configuring asterisk with 2talk

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maverick

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Reply # 612571 20-Apr-2012 19:44
Yes your Asterisk box so will need the licence if you wish to have G729, cisco have already paid the fees as part of their box price, you will have problems with the wrtp as it has the same details, so unless the guys adjusted the config you will be best to factory default... off to watch some rugby now, have fun .... :) Yes I am a employee of WxC (My Profile) ... but I do have my own opinions as well

gchiu

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Reply # 612574 20-Apr-2012 19:57
I'll have to find time to reset and then reprovision the wrtp54g. I can confirm that I can have two concurrent calls. I sent one to the spa303, and one to the 3cx softphone. When I sent both to the spa303, then the second line came up and I was able to switch between the two calls.

maverick

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Reply # 612575 20-Apr-2012 20:01
Yes that is what should happen on a vfx line, you will not be able to rx a 3rd call though Yes I am a employee of WxC (My Profile) ... but I do have my own opinions as well

gchiu

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Reply # 612625 20-Apr-2012 21:59
I managed to get my googlevoice number coming in as well. This links to a free IPKall number which then routes to my asterisk box. I have to "Allow Anonymous Inbound SIP Calls" for this to work, but have set up a separate inbound route to terminate all calls that don't have a matching DID. Hopefully this will not be too much of a security hole. I want the GV number to come to my softphone. I used to use voxalot but that has dropped this service now. So, I guess what I can do is setup Hamachi on Asterisk and my notebook, and this should allow my softphone to act as a remote extension.

maverick

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Reply # 612691 21-Apr-2012 06:06
gchiu: I managed to get my googlevoice number coming in as well. This links to a free IPKall number which then routes to my asterisk box. I have to "Allow Anonymous Inbound SIP Calls" for this to work, but have set up a separate inbound route to terminate all calls that don't have a matching DID. Hopefully this will not be too much of a security hole. I want the GV number to come to my softphone. I used to use voxalot but that has dropped this service now. So, I guess what I can do is setup Hamachi on Asterisk and my notebook, and this should allow my softphone to act as a remote extension. You need to be very very very careful here, the responsibility for security of this box is yours... you really have to know what you are doing with security on Asterisk when you start opening this up I cant stress this enough, 90% of all hacks we have seen have been insecure Asterisk boxes. Yes I am a employee of WxC (My Profile) ... but I do have my own opinions as well

sbiddle

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Reply # 612699 21-Apr-2012 07:21
gchiu: I managed to get my googlevoice number coming in as well. This links to a free IPKall number which then routes to my asterisk box. I have to "Allow Anonymous Inbound SIP Calls" for this to work, but have set up a separate inbound route to terminate all calls that don't have a matching DID. Hopefully this will not be too much of a security hole. I want the GV number to come to my softphone. I used to use voxalot but that has dropped this service now. So, I guess what I can do is setup Hamachi on Asterisk and my notebook, and this should allow my softphone to act as a remote extension. Allowing this as well as allowing unrestricted SIP access to port 5060 is a security nightmare and pretty much like leaving your house door wide open. Eventually somebody will walk right in and steal all your belongings. Unless you fully instand dialplans and exactly how Asterisk works then this is a very bad idea. If you have any exploits allowing external access to a from-internal context you will get hacked, guaranteed. *Need help configuring your Linksys ATA or IP Phones for New Zealand? Check my blog post

gchiu

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Reply # 612744 21-Apr-2012 09:59
Ok, warnings accepted. I'll disable this for the moment. But isn''t what I've done documented in that comment by Joe Roper to your blog post http://www.geekzone.co.nz/sbiddle/7183 ?

maverick

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Reply # 612750 21-Apr-2012 10:18
No idea really Graham as that is a guide, the biggest problem with Asterisk and no disrespect here is that it is a powerful freeware box but some huge exploits if you do not really know what your doing around the security side of it, if you are leaving it open to external SIP connections apart from your providers IP you are basically letting any one see you have a SIP connection,you will be scanned and then attacked again and again with every known Asterisk exploit and you wont even see it, if they find it .. they can and will use your box as a proxy to send calls via your connection ... you will be liable then for the calls cost. So please be very sure of what you doing around security with this platform Final word.. .Asterisk is a powerful tool but dangerous Yes I am a employee of WxC (My Profile) ... but I do have my own opinions as well

gchiu

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Reply # 612959 21-Apr-2012 20:57
I used the wizard to upgrade from Freepbx 2.8 to 2.9 to see if that would fix the googlevoice module. But I came up sgainst other issues. I was left with an amportal.conf not writeable message. As root running amportal chown was supposed to fix this, but even though it was now owned by the asterisk user, it still came up with this message, and said as a result asterisk was going to be running in a severely degraded fashion. Other people seemed to have the same issue. In the end I decided to do more updates, and using update-programs, update-fixes and update-source I upgraded the Centos 6.2 kernel and at the end of this the error was gone. There does remain a dangling symbolic link error but it doesn't seem to be effecting things. But after this I still couldn't get the Googlevoice module working. But I found another website which detailed on how to setup a sip trunk to Ipkall. So, with anonymous sip connections turned off, I am now able to receive calls from ipkall, and therefore Googlevoice. I guess i'll need a working GV module to make calls out. From what I read, this module keeps breaking.

gchiu

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Reply # 613184 22-Apr-2012 15:48
I've written a CGI script which is run by the Apache webserver running as root. This creates a call file in the /var/spool/asterisk/tmp directory, chowns it to user asterisk, and then moves it to outgoing. It is working. Since the webserver is not exposed to the internet, I don't see any major external security risks. Do call scripts bypass dialplans or use them?

gchiu

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Reply # 613343 22-Apr-2012 21:13
Come across an unexpected problem. I installed Hamachi on the asterisk box, and PC. Now when I call from the softphone to an extension, I only get one way audio. that is sound goes from the PC softphone but not back again. I changed the Hamachi network settings on the asterix box entry to allow all traffic but it makes no change. One way audio is said to be due to NAT problems, and blocking of RTP but there is no NAT involved. Anyone seen this with Hamachi?

gchiu

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Reply # 613490 23-Apr-2012 08:57
Advised to add the VPN network in the general sip settings and that has fixed it.

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