Hi,
I’ve just got started with Asterisk.
I’ll give some background, Im a computer programmer, Linux enthusiast, and general geek.
I’ve just got my asterisk box setup, certainly a steep learning curve but after a few hours play got my it talking to IAX2 and SIP service providers with multiple extensions and voicemail.
I like this.
Im using 1.4.0-beta3, and Im really confused on how the different contexts key together (in terms of features.conf for parked calls, sip.conf, extensions.conf and voicemail.conf), there’s a lot of different sections, a lot of commands, and the extension language is out of this world (I love it!)
But first things first. Parking calls.
OK I can type from extension 5001, #700 and I get told “701”.
Fine.
Trouble is, when I type 701 on the phone to retrieve a call, I get a 404 status.
I’ve done a debug, and here’s the (what I think is) relevent bit (I have more if needed):
[code]Looking for 701 in default (domain 10.0.0.10)
<— Reliably Transmitting (no NAT) to 10.0.0.79:5060 —>
SIP/2.0 404 Not Found[/code]
extensions.conf contain:
[code][general]
static=yes
writeprotect=no
clearglobalvars=no
[globals]
CONSOLE=Console/dsp ; Console interface for demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
;IAXINFO=guest ; IAXtel username/password
;IAXINFO=myuser:mypass
;TRUNK=Zap/g2 ; Trunk interface
[default]
include => parkedcalls
; extensions
exten => 5001,1,Dial(SIP/simon.dean.grandstream,10,rtT)
exten => 5001,2,Voicemail(u5001)
exten => 5002,1,Dial(SIP/simon.dean.xlite,10,rtT)
exten => 5002,2,Voicemail(u5002)
; outgoing sipgate
exten => _9.,1,Dial(SIP/${EXTEN:1}@sipgate-out,30,rtT)
exten => _9.,2,Playback(invalid)
exten => _9.,3,Hangup
; voicemail extensions
exten => 4500,1,VoicemailMain
exten => 4501,1,VoiceMailMain(${CALLERID(num)})
; sipgate incoming
exten => 12345678,1,NoOp(— ${CALLERID} calling on sipgate (${EXTEN}) —)
exten => 12345678,2,Dial(SIP/simon.dean.xlite&SIP/simon.dean.grandstream,10)
exten => 12345678,3,Answer
exten => 12345678,3,Wait,1
exten => 12345678,4,Voicemail(u5001)
exten => 12345678,5,hangup
; voiptalk outgoing
exten => _0[1-9].,1,Dial(IAX2/12345678@voiptalk/44${EXTEN:1})
exten => _00.,1,Dial(IAX2/12345678voiptalk/${EXTEN:2})
; voiptalk incoming
exten => 0871xxxxxxx,1,Dial(SIP/simon.dean.xlite&SIP/simon.dean.grandstream,10)
exten => 0871xxxxxxx,2,Answer
exten => 0871xxxxxxx,3,Wait,1
exten => 0871xxxxxxx,4,Voicemail(u5001)
exten => 0871xxxxxxx,5,hangup
[/code]
If you need any more info, let me know.
Im not quite sure how the rtT flags are supposed to work on the extensions, well, I do, but Im not sure where they’re supposed to be used… ie, on the extensions themselves, or on the incoming line?
And Im not sure why I can’t internally transfer say by typing #5002… I just get an english sounding person tell me “sorry, this number is not yet in service” or something.
After I’ve typed the two I get:
* DTMF-relay event received: 2
<--- Transmitting (no NAT) to 10.0.0.79:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.79;branch=z9hG4bKf64f06bda4c3626b;received=10.0.0.79
From: <sip:simon.dean.grandstream@10.0.0.79>;tag=730b5819c0e22459
To: "07771xxxxxx" <sip:07771xxxxxx@10.0.0.10>;tag=as51c67530
Call-ID: 2665e86e2ff781a75a6d00c05aedb7f4@10.0.0.10
CSeq: 19192 INFO
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:07771xxxxxx@10.0.0.10>
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '2665e86e2ff781a75a6d00c05aedb7f4@10.0.0.10' in 32000 ms (Method: INFO)
set_destination: Parsing <sip:simon.dean.grandstream@10.0.0.79> for address/port to send to
set_destination: set destination to 10.0.0.79, port 5060
Reliably Transmitting (no NAT) to 10.0.0.79:5060:
BYE sip:simon.dean.grandstream@10.0.0.79 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.10:5060;branch=z9hG4bK21cf7fc6;rport
From: "07771xxxxxx" <sip:07771xxxxxx@10.0.0.10>;tag=as51c67530
To: <sip:simon.dean.grandstream@10.0.0.79>;tag=730b5819c0e22459
Contact: <sip:07771xxxxxx@10.0.0.10>
Call-ID: 2665e86e2ff781a75a6d00c05aedb7f4@10.0.0.10
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
neil*CLI> sip debug
<--- SIP read from 10.0.0.79:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.10:5060;branch=z9hG4bK21cf7fc6;rport
From: "07771xxxxxx" <sip:07771xxxxxx@10.0.0.10>;tag=as51c67530
To: <sip:simon.dean.grandstream@10.0.0.79>;tag=730b5819c0e22459
Call-ID: 2665e86e2ff781a75a6d00c05aedb7f4@10.0.0.10
CSeq: 103 BYE
User-Agent: Grandstream BT100 1.0.4.55
Contact: <sip:simon.dean.grandstream@10.0.0.79>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0Hope you guys can lend me some answers.
Thanks
Simon
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