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How to call my Trixbox through my mobile phone and pick up a line

Posted on 2010-08-20
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Last Modified: 2012-05-10
Hi, we have a Trixbox at our office. Connected to it we hae a GSM Mobile line. We would like to be able to call from our cellphones to the GSM line connected to the trixbox and be able to pick up a SIP Trunk to dial to any country at very low prices or even dial through our office flat rate plan.

Logically I gues this should work somehing like this:

CALL TO MY OFFICE FROM MY CELL PHONE > OFFICE ANSWERS WITH IVR > I DIAL A CODE > AND I CAN DIAL THE DESTINATION NUMBER I NEED.

Thanks a lot for your help.
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Question by:Surferride
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Expert Comment

by:BNG4EVR
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What you are describing is normally referred to as a "Calll Out" feature.  I am not sure with trixbox CE, but with trixbox Pro you would -

   1.  Call into your voicemail box
   2. Press 3 for advanced options
   3. Press 4 for voicemail callout
   4. Follow the system prompts to dial the number you wish to call
   5. Your outgoing Caller ID will match your company's Global Caller ID so it appears that you are calling the person from your office.


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by:Surferride
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Thanks for the details. I am trying to do this with Trixbox CE. Anyone knows if this is possible? Thanks!
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Assisted Solution

by:jfaubiontx
jfaubiontx earned 500 total points
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You want the DISA feature. Go to the Module Admin and install the DISA module. Once done go to the DISA section under Internal Options. Click add DISA. Enter a name and the code you want to dial to access the DISA.

Now that you have a DISA now we need to add a way to access it. This can be done easily with a inbound route using a DID or an option off an IVR. In your case you probably want to use the former with a CID of your cell phone. This way you can call into the number and with your CID match you will be sent to the DISA. Upon reaching the DISA just follow the prompts.


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by:Surferride
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Hi jfaubiontx, thanks for your comments, this is a good approach to what I need. The problem is that I cant see DISA on the module´s list so I can install it and check how to configure it. Do you know where I can find it?
I am using Trixbox CE 2.6.2.3.

Thanks!
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by:Surferride
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OK I managed to fund the DISA function, but I cant make it work. I am trying to use the WebInterface to make it work.
I´ve been reading online documentation and it says It has to be configured through the IVR. The problem is that I cant find a way to configure an extension to DISA so It can be configured on the dialrule. Any ideas? Thanks!
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jfaubiontx earned 500 total points
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All of the setup instructions I've given assume your using the web GUI. With a DISA you don't need to associate as an extension. In the DISA you have an option to set the context. It defaults to from-internal which basically assumes you want to dial using the same rules as a phone on the local network.

As I mentioned before you can use an option off the IVR or you can configure an inbound route. If you already use an IVR, add an option to select the DISA. Don't add an announcement for it though unless you want people trying to phreak it.

If you don't already have an IVR and/or you have an inbound route for your GSM interface, copy that inbound route with a CID of your cell phone. As an example, go to Inbound Routes and add a new route. Set the Description to "to-DISA". Set the DID Number to that of your GSM trunk. Set the CID Number to your cell phone number. In the Set Destination section select the DISA that you built. Submit it and click the Apply changes bar. Now when you dial into your GSM trunk with your cell phone, you should get a prompt that says, "Please enter your password followed by the pound sign." This is where you enter the PIN number that you specified when you added the DISA. Once done you will get dialtone and you can dial out as if you were on a phone on the local network.

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by:Surferride
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Thanks a lot I will give it a try and I ´ll let you know. I think I should be close enought with your help. Thanks again.
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Expert Comment

by:jfaubiontx
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Let me know if you have any issues. We have setup DISA lines for several customers so I have many examples to use.
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by:Surferride
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Mmm.´I´ve tried what you told me but when I call to the CellPhone I get the IVR Answering.
I need to be able to have the 5 standard lines to answer to the IVR, and also de GSM line as people at the company use the lines to communicate with people at the office when they are on thre street. (They do this for free because of the free intergroup cell plan).
So basically, I need all functions to remain as they are right now, but with the chance to be able to use DISA also, not only through the GSM cell line, but also from the standard lines. The idea is that on the Asterisk I have ActionVoip provider configured so people who have the correct DISA password can dial outside. We will be changing this password frecuently to avoid phreaking, I guess this security is tight, what do you think?. Sorry for my basic knowledge on Asterisk, thanks for explaining everything for a rookie.
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by:Surferride
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OK, We managed to make it work with some of your tips and Google search. The problem now is that after we call the main number and go through DISA, we try to dial out and it the call is not being heard. Some audio and a lot of failure. Any ideas? Thanks!
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by:jfaubiontx
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Sorry I missed you question on Wednesday. As long as you use a decent pin length and change it frequently you should be fine.

The no audio sounds like maybe a NAT issue. I assume you can dial out and talk from a local phone but the call has no audio after a DISA dial out, correct? You mention, "Some audio and a lot of failure" does this mean that some call have audio? What UDP ports are open on the router?
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by:Surferride
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No problem. There shouldnt be any NAT problems when using DISA as there are no problems on any other call routing. There are even some extensions configured outside the network without problems. I dont know what could the problem be that there are problems when generating the communication or the communication seems to get established correctly but the speakerphone is muted for some reason. Thanks again for your help.
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by:Surferride
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Hi, wanted to let you know that I´ve tried again and the call can be generated.

I called from my home phone > To the Office > Dialed a Code > Dialed the DISA password > Call to my Cell Phone.

The call is getting routed without problems. The problem now is that the sound is so bad that is useless, I hear all the voice cuted off, and ten a big permanent noise. I dont know what can be wrong. Any ideas?

Thanks!
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Expert Comment

by:jfaubiontx
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What codec are you using for the PSTN and the GSM trunks? Is the box having to do a conversion? Can you provide a the log of a test call?
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by:Surferride
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Hi, this is the log I can see:

1.         2010-08-29 13:44:32        Local/0154...                      0(MYCELLPHONE NUNMBER DIALED THROUGH DISA)        ANSWERED        00:04
2.        2010-08-29 13:44:03       Zap/1-1...                   1       ANSWERED       00:29

How can I send you the full log of the whole call router capturing it through Putty?

Where do I see the CODEC I am using?

Thanks!
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by:jfaubiontx
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Yes using Putty is probably the easiest. Start asterisk with "asterisk -vvvr" and then make a call. You can then scroll up and select the text with the mouse. Then paste it into here.

As for the codec, look at the trunks in the web interface. In the peer details you should see a allow=XXXX  line. Where XXXX is the codec.
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by:Surferride
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Here is the Log file on the TXT attached. I am sorry but I am unable to get to know which codec is my Asterisk using.

I am attaching the Trunks screen and a "show translations" screen for what I could find on Google. Maybe this info works?

Thanks!
Log-Asterisk.txt
Asterisk-Trixbox-2623.jpg
ShowTranslations.jpg
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by:jfaubiontx
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The allow= line would be in your SIP trunk. Your zap will always be ulaw or alaw depending on the zone set in the /etc/zaptel.conf. However from your call log it appears that you call came in and went out on the zap which I assume is your GSM gateway. Therefore transcoding is probably not the problem. At this point I would be suspicious of the GSM gateway configuration. Which GSM gateway are you using?
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by:Surferride
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Today we received a call from the customer telling that they hear a lot of audio problems when one side talks the other mic kind of mute´s. We used to have a generic card, now we have a Digum card but I dont know if we have to change any drivers or something. It started working the same without changing anything...maybe this may help to better understand what might be wrong?

Zaptel.conf seems to have generic information, nothing seems to be relevant. I already asked the customer to send me the model of the GSM gateway. But because of the user complaints there might be a sound problem or driver problem? Is there a way to check this easily?

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by:jfaubiontx
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Yep sounds like there is a difference in the config between the GSM gateway and the zaptel card. With the model of the GSM gateway we should be able to get you the proper config for the card.
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by:Surferride
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While the customer sends me this information. Is there a simple way to check if Asterisk is correctly configured with the Digium AEX2400 card which has hardware echo cancelation?

Thanks!
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by:Surferride
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Sorry regarding to DISA, I´ve been making some more tests last night and I found that somehow the asterisk didnt realized when I hanged up so my home line which I used for testing got "sucked" by the asterisk. It wasnt until I stoped the asterisk service until I could have my home line back for user. Thanks!
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by:Surferride
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This is the GSM gateway model Hal Tel HT-1900 GSM. Now they tell me they cant even take a line with it. Thanks!
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by:jfaubiontx
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Trixbox has a command to setup the PSTN card. You can use setup-pstn which will configure the /etc/zaptel.conf file. You will also need to use the genzaptelconf program to build the channels. Use "genzaptelconf -d -s -M -F" from the command line. This usually does a pretty good job but it isn't perfect so the config may still need some tweaking once your done.

If the line isn't dropping you may still have an issue with you config. Make sure the correct signaling is used for your location. I am not familiar with the Hal Tel box. What do you mean by "they can't even take a line with it"?
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by:Surferride
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They used to dial a cell phone (in Argentina the dial plan would be: 0 to tell Asterisk is an outside line, 15 is the national identifer that the destination is a cell phone), the dial plan was allowing that all calls that started with 015 would be routed directly to the Hal Tel Box, now it doesnt work either. :(

I think there is a big mess with th Asterisk since we took away the ATCOM card and now has the DIGIUM card. We would probably reinstall Trixbox to the latest version. What would you recommend?
THanks!
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by:jfaubiontx
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Wiping out the box is a drastic step to fix the issues. While it will likely work, I think you might be better served to have someone get in there to see if the issues can be fixed. The routing problem should be in the Outbound Routes. If not it should be a simple matter to add a new route for this. The ATCOM interfaces are pin for pin compatible with the Digium. So I really don't think this is the issue but perhaps the config files are not right. Again this should be fairly easy to check. Send a private email to john@ivsystems.com and we can discuss it further.
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by:Surferride
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The help and patience was really good. The server had other problems which where preventing DISA to work correctly, but jfbaubiontx really helped a lot. Thanks a lot for your help.
I found a local partner who solved me the drivers issue I was facing between the incompatibility of the Asterisk version and the Digium AEX2400 card.
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