Comments for Asterisk Forum

Asterisk Forum

2005-01-17T22:16:11Z

The Asterisk forums have been pretty active lately on TMC's website. Someone posted asking for help getting SIP clients to register on his Asterisk server. If you think you can lend him a helping hand, go check out the Asterisk forums. Looks like someone deleted that post, but I found another interesting Asterisk post: How to Connect SPA-3000 to Asterisk so Asterisk will answer?

How to Connect SPA-3000 to Asterisk so Asterisk will answer?

Comment from Guilherme on 2005-02-03

2005-02-03T17:37:46Z

Hi! I´m having problem on trying to configure my asterisk server at the sip.conf. I woul d like to know what should I put on the general? I´m using a no NAT option. I'll be greatful.

Guilherme Mendonça Friere

Comment from Redfone Communications on 2005-02-24

2005-02-24T20:35:12Z

Redfone Communications LLC announces foneBRIDGE, the Quad T1 to ethernet bridge. Links Asterisk software with and a commodity server, cleans up wiriing, simplifies installations, and provides for redundant servers.

More information at http://www.red-fone.com/pdf/fonebridge.pdf. Special show pricing available til 25 Feb, see order form at http://www.red-fone.com/pdf/fonebridge_of.pdf.

Inquiries at sales@red-fone.com

Comment from anas on 2005-06-10

2005-06-10T17:06:56Z

hi,
i'd like to connect two asterisk using a long distance connection, the trunck would be a SIP trunk, i'll be gratefull for your help.

Comment from Claudio on 2005-06-24

2005-06-24T17:10:30Z

Need help in conecting Asterisk with Nextone in sip. ?

Comment from Enkhee on 2006-09-04

2006-09-04T12:12:19Z

Hi All

Asterisk,
is it possible to use over VPN? not over internet, pls tell me!

Comment from Cosmin on 2006-11-30

2006-11-30T13:01:59Z

Does somebody has any ideea how to make my asterisk to accept incoming calls from my sipphone?... please give me some help over here because i'm driving crazy... thank you

Comment from ether on 2007-01-18

2007-01-18T05:14:39Z

Hello All,

I am working on a project of making multiple calls from trixbox but now I am stuck on one place. The system is working fine with multiple calls to softphones but when I make a call to PSTN network it get some wrong status.

Basically I am making calls through PHP code using manager API. I use Originate command to make a call on selected trunk and when the call is answered it bridged the context to that trunk which plays the message to the called party and gives the option to replay the message or confirm it by pressing 1 or 2 keys.

Now what happens when I make call to PSTN network (i.e. Mobile or Landline) the system immediately show the status as answered and pass the control to the context which starts playing message immediately without waiting for the other end to pickup the call. Also if nobody picks up the call it still show the status as answered and deduct money from the users account.

I am very much confused in this matter that how I will find the right status and duration of the call using originate.

Please help me because I have not much time left to make this system working. please forgive me for my poor English.

Ether

Comment from fintech on 2007-10-04

2007-10-04T21:52:59Z

A new Asterisk Motherboard chipset compatibility site.

http://www.asteriskmotherboards.com

Comment from sunil on 2008-09-08

2008-09-08T05:40:06Z

Hi Ether,

Have you solved the pstn callstatus problem..? Plz help me,Im struggling with the same problem.

Comment from arpit modi on 2010-02-15

2010-02-15T11:12:57Z

Hello All,

Basically i am a asterisk developer, works on opensources like a2billing, vicidial, freepbx etc etc.

I want to implement a playback functionality as below :

Whenever any user call to asterisk box, i am currently playing an audio file, which works fine, but if the audio is 2 or 3 hour long like training session, and user got disconnected in between then when he call again, system starts playing that file from start.

Instead of that i want to store the duration of file which is played before, for example, the audio file is 2 hour long and user got disconnected after 1 hour, then when he call again it should start playing from the remaining 1 hour, so he will not have to listen the whole file again.

Can anybody please let me know if this can be done with playback / controlplayback or any other asterisk applications????

or is there any patch for storing the location of file when the call got disconnected?

Any kind of help will be greatly appreciated.

Looking forward to your suggestions. please help me out.....

Thanks,
Arpit

Comment from Mike on 2010-05-19

2010-05-19T09:26:31Z

I’ve been following NerdVittles for about 2 years now and I love all of the development that’s gone into the great PBX In A Flash. Today a coworker introduced me to Google Voice which was formerly GrandCentral. As good as PIAF is at user friendliness, I think Google Voice has the perfect interface for managing your phones with one phone number. I think it’d be awesome to be able to use Asterisk as a Google Voice replacement or supplement with a similar interface. I think that would lift Asterisk out of the nerd/geek/hobbyist status that most people see it in and bring it to more and more businesses (and maybe even homes!) that maybe don’t have a dedicated IT staff or just want a simple ip-PBX that can perform functions similar to Google Voice and with the same ease-of-use, but on a broader scale. Like I said above, I love PIAF and think that if Digium would stop deprecating so many commands, Asterisk could be a great platform for most business needs.
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