Budgetphone met Asterisk aan de praat krijgen

[torrie666]

Beste Medetweakers,

Helaas ben ik meer een hardwareman dan een softwareman. Het gevolg; ik snap het allemaal wel, maar ben zelf een beetje snel de weg kwijt.

Ik heb nu een Asterisk PBX geinstalleerd en ik heb de volgende pporten geforward naar mijn PBX vanuit mijn router: (Mijn PBX heeft als IP 192.169.10.20)

5004-5082 (UDP / TCP)
10001-20000 (UDP / TCP)
4569 (UDP / TCP)

Nu ben ik via een aantal tutorials en youtube tutorial behoorlijk op weg gekomen, maar ik loop gewoon vast.

Ik heb 2 extenties aangemaakt: 200 en 300. Intern bellen is geen probleem dit werkt prima.

De ZAP Trunk heb ik verwijderd (Dit is voor PSTN als ik het goed heb...)

Ik heb een SIP Trunk aangemaakt;

Dial Rules:
0XXXXXXXX
0XXXXXXXXXX
31XXXXXXXX
31XXXXXXXXXX

Peer Details:
context=from-pstn
fromdomain=budgetphone.nl
fromuser=31XXXXXXXXX
host=budgetphone.nl&dynamic
insecure=invite,port
secret=XXXXXXXX
type=peer
user=31XXXXXXXXX
username=31XXXXXXXXX

User Details
[leeg]

Register String:
[e-mail]:password:[e-mail]/31XXXXXXXXX


Ook heb ik een Ring Group aangemaakt:

Ringall, dus alles moet afgaan als er iets binnenkomt (Ext 200 en 300)

De inbound Route staat ingesteld op de ringgroup "ringall"

Nu kan ik dus niet naar buiten bellen of gebeld worden. Wel zie ik dat als ik bel naar mijn vaste nummer (van Budgetphone) dat er e.e.a. gebeurd in de CLI:

(Wie kan mij een duw in de goede richting geven of verder helpen?)


<------------>
-- Executing [31107142784@from-sip-external:1] NoOp("SIP/83.143.188.183-0a0d2678", "Received incoming SIP connection from unknown peer to 31107142784") in new stack
-- Executing [31107142784@from-sip-external:2] Set("SIP/83.143.188.183-0a0d2678", "DID=31107142784") in new stack
-- Executing [31107142784@from-sip-external:3] Goto("SIP/83.143.188.183-0a0d2678", "s|1") in new stack
-- Goto (from-sip-external,s,1)
-- Executing [s@from-sip-external:1] GotoIf("SIP/83.143.188.183-0a0d2678", "0?from-trunk|31107142784|1") in new stack
-- Executing [s@from-sip-external:2] Set("SIP/83.143.188.183-0a0d2678", "TIMEOUT(absolute)=15") in new stack
-- Channel will hangup at 2012-01-15 21:34:15 UTC.
-- Executing [s@from-sip-external:3] Answer("SIP/83.143.188.183-0a0d2678", "") in new stack
Audio is at 192.168.10.20 port 13884
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
pbx*CLI>
<--- Reliably Transmitting (no NAT) to 83.143.188.165:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 83.143.188.165;branch=z9hG4bK3c91.43b940b5.0;received=83.143.188.165
Via: SIP/2.0/UDP 83.143.188.183:5060;received=83.143.188.183;branch=z9hG4bK0dba45ff;rport=5060
Record-Route: <sip:83.143.188.165;lr=on;ftag=as6f471567>
From: "06XXXXXXXX" <sip:06XXXXXXXX@83.143.188.183>;tag=as6f471567
To: <sip:[e-mail]>;tag=as3bb92cf1
Call-ID: 1cef0ddd39f7acc40ad185e70d82b35a@83.143.188.183
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:31107142784@192.168.10.20>
Content-Type: application/sdp
Content-Length: 264

v=0
o=root 2911 2911 IN IP4 192.168.10.20
s=session
c=IN IP4 192.168.10.20
t=0 0
m=audio 13884 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
-- Executing [s@from-sip-external:4] Wait("SIP/83.143.188.183-0a0d2678", "2") in new stack
pbx*CLI>
<--- SIP read from 83.143.188.165:5060 --->
ACK sip:31107142784@178.85.212.91:5060 SIP/2.0
Via: SIP/2.0/UDP 83.143.188.165;branch=z9hG4bK3c91.43b940b5.2
Via: SIP/2.0/UDP 83.143.188.183:5060;received=83.143.188.183;branch=z9hG4bK64c9ff3c;rport=5060
Max-Forwards: 12
From: "06XXXXXXXXX" <sip:06XXXXXXXX@83.143.188.183>;tag=as6f471567
To: <sip:[e-mail]>;tag=as3bb92cf1
Contact: <sip:06XXXXXXXX@83.143.188.183>
Call-ID: 1cef0ddd39f7acc40ad185e70d82b35a@83.143.188.183
CSeq: 102 ACK
User-Agent: GW02
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
-- Executing [s@from-sip-external:5] Playback("SIP/83.143.188.183-0a0d2678", "ss-noservice") in new stack
-- <SIP/83.143.188.183-0a0d2678> Playing 'ss-noservice' (language 'en')
-- Executing [s@from-sip-external:6] PlayTones("SIP/83.143.188.183-0a0d2678", "congestion") in new stack
-- Executing [s@from-sip-external:7] Congestion("SIP/83.143.188.183-0a0d2678", "5") in new stack
pbx*CLI>
<--- SIP read from 83.143.188.165:5060 --->
OPTIONS sip:178.85.212.91:5060 SIP/2.0
Via: SIP/2.0/UDP 83.143.188.165:5060;branch=0
From: sip:[e-mail];tag=348a5314
To: sip:178.85.212.91:5060
Call-ID: ad9fdbd1-670428e8-768e44@83.143.188.165
CSeq: 1 OPTIONS
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
Looking for s in from-sip-external (domain 178.85.212.91)
pbx*CLI>
<--- Transmitting (no NAT) to 83.143.188.165:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 83.143.188.165:5060;branch=0;received=83.143.188.165
From: sip:[e-mail];tag=348a5314
To: sip:178.85.212.91:5060;tag=as411dc216
Call-ID: ad9fdbd1-670428e8-768e44@83.143.188.165
CSeq: 1 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:192.168.10.20>
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'ad9fdbd1-670428e8-768e44@83.143.188.165' in 32000 ms (Method: OPTIONS)
Really destroying SIP dialog 'ad9fdbd1-89a228e8-948e44@83.143.188.165' Method: OPTIONS
pbx*CLI>
<--- SIP read from 83.143.188.165:5060 --->
BYE sip:31107142784@178.85.212.91:5060 SIP/2.0
Via: SIP/2.0/UDP 83.143.188.165;branch=z9hG4bK4c91.087c341.0
Via: SIP/2.0/UDP 83.143.188.183:5060;received=83.143.188.183;branch=z9hG4bK0573b4ee;rport=5060
Max-Forwards: 12
From: "06XXXXXXXX" <sip:062XXXXXXXX@83.143.188.183>;tag=as6f471567
To: <sip:[e-mail]>;tag=as3bb92cf1
Call-ID: 1cef0ddd39f7acc40ad185e70d82b35a@83.143.188.183
CSeq: 103 BYE
User-Agent: GW02
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---
Sending to 83.143.188.165 : 5060 (no NAT)

<--- Transmitting (no NAT) to 83.143.188.165:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 83.143.188.165;branch=z9hG4bK4c91.087c341.0;received=83.143.188.165
Via: SIP/2.0/UDP 83.143.188.183:5060;received=83.143.188.183;branch=z9hG4bK0573b4ee;rport=5060
From: "06XXXXXXXX" <sip:06XXXXXXXX@83.143.188.183>;tag=as6f471567
To: <sip:[e-mail]>;tag=as3bb92cf1
Call-ID: 1cef0ddd39f7acc40ad185e70d82b35a@83.143.188.183
CSeq: 103 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:31107142784@192.168.10.20>
Content-Length: 0


<------------>
== Spawn extension (from-sip-external, s, 7) exited non-zero on 'SIP/83.143.188.183-0a0d2678'
-- Executing [h@from-sip-external:1] NoOp("SIP/83.143.188.183-0a0d2678", "Hangup") in new stack
-- Executing [h@from-sip-external:2] Set("SIP/83.143.188.183-0a0d2678", "DID=s") in new stack
-- Executing [h@from-sip-external:3] Goto("SIP/83.143.188.183-0a0d2678", "s|1") in new stack
-- Goto (from-sip-external,s,1)
-- Executing [s@from-sip-external:1] GotoIf("SIP/83.143.188.183-0a0d2678", "0?from-trunk|s|1") in new stack
-- Executing [s@from-sip-external:2] Set("SIP/83.143.188.183-0a0d2678", "TIMEOUT(absolute)=15") in new stack
-- Channel will hangup at 2012-01-15 21:34:27 UTC.
-- Executing [s@from-sip-external:3] Answer("SIP/83.143.188.183-0a0d2678", "") in new stack
== Spawn extension (from-sip-external, s, 3) exited non-zero on 'SIP/83.143.188.183-0a0d2678'
Really destroying SIP dialog '1cef0ddd39f7acc40ad185e70d82b35a@83.143.188.183' Method: BYE
pbx*CLI>
<--- SIP read from 192.168.10.10:62636 --->