Hi all,
I followed the instruction http://wiki.linuxmce.org/index.php/How_to_properly_add_support_for_SIP_providers (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
"","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
Get a freepbx configuration for it. It will be easier to create a create_amp_provider.pl from it.
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
Take a look at the #freepbx irc channel or forums.... for sure you will find help.
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.
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.
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
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 (http://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
@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
@Aviator, thank you for the tip I will compare them
also curious to see Pointman87s working settings
BR Stefan
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
I made a new create_amp and providers list and submitted a ticket for it for future users.
All the best /Daniel
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
@ladekribs: In my setup my DID number is my actual phonenumber.
BR Daniel
@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
Ok i thought you also used affinity telecom, a tip is trying the sipgate template or download mine from tracker, search affinity. I didn´t get the broadcom template to work either, same problem.
BR Daniel
@pointman87: I didn't get your script for affinity to work. It didn't successfully register to the sip server. I don't have much time to troubleshoot for the moment so I tried the sipgate template instead. I got the same result as you. Incoming calls work but not outgoing. I tried experimenting with the outbound dial patterns in the Free PBX settings without success. I also noted there are two sets of dial patterns; one under Basic->Outbound routes-sipgate->Dial patterns and another under Basic->Trunk SIP/Sipgate->Dial rules. They seem unrelated and I don't know the difference. Which one did you change, and what rules did you use?
regards
Easiest way is to use x for rules. in my case in gävle, we have 026 + 6 numbers = xxxxxxxxx
Do you understand? You just use the number of x apposite to the numbers you want to be able to dial.
BR Daniel
the syntax wasn't my problem. everything worked fine once i saw the "apply changes" button on top of the page :)
I also managed to connect the linksys rtp300 to asterisk. When I get the time I will make a wiki page about it.
For those who have a linksys RTP300 collecting dust in their drawer, here is how to integrate it in LinuxMCE:
http://wiki.linuxmce.org/index.php/Linksys_rtp300
regards