StanaPhone Trunk Setup. Here are the entries for the Stanaphone SIP trunk. Create a SIP trunk for the service with a Trunk Name of voipdiscount. VoipDiscount doesn't support an outbound CallerID number so leave it blank. Here's the complete list of functions that will work out of the box from any extension on your TrixBox system: 611. Getting started with FreePBX – Part 1 Setting up a trunk. Give the trunk a name in the “Trunk Name” box. Password, secret, setup, sip, trunk. Previous post Trixbox, Elastix and Asterisk videos Next post Getting started with FreePBX – Part 2 Setting up an extension. Related Articles.
Active1 year, 1 month ago
I set up a SIP TRUNK in FreePBX/Asterisk that works perfectly for incoming calls. Here' s the relevant configuration:
However, whenever I try to place an outgoing call (through the same trunk) I have a 'all lines busy' signal from asterisk. If I enable SIP DEBUG this is what I get (apparently my call is being rejected due to an invalid alias at the other side, which I can't control since it's my VOIP provider):
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Any ideas of what might be wrong on my side of things?
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If I connect a simple softphone to my VOIP provider, it works flawlessly (incoming and outgoing calls).
Hi Thanks for the reply. My Setup: I'm using asterisk callback to call back the caller. Later, on the callback, the call is routed to the a2billing context, by which the caller is authenticated using his/her caller id using a2billing. Then click Add SIP Trunk. After you complete the entries for each provider. This auto-upgrade stuff was a pain to set up but now that it’s working it’s really cool!' Tuesday, July 11, 2006 at 6:39 am. Or some way of telling the box to dial a number and callback to connect both sides. Is this possible? Nov 11, 2008 SIP Trunk to Trix Box SIP Trunk to Trix Box 3Sixty (TechnicalUser) (OP) 7 Nov 08 09:54. Just set up an extension on trix of which Avaya registers (IP auth is always better), set the codecs on Trix to G711A or U (depending where you are, U is North America, Canada and parts of Asia), or G729, and set the DTMF to RFC2833. How to configure FreePBX FreePBX is an easy to use GUI (graphical user interface) that controls and manages Asterisk, the world's most popular open source telephony engine software. To configure FreePBX with the mydivert.com trunk you will need the following information.
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My guess is that your Caller ID is offending them. Are you setting to anything other than your actual assigned DID?
Based on:
MichelV69MichelV69
This is a know bug from Asterisk when you use the non standard sip port 5060 for your server. The bug is discussed here https://issues.asterisk.org/jira/browse/ASTERISK-24767 .
You should be able to correct this using fromdomain=prepago.com.py:5060 but Asterisk ignores the directive port and rewrite the from as From: 'sip:[email protected]:5061'.You can patch Asterisk code and recompile it or use the standard sip port in your server.
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Active4 years, 11 months ago
Is anyone out there running predictive dialers or Call Center type products on Asterisk/Trixbox. I'm looking for vicidial type functionality. The best way to characterize it would be political campaigning or marketing surveys.
Also, I really am just trolling for good or bad news either way on the product.
As I don't really have a problem per se, this might need to be marked Community Wiki.
Thanks for the help
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2 Answers
We use Vicidial for as much as 220 seats at a time with a multi-server Asterisk setup. It performs very well. We have in-house programmers that do the customization of the CRM, etc. for us so it's also very cost-effective.
You could contact a consultant from their webpage:
user36579user36579
We looked into this about 18 months ago. We ended up going with a different provider (not Asterisk) but from memory the choices were mostly fairly poor, especially for highly customised environments. And their interfaces were all clearly an afterthought designed by someone with only basic HTML skills.
In the end we went with a VoIP provider called Swyx, and then wrote our own predictive dialling code using their extensive and well documented ActiveX API (integrated it into the Do Not Call register, automatic callback requests when calls failed or went to answering machine, etc).
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protected by Michael Hampton♦Sep 20 '14 at 9:49
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