 16:05:07] == Manager 'sendcron' logged on from 127.0.0.1
[Jan 18 16:05:07] == Manager 'sendcron' logged off from 127.0.0.1
[Jan 18 16:05:43] NOTICE[3084]: chan_sip.c:29904 sip_poke_noanswer: Peer '201' is now UNREACHABLE! Last qualify: 8
[Jan 18 16:05:45] NOTICE[3084]: chan_sip.c:23869 handle_response_peerpoke: Peer '201' is now Reachable. (15ms / 2000ms)
[Jan 18 16:06:01] == Manager 'sendcron' logged on from 127.0.0.1
[Jan 18 16:06:01] == Manager 'sendcron' logged off from 127.0.0.1
[Jan 18 16:06:01] == Manager 'sendcron' logged on from 127.0.0.1
[Jan 18 16:06:02] == Manager 'sendcron' logged off from 127.0.0.1
[Jan 18 16:06:06] == Manager 'sendcron' logged on from 127.0.0.1
[Jan 18 16:06:06] == Manager 'sendcron' logged off from 127.0.0.1
[Jan 18 16:06:49] NOTICE[3084]: chan_sip.c:29904 sip_poke_noanswer: Peer '201' is now UNREACHABLE! Last qualify: 15
[Jan 18 16:06:50] NOTICE[3084]: chan_sip.c:23869 handle_response_peerpoke: Peer '201' is now Reachable. (6ms / 2000ms)
[Jan 18 16:07:02] == Manager 'sendcron' logged on from 127.0.0.1
[Jan 18 16:07:02] == Manager 'sendcron' logged on from 127.0.0.1
[Jan 18 16:07:02] == Manager 'sendcron' logged off from 127.0.0.1
[Jan 18 16:07:02] == Manager 'sendcron' logged off from 127.0.0.1
[Jan 18 16:07:07] == Manager 'sendcron' logged on from 127.0.0.1
[Jan 18 16:07:07] == Manager 'sendcron' logged off from 127.0.0.1
[Jan 18 16:07:54] NOTICE[3084]: chan_sip.c:29904 sip_poke_noanswer: Peer '201' is now UNREACHABLE! Last qualify: 6
[Jan 18 16:08:01] == Manager 'sendcron' logged on from 127.0.0.1
[Jan 18 16:08:01] == Manager 'sendcron' logged off from 127.0.0.1
[Jan 18 16:08:01] == Manager 'sendcron' logged on from 127.0.0.1
[Jan 18 16:08:01] == Manager 'sendcron' logged off from 127.0.0.1
[Jan 18 16:08:06] == Manager 'sendcron' logged on from 127.0.0.1
[Jan 18 16:08:06] == Manager 'sendcron' logged off from 127.0.0.1
[Jan 18 16:08:56] == Manager 'sendcron' logged on from 127.0.0.1
[Jan 18 16:08:56] NOTICE[11304][C-00000006]: channel.c:5690 __ast_request_and_dial: Unable to request channel SIP/201
[Jan 18 16:08:57] == Manager 'sendcron' logged off from 127.0.0.1
[Jan 18 16:09:01] -- Registered SIP '201' at 192.168.0.11:44699
[Jan 18 16:09:01] NOTICE[3084]: chan_sip.c:23869 handle_response_peerpoke: Peer '201' is now Reachable. (3ms / 2000ms)
[Jan 18 16:09:02] == Manager 'sendcron' logged on from 127.0.0.1
[Jan 18 16:09:02] == Manager 'sendcron' logged off from 127.0.0.1
[Jan 18 16:09:02] == Manager 'sendcron' logged on from 127.0.0.1
[Jan 18 16:09:02] == Manager 'sendcron' logged off from 127.0.0.1
[Jan 18 16:09:07] == Manager 'sendcron' logged on from 127.0.0.1
[Jan 18 16:09:07] == Manager 'sendcron' logged off from 127.0.0.1
[Jan 18 16:09:20] == Manager 'sendcron' logged on from 127.0.0.1
[Jan 18 16:09:20] == Manager 'sendcron' logged on from 127.0.0.1
[Jan 18 16:09:20] -- Executing [55558600051@default:1] MeetMeAdmin("Local/55558600051@default-00000000;2", "8600051,K") in new stack
[Jan 18 16:09:20] WARNING[13088][C-00000007]: app_meetme.c:5053 admin_exec: Conference number '8600051' not found!
[Jan 18 16:09:20] -- Executing [55558600051@default:2] Hangup("Local/55558600051@default-00000000;2", "") in new stack
[Jan 18 16:09:20] == Spawn extension (default, 55558600051, 2) exited non-zero on 'Local/55558600051@default-00000000;2'
[Jan 18 16:09:20] -- Executing [h@default:1] AGI("Local/55558600051@default-00000000;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Jan 18 16:09:20] -- <Local/55558600051@default-00000000;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Jan 18 16:09:21] == Manager 'sendcron' logged off from 127.0.0.1
[Jan 18 16:09:21] == Manager 'sendcron' logged off from 127.0.0.1
[Jan 18 16:09:59] == Manager 'sendcron' logged on from 127.0.0.1
[Jan 18 16:09:59] == Using SIP RTP CoS mark 5
[Jan 18 16:10:01] == Manager 'sendcron' logged on from 127.0.0.1
[Jan 18 16:10:01] == Manager 'sendcron' logged off from 127.0.0.1
[Jan 18 16:10:01] == Manager 'sendcron' logged on from 127.0.0.1
[Jan 18 16:10:02] == Manager 'sendcron' logged off from 127.0.0.1
[Jan 18 16:10:06] == Manager 'sendcron' logged on from 127.0.0.1
[Jan 18 16:10:06] == Manager 'sendcron' logged off from 127.0.0.1
[Jan 18 16:10:07] == Manager 'sendcron' logged on from 127.0.0.1
[Jan 18 16:10:07] -- Executing [8600051@default:1] MeetMe("Local/8600051@default-00000001;2", "8600051,F") in new stack
[Jan 18 16:10:07] > Channel Local/8600051@default-00000001;1 was answered
[Jan 18 16:10:07] -- Executing [912485959696@default:1] AGI("Local/8600051@default-00000001;1", "agi://127.0.0.1:4577/call_log") in new stack
[Jan 18 16:10:07] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=TESTCAMP))
[Jan 18 16:10:07] -- <Local/8600051@default-00000001;1>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Jan 18 16:10:07] -- Executing [912485959696@default:2] Dial("Local/8600051@default-00000001;1", "newsip:test@192.168.0.33:5060/2485959696,,tTor") in new stack
[Jan 18 16:10:07] WARNING[16219][C-00000009]: channel.c:6009 ast_request: No channel type registered for 'newsip:test@192.168.0.33:5060'
[Jan 18 16:10:07] WARNING[16219][C-00000009]: app_dial.c:2455 dial_exec_full: Unable to create channel of type 'newsip:test@192.168.0.33:5060' (cause 66 - Channel not implemented)
[Jan 18 16:10:07] == Everyone is busy/congested at this time (1:0/0/1)
[Jan 18 16:10:07] -- Executing [912485959696@default:3] Hangup("Local/8600051@default-00000001;1", "") in new stack
[Jan 18 16:10:07] == Spawn extension (default, 912485959696, 3) exited non-zero on 'Local/8600051@default-00000001;1'
[Jan 18 16:10:07] -- Executing [h@default:1] AGI("Local/8600051@default-00000001;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----66-----CHANUNAVAIL----------") in new stack
[Jan 18 16:10:07] -- <Local/8600051@default-00000001;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Jan 18 16:10:07] == Parsing '/etc/asterisk/meetme.conf': Found
[Jan 18 16:10:07] == Parsing '/etc/asterisk/meetme-vicidial.conf': Found
[Jan 18 16:10:07] -- Created MeetMe conference 1023 for conference '8600051'
[Jan 18 16:10:07] WARNING[16218][C-00000009]: file.c:830 ast_readaudio_callback: Failed to write frame
[Jan 18 16:10:07] -- <Local/8600051@default-00000001;2> Playing 'conf-onlyperson.gsm' (language 'en')
[Jan 18 16:10:07] -- Hungup 'DAHDI/pseudo-1029578993'
[Jan 18 16:10:07] == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-00000001;2'
[Jan 18 16:10:07] -- Executing [h@default:1] AGI("Local/8600051@default-00000001;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Jan 18 16:10:07] -- <Local/8600051@default-00000001;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Jan 18 16:10:08] == Manager 'sendcron' logged off from 127.0.0.1
[Jan 18 16:10:15] > 0x7f69000172e0 -- Strict RTP learning after remote address set to: 97.70.178.195:57420
[Jan 18 16:10:15] > Channel SIP/201-00000000 was answered
[Jan 18 16:10:15] -- Executing [8600051@default:1] MeetMe("SIP/201-00000000", "8600051,F") in new stack
[Jan 18 16:10:15] == Parsing '/etc/asterisk/meetme.conf': Found
[Jan 18 16:10:15] == Parsing '/etc/asterisk/meetme-vicidial.conf': Found
[Jan 18 16:10:15] -- Created MeetMe conference 1023 for conference '8600051'
[Jan 18 16:10:15] -- <SIP/201-00000000> Playing 'conf-onlyperson.gsm' (language 'en')
[Jan 18 16:10:16] == Manager 'sendcron' logged off from 127.0.0.1
[Jan 18 16:10:22] > 0x7f69000172e0 -- Strict RTP switching source address to 192.168.0.11:57420
[Jan 18 16:10:22] > 0x7f69000172e0 -- Strict RTP learning complete - Locking on source address 192.168.0.11:57420
[Jan 18 16:10:28] -- Hungup 'DAHDI/pseudo-1361466509'
[Jan 18 16:10:28] == Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/201-00000000'
[Jan 18 16:10:28] -- Executing [h@default:1] AGI("SIP/201-00000000", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Jan 18 16:10:28] -- <SIP/201-00000000>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Jan 18 16:11:0

[williamconley]

[Jan 18 16:10:07] -- Executing [912485959696@default:2] Dial("Local/8600051@default-00000001;1", "newsip:test@192.168.0.33:5060/2485959696,,tTor") in new stack
[Jan 18 16:10:07] WARNING[16219][C-00000009]: channel.c:6009 ast_request: No channel type registered for 'newsip:test@192.168.0.33:5060'

I'm not sure what you did to initiate this call, but it's not using the code I provided you to attempt to make the call.

newsip:test@192.168.0.33 is nowhere in what we've discussed. Thus we are missing some fairly necessary information. Where is newsip:test@192.168.0.33? How did you attempt to make a call?

[Ace2020]

that was the test trunk they had me setup in the manual. how do I change it?

[williamconley]

The sample trunk was using sample information. Obviously they can't print a "one-size-fits-all" trunk. You have to modify that trunk to meet your needs.

The trunk needs to be connected to the FreePBX server not "newsip:test@192.168.0.33", unless you have an account on the FreePBX server that will accept "newsip:test" as credentials and the FreePBX server is at 192.168.0.33.

Regardless of which: Your post (ibb.co/fRTqP6/img) showed completely different credentials, IP, etc. That shows the host at 0.18, not 33, the user is pbx1 and the pass is diamonds2020. None of which is newsip, test, or 0.33 (in fact, this server appeas to be 0.33 ... so something is amiss).

So ... my earlier question of "how are you initiating this call" is relevant.

While we're here:

[Free_Trunk_1]'s "context=" should ALWAYS (in all account entry contexts) be "trunkinbound".

context=trunkinbound

Always. There is NO situation in which that should differ for a stock installation. When that changes, it'll be because you've been running the server for a few years and found a really cool think you can do with inbound and want to make an adjustment. lol

You may want to repost your carrier configuration along with "how you initiated this call". Perhaps your page number and line number of the manual to get you back on track.

Before you go nuts: Remember that this is the one thing that can't be comprehensively documented. Each user has the ability to use any one of several thousand (million?) carriers. Thus documentation has to be more along the lines of "kinda like this". In the end, the SIP Account (in the "Account Entry" field) is just a SIP context which is very well documented throughout Asterisk and FreePBX sites worldwide. So, convince the server to register or IP link via the Account Entry using "other references", and then come here and we can get it going. Or continue as you are now and we'll get you there. Eventually. It's quite possibly (if not likely) a simple issue concerning how you initiated the call. Oh: And turn off all the carriers that aren't this one. Conflicting Dialplan Entries are annoying.

[Ace2020]

deleted that trunk all together and this is the new output.

[Jan 18 16:53:02] > 0x7f69100322e0 -- Strict RTP learning after remote address set to: 97.70.178.195:39816
[Jan 18 16:53:02] -- Executing [913136628934@default:1] AGI("SIP/201-00000008", "agi://127.0.0.1:4577/call_log") in new stack
[Jan 18 16:53:02] -- <SIP/201-00000008>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Jan 18 16:53:02] -- Executing [913136628934@default:2] Dial("SIP/201-00000008", "IAX/Free_Trunk_1/13136628934,,To") in new stack
[Jan 18 16:53:02] WARNING[25193][C-0000001f]: channel.c:6009 ast_request: No channel type registered for 'IAX'
[Jan 18 16:53:02] WARNING[25193][C-0000001f]: app_dial.c:2455 dial_exec_full: Unable to create channel of type 'IAX' (cause 66 - Channel not implemented)
[Jan 18 16:53:02] == Everyone is busy/congested at this time (1:0/0/1)
[Jan 18 16:53:02] -- Executing [913136628934@default:3] Hangup("SIP/201-00000008", "") in new stack
[Jan 18 16:53:02] == Spawn extension (default, 913136628934, 3) exited non-zero on 'SIP/201-00000008'
[Jan 18 16:53:02] -- Executing [h@default:1] AGI("SIP/201-00000008", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----66-----CHANUNAVAIL----------") in new stack
[Jan 18 16:53:02] -- <SIP/201-00000008>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Jan 18 16:53:06] == Manager 'sendcron' logged on from 127.0.0.1
[Jan 18 16:53:06] == Manager 'sendcron' logged off from 127.0.0.1

[williamconley]

I don't see your new vicidial SIP or IAX2 Carrier entry. I prefer SIP, BTW, it's simpler for most situations. Work out IAX later. Unless you don't know how to SIP in FreePBX, of course.

I also think you may have IAX where IAX2 belongs if it can't find the type.