Start working on Verboice-raspberry pi

[Ashish Gupta]

Hi there,

I have installed verboice on Raspberry pi and it is up and running as a service. But what is the next step to move further in verboice. How do I create the call flows and everything without using the internet. And if not then how will I configure it to my server.

Thanks,

Ashish Gupta

[Ashish Gupta]

Please help as my all the work is stopped here.

[Gustavo Giraldez]

Hi Ashish,

If Verboice is installed correctly in the Raspberry pi, then you should be able to connect to the web interface with your browser by navigating to the IP address of the Raspberry pi at port 3000, for example http://192.168.1.42:3000 of course using the correct IP address for your Raspberry pi. From the web interface, you should create an account and the proceed to create projects and workflows, just like the online version.

HTH,

Gustavo

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[Ashish Gupta]

Hi Gustavo,

Thanks for your reply. We solve the problem by installing apache2 web server which was missing and then navigating through the IP address. 

Now, we need to configure the GSM 3G dongle with Verboice on Raspberry Pi(using chan_dongle) to receive call via GSM network in offline mode. For that we have some queries regarding your Project Verboice on Local Servers.

1) When working on Local server what is the path or location of the call flows created from the verboice designer?

2) How the Verboice “custom channel” works. What is the entry for the “Dial String” option?

3) How can I configure any Verboice channel with “chan_dongle” to receive call offline via GSM network?

Regards,

Ashish

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B.Tech 3rd Year

The LNM Institute of Information Technology, Jaipur

[Ashish Gupta]

Hi there,

Any ideas about it. It would be a great help.

Regards,

Ashish

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[Gustavo Giraldez]

Hi Ashish,

1) Call flows are executed by Verboice through the AGI (application gateway interface) provided by Asterisk. So, to answer your question, call flows are stored in the MySQL database.

2) Custom channels allow you to use Verboice with manually configured Asterisk channels. The dial string option is a prefix string that Verboice sends to Asterisk when placing calls, which allows you to select different channels, by manually modifying the extensions.conf (or extensions.ael). This is needed to make outgoing calls through Verboice. To receive incoming calls, you have to manually add entries to extensions.conf to redirect incoming calls through your channels to the 'verboice' Asterisk context.

3) Verboice doesn't provide UI to configure channels other than SIP channels. For other types of channels you need to configure them manually and hook them up through extensions.conf as explained in 2). The other type of channel managed by Verboice is Twilio, which doesn't use Asterisk at all.

HTH,

Gustavo

[Ashish Gupta]

Hi Gustavo,

When using Asterisk with chan_dongle(without Verboice) the sample entries for entension.conf that I made was-

[default]
exten => _X.,1,Wait(2)
exten => _X.,2,Answer()
exten => _X.,3,Wait(1)
exten => _X.,n,AGI(/usr/local/Feedback/Feedback.agi)   <==== Location of the agi file.
exten => _X.,n,Hangup()

But here with Verboice where should I redirect the Asterisk. In the default Verboice extension.ael file it is given at "agi://localhost:19000". When I redirect there the call just rings for 1 second and  then hangs up automatically.

So to receive incoming calls from chan_dongle where should I redirect it.

Also, when I go to http://localhost:19000 it redirects to a page showing-

EXEC   RING

Can you please share a sample extension.conf to receive the incoming calls for this case. Also, do you think we need to change anything in sip.conf in this particular case.

Thanks,

Ashish Gupta

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[Ashish Gupta]

Hi Gustavo,

Sorry to bother you again but I'm running on a tight deadline. So is it possible to have a quick Skype call at any convenient time. I guess I'm stuck at this point and it won't take long.

Thanks,

Ashish Gupta 

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[Ashish Gupta]

Hi Gustavo,

As per you advice I added manual entries in the "extensions.conf" to redirect the calls from chan_dongle to verboice. I changed the context in the configuration file of chan_dongle to 'verboice'. Now I am able to redirect the incoming calls from chan_dongle to 'verboice' context in extensions.conf  but not able to play the call flows at the location "agi://localhost:19000". 

Below are the screenshots of my extensions.conf and the verbose of an incoming call. For debugging I added the sample prompt before and after the calling of agi.

Also, there is no calls in my verboice call dashboard.

Thanks,

Ashish Gupta

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[Kakada Chheang]

Hi Ashish,

Please modify your extensions.conf as follows:

[verboice]

exten => 1234, n, AGI(agi://localhost:19000)

As verboice will create a new call session in this thread so it might be conflicted that session was already created by asterisk dialplan before it send the call to Verboice application

HTH,

Kakada Chheang

Kakada Chheang

Software Developer, InSTEDD iLab Southeast Asia

http://www.ilabsoutheastasia.org

Office: +855 23 996750

Mobile: +855 69 860 012

Skype:  kakada.chheang

email: kak...@instedd.org

twitter: @iLabSEA

facebook: www.facebook.com/iLabSEA

[Ashish Gupta]

Hi Kakada,

I modified the extensions.conf. Now the calls are just ringing for 1 sec and then it hangs up. 

I am not able to redirect them to the call flows. I still not understand how the call flows are located at "agi://localhost:19000". Just to clarify we are using GSM network to receive calls with chan_dongle. Now how do we have to make channels in the Verboice UI and Do we have to make any changes in the sip.conf file?

Thanks,

Ashish Gupta  

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[Kakada Chheang]

Hi Ashish,

You have to set up a custom channel in Verboice then assign a call flow to it, then modifies your asterisk dialplan as follows:

[verboice]

exten => 1234, n, AGI(agi://localhost:19000,,channel_id) replace channel_id with the real one that you get from Verboice.

channel_id: you can get it by edit Verboice channel then it will show channel_id in the url(e.g: http://verboice-server/channels/12, so 12 is the channel_id), remember if channel was removed so you have to modify your asterisk dialplan too in order to make work anymore.

Note: please see the call of Verboice in verboice_project_dir/broker/log/*.log, it might help you to identifies the issues.

HTH,

Kakada Chheang

[Ashish Gupta]

Hi Kakada,

It seems that to create a custom channel we need to have a Dial String entry. Can you please suggest what should be the entry for this.

Thanks,

Ashish Gupta

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[Kakada Chheang]

I think, for Dial String, Gustavo already explained well in this thread but if you do not need to send outgoing call from Verboice to caller so you can keep it blank.

[Ashish Gupta]

Hi Kakada,

Thank you for your advise. Now, the incoming calls are answered but no call flow is played. The call is answered and then after a 1-2 wait it hangs up. But I'm able to get the call logs of the incoming calls in the Verboice UI. I have checked the call flow that has to be played, no mistakes there. 

I am attaching the screenshot of both call logs and the verbose.

Thanks,

Ashish Gupta

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[Ashish Gupta]

Hi Kakada,

As seen in the verbose, asterisk ends with 'Auto fallthrough status UNKNOWN'. But it always gives a call log in the dashboard and is not able to play the call flow. Just to clarify I changed the Asterisk version that was pre-installed i.e, from Asterisk1.8 to Asterisk11. It was due to chan_dongle compatibility issues. Will it make any difference? 

Thanks,

Ashish Gupta 

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[Gustavo Giraldez]

Hi Ashish,

From the screenshots you posted, the call is getting through to Verboice. Please check the details of the failed call logs. Hopefully, it should provide more information on what exactly is failing. Also, as Kakada suggested, check the Verboice broker log at verboice_project_dir/broker/log/ as it may provide more information.

As for the Dial String option in Verboice custom channel, it's specific to the Asterisk channel you are using. In your case, I think it should be Dongle/dongle0/{number} . Verboice will replace {number} with the number to call before sending the string to Asterisk. Again, this is only needed for outgoing calls though.

Asterisk11 should work fine.

Also, I clarification because I made a mistake before. I told you to change the context of the dongle channel to verboice. While it works in this case, it's not right and will break if you try to use Verboice with SIP channels. Verboice configures the SIP channels to also use the 'verboice' context. Since you changed the AGI command to tell Verboice the channel ID, it will think that all incoming calls are from the Dongle. The correct way to set this up is to use another context for the dongle, say 'verboice-dongle', and modify the AGI command there, while leaving the 'verboice' context as it was.

HTH,

Gustavo

[Ashish Gupta]

Hi Gustavo,

For now we need to receive the calls from the the dongle only but anyway I will change the context. 

I am unable to see the details of the incoming calls from the call dashboard. It is giving me the error all the time(screenshot below ). Also, I didn't much much of the details in the /broker/logs/ directory. These files doesn't change when I receive a call but I have attached them too.

I also encountered a website(link below) telling that with Adhearsion we have to use Asterisk 1.8. 

http://adhearsion.com/docs/getting-started/asterisk  

Not sure about the correct version of Asterisk to be installed.

Thanks, 

Ashish Gupta

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[Ashish Gupta]

Hi Kakada & Gustavo,

I have changed the context 'verboice' to 'verboice-dongle' in the dongle configuration. But still there is no call flow. Also, there is no console.log file in my /broker/log directory which contains the logs of calls and all.

Any ideas what could be the problem.

Thanks,

Ashish Gupta

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[Ashish Gupta]

Hi Kakada & Gustavo,

Not sure, but is the syntax for extension.conf seems correct to you all. When I change it to this-

[verboice-dongle]

exten => _., 1, AGI(agi://localhost:19000,,4)             ( Here '4' is the channel_id )

I got this verbose in the Asterisk(screenshot attached). In that '12345' is the "exten" that I have given in my dongle configuration. For reference I am also attaching my dongle configuration file (dongle.conf) here.

Thanks,

Ashish Gupta

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[Ashish Gupta]

Hi Kakada & Gustavo,

I found the broker log in the /var/log/verboice/ directory and it says this when receiving the new incoming call.

11:03:30.209 [info] [<0.12206.0>|0c7d5dd6] session: Session ("58dc4c4c-f059-4725-9b1d-bb64a9051495") answer

11:03:30.317 [info] [<0.12208.0>|e32452fb] session: Start

11:03:30.836 [warning] [<0.12208.0>|eedb0e76] session: The user hang up

11:03:30.838 [info] [<0.12206.0>|0c7d5dd6] session: Session ("58dc4c4c-f059-4725-9b1d-bb64a9051495") terminated normally

My extensions.conf is

[verboice-dongle]

exten => 12345,1,AGI(agi://localhost:19000,,4)

I don't know why it hang up automatically. 

Thanks,

Ashish Gupta

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[Ashish Gupta]

Hi Kakada & Gustavo,

From the past posts on Verboice Group it seems that Verboice uses RTP to transfer the media or the call flows on SIP channels and RTP is used to transfer media on IP network only. But we are not working on an IP network. We are receiving calls from a Dongle via GSM network.

Do you think this can be the issue for abrupt call hang up without call flows.

Thanks,

Ashish Gupta

 

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[Ashish Gupta]

Hi there,

Any ideas about the above issue. I am really stuck at this last thing to deploy Verboice on Raspberry Pi.

Thanks,

Ashish Gupta

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[Gustavo Giraldez]

Hi Ashish,

Verboice delegates to Asterisk for transfering media, so it uses RTP for SIP calls but it'll use other methods depending on the underlying channel. That shouldn't be a problem.

Please add a line for the hangup context to the dialplan context, otherwise Verboice is invoked twice by call: once on the call start, then again on hangup. Try:

[verboice-dongle]

exten => h,1,NoOp()

exten => _.,1,AGI(agi://localhost:19000,,4)

I'm not sure it will fix anything though.

Keep an eye on /var/log/verboice/broker.log. If the failure is happening from the Verboice side, you'll see error messages there. Also check the call log entries from the UI.

Other questions: what does your call flow look like? Also, what version of Verboice are you using? Are you using the image linked from the wiki at https://github.com/instedd/verboice/wiki/Installing-on-Raspberry-PI? Or did you install your own? When you installed the updated Asterisk, did you completely overwrite the configuration files? Verboice needs access to the AMI interface and the corresponding credentials need to be configured in Asterisk. Check the manager.conf file. Do outgoing calls work? Configure the dial string if you haven't done so, and try scheduling a call from the UI.

Regards,

Gustavo

[Ashish Gupta]

Hi Gustavo,

I an using the provided verboice-raspbian image for my project. Now with the pre-installed asterisk 1.8.13.1 I was not able to compile the chan_dongle. So I installed the asterisk 11 and then installed chan_dongle for it. I overwrite the configuration files present at /etc/asterisk/ directory and followed the commands given at https://github.com/instedd/verboice/wiki/Installing to setup Asterisk. What else do I need to change in the Asterisk to configure it with Verboice.

Also, as I mentioned earlier is there any issue related to compatibility of Adhearsion in this case.

Thanks,

Ashish Gupta

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[Gustavo Giraldez]

Hi Ashish,

If you followed the instructions in the wiki it should be enough.

What are you planning to do with Adhearsion? I don't think you'll have any problems, but I ask just in case.

Did you try make an outgoing call? Also, if you can give the Raspberry pi internet access, can you try configuring a SIP channel to see if it works. Callcentric channels are the easiest to configure in my experience.

HTH,

Gustavo

[Ashish Gupta]

Hi Gustavo,

As per your advice I tested a callcentric channel with Raspberry pi and it was working perfectly fine with it. But there is a little issue with it. Whenever I click on the 'View Details' in the call dashboard Verboice UI it always gives me this error(screenshot below).

Do you have any ideas how to resolve this issue? The same was also there when I am performing with custom channel and Dongle.

Thanks 

Ashish Gupta

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[Ashish Gupta]

Hi Gustavo,

After a lot of attempts I started with a new fresh image and created a new call flow. I noticed that the play template in Verboice Designer is stored as *.gsm file in the directory "/usr/share/asterisk/sounds/verboice/" but when I installed the asterisk 11 there was no such "verboice" directory in the sounds folder. I think that can be a problem. Now how can we to configure verboice to store sound files in that directory again after upgradation to asterisk 11. 

Thanks,

Ashish Gupta

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[Gustavo Giraldez]

Hi Ashish,

To debug the error happening when accessing a call log details, please find the log file /var/log/verboice/production.log (or web.log). That's the log file for the UI and should indicate why that page is failing.

It's possible that the problem is the missing folder in /usr/share/asterisk/sounds, but I'd expect an error to show up in the broker.log file. In any case, you can recreate the folder and give the verboice user full permissions on it so it can store the voice files.

HTH,

Gustavo

[Ashish Gupta]

Hi Gustavo,

I followed the production.log while accessing the call log details and found this error-

[2015-12-21 16:36:41.486]  INFO  Started GET "/calls/4" for 192.168.1.100 at 2015-12-21 16:36:41 +0000

[2015-12-21 16:36:41.152]  INFO  Processing by CallLogsController#show as HTML

[2015-12-21 16:36:41.154]  INFO    Parameters: {"id"=>"4"}

[2015-12-21 16:36:41.324]  INFO  Completed 500 Internal Server Error in 165.7ms

[2015-12-21 16:36:41.428] FATAL  Errno::EAFNOSUPPORT (Address family not supported by protocol - socket(2)): app/controllers/call_logs_controller.rb:33:in `show'

Any idea what this error means. 

Thanks,

Ashish Gupta

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[Ashish Gupta]

Hi Gustavo & Kakada,

Now, I think the problem is with Asterisk 11 and Verboice configuration. When I started with a new image with Asterisk 1.8.13.1 then it worked for the callcentric channel. But when I update it with Asterisk 11 and installing the chan_dongle then it is not working for the callcentric channel also. I also did not executed "$sudo make samples" this time so the files of the /etc/asterisk/ directory remains same. But the call could not connect this time. I call see there is no green box in front of the callcentric channel this time which was coming before with Asterisk 1.8.13.1.

Also to mention that in previous case the asterisk 1.8.13.1 keep reloading the sip files(screenshot below) but this not the case in Asterisk 11. It would be better if you could guide me to install chan_dongle with the previous asterisk that came preinstalled. 

Please help me as soon as possible.

Thanks,

Ashish Gupta

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[Ashish Gupta]

Hi Gustavo & Kakada,

I just want to mention that to install chan_dongle it needs to be compiled with the asterisk. But here in the image you provided there is no compiled asterisk source. I managed to download the exact same version of asterisk for the chan_dongle from the link

https://packages.debian.org/wheezy/asterisk

 but chan_dongle did not compiled with it. Do you have any idea how to compile it with the asterisk preinstalled or with the one I downloaded without changing the configuration files.

Thanks,

Ashish Gupta 

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[Kakada Chheang]

Hi Ashish,

I never tried with chan_dongle but I used to faced some problem too on the process to setted up with chan_dahdi Asterisk channel with E1, it's very difficult to find out the document that let me known the lastest library package dependencies(wanpipe) doesn't support with the Sangoma A102DE card so I have to downgrade it in order to made it up and ran with Asterisk, in this case it might be same as what I used to face but I'm not sure just give you an idea, or did you try to contact your supplier for more information.

Best,

[Ashish Gupta]

Hi Kakada,

I tried the chan_dongle with my Huawei dongle without Verboice and it was running perfectly fine. But the problem here is how can I install chan_dongle with the preinstalled asterisk version. Otherwise, please give me the list of changes that I have to make to configure the Verboice with Asterisk 11(after Upgradation).

Thanks, 

Ashish Gupta

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[Gustavo Giraldez]

Hi Ashish,

For Verboice to work properly with Asterisk it needs the following:

• AMI interface enabled and a user 'verboice' with password 'verboice' and full permissions. This is usually configured in the file manager.conf. A sample for this can be found in https://github.com/instedd/verboice/blob/master/etc/asterisk/manager.conf. You can check in the asterisk console that the user exists:

*CLI> manager show user verboice

          username: verboice

            secret: <Set>

               ACL: no

         read perm: system,call,log,verbose,command,agent,user,config,dtmf,reporting,cdr,dialplan,originate,agi,cc,aoc,test,security,message,all

        write perm: system,call,log,verbose,command,agent,user,config,dtmf,reporting,cdr,dialplan,originate,agi,cc,aoc,test,security,message,all

   displayconnects: yes

allowmultiplelogin: yes

         Variables: 

Also, check that the management interface is enabled with:

*CLI> manager show settings

Global Settings:

----------------

  Manager (AMI):             Yes            

  Web Manager (AMI/HTTP):    No             

  TCP Bindaddress:           0.0.0.0:5038   

  HTTP Timeout (minutes):    60             

  TLS Enable:                No             

  TLS Bindaddress:           Disabled       

  TLS Certfile:              asterisk.pem   

  TLS Privatekey:                           

  TLS Cipher:                               

  Allow multiple login:      Yes            

  Display connects:          Yes            

  Timestamp events:          No             

  Channel vars:                             

  Debug:                     No       

• A default 'verboice' context for managed SIP channels which invokes the broker through AGI. Sample here: https://github.com/instedd/verboice/blob/master/etc/asterisk/extensions.ael (this an .ael file, a similar configuration can be achieved in extensions.conf). Check this with

*CLI> dialplan show

...

[ Context 'verboice' created by 'pbx_ael' ]

  'h' =>            1. NoOp()                                     [pbx_ael]

  '_.' =>           1. AGI(agi://localhost:19000,,)               [pbx_ael]

...

• Verboice managed SIP configuration files included in sip.conf, and the managed files writable by the broker. The files are sip_verboice_channels.conf and sip_verboice_registrations.conf. Sample sip.conf here: https://github.com/instedd/verboice/blob/master/etc/asterisk/sip.conf. You can list configuration files with:

*CLI> config list

...

chan_sip             /etc/asterisk/sip.conf                  

chan_sip             /etc/asterisk/sip_notify.conf           

chan_sip             /etc/asterisk/sip_verboice_channels.conf

chan_sip             /etc/asterisk/sip_verboice_registrations.conf

...

• You need to have a directory for Verboice to place the audio files. This is usually /var/lib/asterisk/sounds but can be changed in {verboice_project_dir}/broker/verboice.conf. Verboice places its files in a subdirectory 'verboice' inside this directory (eg. /var/lib/asterisk/sounds/verboice). You can check what directory Asterisk is using by running:

verboice*CLI> core show settings

...

* Directories

  -------------

  Configuration file:          

  Configuration directory:     /etc/asterisk

  Module directory:            /usr/lib/asterisk/modules

  Spool directory:             /var/spool/asterisk

  Log directory:               /var/log/asterisk

  Run/Sockets directory:       /var/run/asterisk

  PID file:                    /var/run/asterisk/asterisk.pid

  VarLib directory:            /var/lib/asterisk

  Data directory:              /var/lib/asterisk

  ASTDB:                       /var/lib/asterisk/astdb

  IAX2 Keys directory:         /var/lib/asterisk/keys

  AGI Scripts directory:       /var/lib/asterisk/agi-bin

The relevant line is the "Data directory".

Up to here are the default and minimal settings Verboice needs.

To configure a custom channel (using chan_dongle in your case), you also need:

• The channel configured in Asterisk properly. This varies with the type of channel, but you need to point it to a context which invokes the broker through AGI, with the channel id of the Verboice custom channel. For example:

dongle.conf

...

[defaults]

context=verboice-dongle

...

extensions.conf

[verboice-dongle]

exten => h,n,NoOp()

exten => _X.,n,AGI(agi://localhost:19000,,CHANNEL_ID)

Replacing CHANNEL_ID with the id of the custom channel created in Verboice.

• To make outgoing calls from Verboice, you need to provide the dial string in the custom channel configuration that Asterisk can use to select the appropriate channel. For example: Dongle/dongle0/{number} . Verboice will replace {number} with the actual number to dial.

If these configuration options are correctly set, it doesn't matter much which version of Asterisk you are using. We've successfully run Verboice with Asterisk 1.8, 11 and 13.

HTH,

Gustavo

[Ashish Gupta]

Hi there,

After a lot of attempts I was not able to play the call flows via custom channel through my Dongle. I was thinking, is there any way to access the call flows directly made by the Verboice through extensions.conf file without making any channel just like we access any '.agi' file using its location. Also can you suggest me some alternative to VoiceBlue GSM Gateway supported by Verboice. Can we built it with another Raspberry pi and Huawei Dongle.

Regards,

Ashish Gupta

[Ashish Gupta]

Hi Everyone,

Does anyone has any ideas for the above. Please help.

Regards,

Ashish Gupta

[Ashish Gupta]

Hi everyone,

I tested a new thing. I connected the RPi with verboice installed on a different port and then tried to connect to it with a different sever machine with my Dongle with it. I provided the following in the extensions.conf

[verboice]

exten => _X.,1,Wait(1)

exten => _X.,n,Answer()

exten => _X.,n,AGI(agi://172.22.47.103:19000,,2)       <==== IP of the Verboice RPi

exten => _X.,n,Hangup()

But when I try to call it gives the warning and does not plays the call flow. I can hear a long Ringing tone and the call disconnects itself. I tried it with both SIP and CUSTOM channel in the Verboice. I have attached the screenshot of the asterisk verbose.

Regards,

Ashish Gupta

[Gustavo Giraldez]

Hi Ashish,

It seems Asterisk is forwarding the call to Verboice through AGI. What do the Verboice broker logs show? Is there a call log entry created that you can see from the Verboice UI?

To answer to your previous question, no, there's no place where you can get the call flows, as they are dynamically executed through the AGI interface.

To clarify, is Verboice still working fine with Callcentric channels inside the Raspberry Pi?

Regards,

Gustavo

[Ashish Gupta]

Hi Gustavo,

Yes, I am getting the call log entry in the Verboice UI, but when I select the 'View Details' it shows me the error as I explained before also. 

The Callcentric channel is working fine inside the Raspberry Pi.

I do not understand how I am able to connect the Verboice RPi by providing its IP Address in extensions.conf file.

Is there any changes I need to make from the Verboice RPi side. Till now I just created a SIP and CUSTOM channel and tried with the channel ID of both of them. Do I need to edit the extensions.conf of the verboice UI.

Regards,

Ashish Gupta

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