Testing Asterisk SIP and DAHDI local calls

1
votes

I am a real beginner in asterisk, so please tolerate my question :)

I tried to configure asterisk for realtime and it is working fine for local sip calls. Now, I am trying to make the following test with dahdi calls:

I connected an analog phone to an FXS channel of my Digium card and tried to call this phone (exten 124) from a sip softphone (X-lite).

I get the following error:

-- Executing [124@from-sip:1] Dial("SIP/2000-00000004", "SIP/124")

[May 31 10:24:22] WARNING[5457]: chan_sip.c:5667 create_addr: Purely numeric hostname (124), and not a peer--rejecting

my extensions.conf:

[from-sip]
switch =>Realtime
[from-pstn]
exten => 124,1,Dial(DAHDI/3)

It seems that the dial is done using from-sip context not from-pstn context as required.

Anyone to advise or correct my understanding?

Thanks million

Zak

 

 



1
asterisk

1 Answers

0
votes

In Asterisk realtime and not realtime you can configure where to send calls from particular extension, this should be configured in "context"(for realtime check context column), so I believe in your case it is "from-sip". This means all calls from that extension will hit this context, you can't send one call from same extension to one context and other to another, all calls will hit "from-sip" context.