Author Topic: Problems with create_amp_*.pl  (Read 2391 times)

willow3

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Problems with create_amp_*.pl
« on: December 08, 2010, 05:18:14 pm »
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

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Re: Problems with create_amp_*.pl
« Reply #1 on: December 09, 2010, 12:04:37 am »
Get a freepbx configuration for it. It will be easier to create a create_amp_provider.pl from it.

willow3

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Re: Problems with create_amp_*.pl
« Reply #2 on: December 13, 2010, 07:54:18 pm »
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

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Re: Problems with create_amp_*.pl
« Reply #3 on: December 13, 2010, 08:44:34 pm »
Take  a look at the #freepbx irc channel or forums.... for sure you will find help.

pointman87

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Re: Problems with create_amp_*.pl
« Reply #4 on: January 09, 2011, 03:03:47 am »
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.
« Last Edit: January 09, 2011, 03:25:09 am by pointman87 »

willow3

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Re: Problems with create_amp_*.pl
« Reply #5 on: January 09, 2011, 09:48:08 pm »
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

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Re: Problems with create_amp_*.pl
« Reply #6 on: January 10, 2011, 10:59:14 am »
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

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Re: Problems with create_amp_*.pl
« Reply #7 on: January 13, 2011, 03:49:28 pm »
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

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Re: Problems with create_amp_*.pl
« Reply #8 on: January 13, 2011, 10:28:27 pm »
@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

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Re: Problems with create_amp_*.pl
« Reply #9 on: January 14, 2011, 09:09:04 am »
@Aviator, thank you for the tip I will compare them

also curious to see Pointman87s working settings

BR Stefan

pointman87

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Re: Problems with create_amp_*.pl
« Reply #10 on: January 14, 2011, 05:49:27 pm »
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

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Re: Problems with create_amp_*.pl
« Reply #11 on: January 16, 2011, 11:52:32 pm »
I made a new create_amp and providers list and submitted a ticket for it for future users.

All the best /Daniel

ladekribs

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Re: Problems with create_amp_*.pl
« Reply #12 on: January 22, 2011, 08:05:52 pm »
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

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Re: Problems with create_amp_*.pl
« Reply #13 on: January 22, 2011, 11:13:33 pm »
@ladekribs: In my setup my DID number is my actual phonenumber.

BR Daniel

ladekribs

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Re: Problems with create_amp_*.pl
« Reply #14 on: January 23, 2011, 11:53:43 am »
@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