Asterisk Forum

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?

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Listed below are links to sites that reference Asterisk Forum:

Asterisk Forum TrackBack URL : http://blog.tmcnet.com/mt/mt-tb.cgi/1074

9 Comments

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

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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.

Need help in conecting Asterisk with Nextone in sip. ?

Hi All

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

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

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

A new Asterisk Motherboard chipset compatibility site.

http://www.asteriskmotherboards.com

Hi Ether,

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

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