I am having problems with out going calls

I am new at this, but have been working on this program or about 4 weeks now. I am using Free PBX.

I have extension to extension working fine, I have paging working fine, I have IVR working fine and I am getting incoming calls fine.

This is the problem, When I make a out going call. I will dial the number and hit send, the audio goes blank (as if it goes dead) for about 20 seconds then I get a recording saying “The number is not answering” then I get a busy signal. I am calling my cell phone and it is not ringing, I have also tried other number and still the same thing.

I have worked on this for two or three days now, and did plenty of research and could not find anyone else having this problem. I hope some one out there will be able to give me some suggestions. I am so close not to have a operating system.

I still have the “Could not reload the FOP server” Error, but I can use the fop module. Would this have any thing to do with it?

Thanks for any help,
Daniel Bo

What is the configuration for your outgoing “trunk”? As you have asked on an Asterisk forum, rather than a FreePBX termm, you need to give this in Asterisk terms (no GUI) not FreePBX ones.

What is the CLI output at at least verbosity 3?

I am really only looking at freePBX. I understand what you are saying. I am trying to learn to use freePBX, but also picking up stuff on asterisks. I can show you my config on PBX, but dont know how to show you it in asterisks. I am probably not making since to someone that knows much more then I do. If you could explain to me how and what you need, it would be great to learn how to do this and I will do it. Thanks for any help you can assist me with my problem.


So what sort of trunk are you using? Do you have a SIP or IAX2 trunk over an Internet connection or do you have some hardware installed for an analogue or ISDN-2 trunk. Have you tried the FreePBX forum as well. They are also very helpful there…

I am using SIP trunk, and I am also working with people on FreePBX. I am just trying to find some help. I have read about half and half saying this it will not work with vonage and it will work. I have entered every one of the set ups that they have suggested that would work. This is my most current.
secret=9XXXXXXXXX (this is the password that vonage gave me.

Have you created a custom context call vonage?

insecure=very is no longer recognized by currently supported versions of Asterisk, although it was normally wrongly used before then. However this will tend to cause incoming calls to fail rather than outging ones.

Also, most GUIs create a separate user and peer entry, which should avoid that issue.

If you use a single entry, which would normally be peer, rather than friend, it would normally need insecure=invite, because the ITSP will not try to authenticate itself to you, but will not try to register, either.

You may, of course, have installed an obsolete version.

No I dont, I looked at it and not knowing what I am to put in there. I am really confused on this one. Its not giving me enough information on what it is needing, could you give me an idea of what I should do? Thanks you in advance for any help.

You have your hands tied behind your back because, as a result of using FreePBX you don’t understand what the configuration files mean and, unless you are using an obsolete version of Asterisk (which is possible if you are GUI user) you are using guidelines from the ITSP that no longer work on current versions. It would probably help if you tried to get a simple configuration going starting from the Asterisk sample configuration, rather than using the GUI, and trying to understand what each option means.

In any case, you have provided no diagnostics which would give us as clue as to where the call is specifically failing. Unfortunately diagnostic output from GUI configurations of Asterisk are rather bulky and a pain to sort through, but they are better than none at all.

You need to start with at least core set verbose 3, and, if that doesn’t help, you need to use sip set debug on.

You should also check logs for errors during startup. There will be errors if you are using a supported version of Asterisk with that configuration.

As I don’t think the audio you are getting back is part of Asterisk. It may be part of FreePBX.

It seems that you have actually posted to a FreePBX forum (freepbx.org/forum/freepbx/in … ng-message) with some very relevant diagnostics. You are getting no reply to critical packet errors, which generally indicates, firewall or routing problems (e.g. NAT errors), or misconfigured addresses. The fragment of sip.conf that you have provide does not include any NAT related parameters (and I don’t mean nat=…).

It looks like your first packet isn’t reaching them, or their reply isn’t reaching you. As such, it is probably not worth running the SIP debug, but, in general, with critical packet errors, you need to do that to find out which one is missing.

I would say you are either missing NAT options for Asterisk (localnet, extenhost/externaddr/stunaddr) or the problem lies outside Asterisk.

Yes, I am reaching out where I can. I do a goggle search on Vonage and FreePBX, what ever I find I try, plus I also try to make a posting to pick minds because I am really wanting to get this to work.

I am using FreePBX because as I was doing research on what to try, I found that FreePBX would be the easiest to understand and set up. Well I guess I was wrong.

This is all new to me, but I am picking up some ideas, and trying what ever is suggested. I had just thought that Freepbx would be just as easy to set up as 3CX. I was wrong.

As you well know I know nothing about Astrisks or FreePBX, I am a new bee and just learning, the same way as I did with anything, you have to have a starting point. Maybe I have bit of more then what I can chew. I just hope not, I have been able to get the complete system working all but outgoing calls. So I feel I am making some progress. From many posting I have read, its been 50-50 that it will work or not.

I have posted on any community boards that goggle has sent me to, asking for suggestions. I was told to set sip set debug on, from another and will be doing that next. Do I do this in astrisks, by logging into astrisks?

Could you explain “core set verbose 3” a little more, I will also research it while I am waiting to hear back from you.

I have kept a watch on everything loading when I boot up, and I do not see any warnings or errors. Its hard being new to something, but everyone was at one time, so I try to keep that in mind when I get frustrated with not understanding what is being told to me. I really do not want to go back to 3CX. Was using the free version, and you dont get paging or fax. This is what is the best in FreePBX. Running a small, really nonprofit company its hard to get money to spend for a paid version and with Vongage it really saves us money.
We help people who cant normally get a decent home to live in, one. With grants, community help, working with landlords and what ever it take to keep people from living under the bridge. Right now was a bad time to start something new because it really a busy time, because of the bad weather we have had and with winter coming fast. So I not being able to put as much time into it as I had hoped. So thanks for all your help.

I also wanted to ask if maybe you might suggest I try another VOIP company? I hate to switch, its so timely with trying to keep the same phone numbers. But open for any suggestions.

GUIs are only easy when they work first time. If they don’t work, the details you need to fix the problem have been hidden from you.

I strongly suspect you need to set localnet and externip. Look at the sample configuration files, that come with the Asterisk source code, for more details.

However, the problem could also be in your router configuration.

Thanks so much, I believe I bit off to much. I found a few different programs that can be used with windows, and using 3cx for about 3 years now, I believe I might be able to get one of them working.

I did find some know windows programs that is free and also have the page feature. This is one of the main thing we was needing. The fax feature was the second. The new release 3CX put out killed a bunch of features for the free version. Like groups, we use that a lot when we are all out of the office. Another fetcher the cut was to call all, we could put a list of numbers in and it would call all the number until some one would answers.

This was a great features too. We just cant afford to put out $395.00 right now. Most of our workers, work as volunteers. What money we do make helps with the utilities and if we find a landlord that just will not cut the deposit, or maybe not cut it enough, it comes out of our budget. I have one person that spend most of the day applying for grants, that there is few of. So asking for money for a phone system would be wrong, when it could help someone get into a home before the holidays. We will yell across the office until we get something different.

I love this program if we could only get out going calls to work. That is my only hang up, but I am not understanding it, so I need to stop wasting your time and mine.

Thank you all, and hope you have some great Holidays,
Daniel Bonnell

If I dont find something else to try, I might see you guys again after the New Years. Some thing to look forward to. LOL