Problems with create_amp_*.pl

[willow3]

Hi all,

I followed the instruction http://wiki.linuxmce.org/index.php/How_to_properly_add_support_for_SIP_providers
to add support for my Swedish SIP provider (www.affinity.se). The tech support claims they have customers running their service with asterisk. When opening up the phone line in the web admin it says "Registered <date> <hour>" under status. My guess is that this means it successfully connected to the host, and the host accepted the credentials.

Now the problem is that neither outgoing nor incoming calls work. I don't know anything about asterisk so I'm quite lost what I have messed up. The instruction was very simple and straightforward though... The only magic number I found was  $DECLARED_PREFIX = "9". It seems to be set to 9 for most, but not all, available providers. I don't know what it is, or if it is important. I tried "9" and "", same result.

When I placed a test call from my cell phone I noted the following entry in /var/log/asterisk/cdr-csv/Master.csv

Code: [Select]

"","NNNNNNNNNN","s","from-pstn","""NNNNNNNNNN"" <NNNNNNNNNN>","SIP/XXXXXXXXX-b54972b0","","SayAlpha","","2010-12-07 21:46:34","2010-12-07 21:46:34","2010-12-07 21:4
6:45",11,11,"ANSWERED","DOCUMENTATION","asterisk-1291758394.6",""
Where NNN is my cell phone number and XXX is my phone number assigned by SIP provider.

Does this say anything meaningful? When I place the call a voice says "The number you have dialed is not in service. Please check the number and try again". Are there any other relevant logs that could give me a hint about whats going on?

When I try to place an outgoing call, I get the message "Call dropped. Reason: Normal clearing".

Any help is appreciated!

regards

[pw44]

Get a freepbx configuration for it. It will be easier to create a create_amp_provider.pl from it.

[willow3]

That is a good idea. Unfortunately, tech support wasn't up to the challenge. They didn't even know what free pbx is. I am afraid I am on my own here.

Is there any more documentation about create_amp that I could read to understand more?

regards

[pw44]

Take  a look at the #freepbx irc channel or forums.... for sure you will find help.

[pointman87]

Hi Willow3.

I also use affinity in sweden. I tried the sipgate template, that allow you to set username, passwd, host and number.
My incoming calls work perfectly but i have problems calling out. Maybe we can put our minds together?

Mvh  Daniel, Gävle

Edit: got it working with outgoing calls, FreePBX, outbound routes, modify dial patterns with the right prefixes.

[willow3]

Thanks for your input. I will try the sipgate template when I get some time. I'll let you know if I have any progress.

[ladekribs]

Hi,

I used Broadcom template to create a digisp file but I also get the message "The number you have dialed is not in service. Please check the number and try again" when calling in.
maybe I should try the sipgate template?

I can dial out ok just need to dial 9 before the phonenumber.

BR Stefan

[Aviator]

ladekribs

I had the exact same issue when setting up my provider, voipgo. I had to email their support staff to get a basic asterisk sip.conf example, which still had to be modified a bit. The changes I made to the create_amp_*.pl are attached tohttp://svn.linuxmce.org/trac.cgi/ticket/942.  Maybe your provider can help you by providing some basic asterisk configuration that you can use in creating a template?

Regards, Michael

[willow3]

@pointman87: Since you got it to work with the same provider as I have, maybe you could post your create_amp file in this thread?

regards

[ladekribs]

@Aviator, thank you for the tip I will compare them

also curious to see Pointman87s working settings

BR Stefan

[pointman87]

You dont have to make an create_amp_*.pl. All you need to do is, go to webadmin, phone lines and choose sipgate (try for free, pay as you go) template. Then you supply your phonenumber, sip server, username and passwd. Easy as that.

BR Daniel

[pointman87]

I made a new create_amp and providers list and submitted a ticket for it for future users.

All the best /Daniel

[ladekribs]

if I go to advanced - configuration - phones setup inbound routes, and clear the DID number then inbound calls works

we should not change the settings in freepbx manually so is there som other way to change the DID settings?


BR Stefan

[pointman87]

@ladekribs: In my setup my DID number is my actual phonenumber.

BR Daniel

[ladekribs]

@Daniel yes so was mine, and that resultet in the "The number you have dialed is not in service"
so I removed it and now it works
i am using digisip, now converted to bredband2

BR Stefan