Probleme mit Asterisk beim rauswählen

ululu

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Hi @ all!

Bin schon eine weile am rumbasteln, schaffe es aber nicht das das ich über sipgate rauswählen kann. Intern geht alles einwandfrei und ich kann auch über die sipgatenummer rein wählen. Nur eben raus geht leider nicht. Ich poste mal die configs und das debug.

sip.conf
Code:
[general]
context=default			; Default context for incoming calls
port=5060			; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0		; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes			; Enable DNS SRV lookups on outbound calls
				; Note: Asterisk only uses the first host 
				; in SRV records
				; Disabling DNS SRV lookups disables the 
				; ability to place SIP calls based on domain 
				; names to some other SIP users on the Internet
				
language=de			; Default language setting for all users/peers
externip=xxxxx.mine.nu	; Address that we're going to put in outbound SIP messages
localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks

;qualify=no
disable=all
allow=alaw
allow=ulaw
allow=g729
allow=gsm
allow=slinear
canreinvite=yes 

register => SipID:[email protected]/SipID


[SipID]
type=peer
username=SipID
secret=SipPW
host=sipgate.at
fromuser=SipID
fromdomain=sipgate.at
insecure=very
canreinvite=no
nat=no
;disallow=all
;allow=ulaw
context=abgehend

[sipgate_at_in]
type=peer
fromdomain=sipgate.at
host=sipgate.at
context=ankommend

[101]
type=friend
username=101
secret=101
callerid="101" <101>
host=dynamic
nat=no                       
canreinvite=no                
;disallow=all
;allow=gsm                    
allow=ulaw
;allow=alaw
context=home
dtmfmode=rfc2833

[102]
type=friend
username=102
secret=102
callerid="102" <102>
host=dynamic
nat=no                      
canreinvite=no                
;disallow=all
;allow=gsm                   
allow=ulaw
;allow=alaw
context=home
dtmfmode=rfc2833
extensions.conf
Code:
[general]
static=yes
writeprotect=no

[globals]

[default]

;exten => s,1,dial(SIP/101&SIP/102,8,t)

include => home
include => ankommend
include => abgehend

[home]
exten => 101,1,Dial(SIP/101,30)
exten => 101,2,Hangup

exten => 102,1,Dial(SIP/102,30)
exten => 102,3,Hangup


[ankommend]
exten => SipID,1,Dial(SIP/101)

[abgehend]
exten => _0.,1,Dial,SIP/${EXTEN}@SipID|45|r
Und das passiert wenn ich vom Phone eine Nummer anrufe:

Code:
Sip read:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.201:5060;branch=z9hG4bK08e4f82f
From: "102" <sip:[email protected]>;tag=0003e3a51752002113a72296-17905334
To: <sip:[email protected]>;tag=as4547effc
Call-ID: [email protected]
CSeq: 102 ACK
Content-Length: 0


Urgent handler
7 headers, 0 lines
Urgent handler
Destroying call '[email protected]'
Urgent handler


Sip read:

Urgent handler
1 headers, 0 lines
Urgent handler


Sip read:

Urgent handler
1 headers, 0 lines
Urgent handler


Sip read:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.201:5060;branch=z9hG4bK1372b47d
From: "102" <sip:[email protected]>;tag=0003e3a5175200221209b25f-72bd2e97
To: <sip:[email protected]>
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 INVITE
User-Agent: Cisco-CP7960G/8.0
Contact: <sip:[email protected]:5060;transport=udp>
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Remote-Party-ID: "102" <sip:[email protected]>;party=calling;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,norefersub
Content-Length: 275
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 25461 0 IN IP4 192.168.0.201
s=SIP Call
t=0 0
m=audio 22466 RTP/AVP 0 8 18 101
c=IN IP4 192.168.0.201
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/0
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

Urgent handler
17 headers, 13 lines
Urgent handler
Using latest request as basis request
Urgent handler
Sending to 192.168.0.201 : 5060 (non-NAT)
Urgent handler
Reliably Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.0.201:5060;branch=z9hG4bK1372b47d
From: "102" <sip:[email protected]>;tag=0003e3a5175200221209b25f-72bd2e97
To: <sip:[email protected]>;tag=as0cfc1e13
Call-ID: [email protected]
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[email protected]>
Proxy-Authenticate: Digest realm="asterisk", nonce="67a6e27d"
Content-Length: 0


 to 192.168.0.201:5060
Urgent handler
Scheduling destruction of call '[email protected]' in 15000 ms
Urgent handler
Found user '102'
Urgent handler


Sip read:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.201:5060;branch=z9hG4bK1372b47d
From: "102" <sip:[email protected]>;tag=0003e3a5175200221209b25f-72bd2e97
To: <sip:[email protected]>;tag=as0cfc1e13
Call-ID: [email protected]
CSeq: 101 ACK
Content-Length: 0


Urgent handler
7 headers, 0 lines
Urgent handler


Sip read:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.201:5060;branch=z9hG4bK620a60c2
From: "102" <sip:[email protected]>;tag=0003e3a5175200221209b25f-72bd2e97
To: <sip:[email protected]>
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 102 INVITE
User-Agent: Cisco-CP7960G/8.0
Contact: <sip:[email protected]:5060;transport=udp>
Proxy-Authorization: Digest username="102",realm="asterisk",uri="sip:[email protected]",response="ccc3b065ffc966a04bbcfa522f262e8f",nonce="67a6e27d",algorithm=md5
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Remote-Party-ID: "102" <sip:[email protected]>;party=calling;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,norefersub
Content-Length: 275
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 25461 0 IN IP4 192.168.0.201
s=SIP Call
t=0 0
m=audio 22466 RTP/AVP 0 8 18 101
c=IN IP4 192.168.0.201
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/0
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

Urgent handler
18 headers, 13 lines
Urgent handler
Using latest request as basis request
Urgent handler
Sending to 192.168.0.201 : 5060 (non-NAT)
Urgent handler
Found user '102'
Urgent handler
Found RTP audio format 0
Urgent handler
Found RTP audio format 8
Urgent handler
Found RTP audio format 18
Urgent handler
Found RTP audio format 101
Urgent handler
Peer audio RTP is at port 192.168.0.201:22466
Urgent handler
Found description format PCMU
Urgent handler
Found description format PCMA
Urgent handler
Found description format G729
Urgent handler
Found description format telephone-event
Urgent handler
Capabilities: us - 0x8014e (gsm|ulaw|alaw|slin|g729|h263), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729)
Urgent handler
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Urgent handler
Looking for TELEFONNUMMER in home
Urgent handler
Reliably Transmitting (no NAT):
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.201:5060;branch=z9hG4bK620a60c2
From: "102" <sip:[email protected]>;tag=0003e3a5175200221209b25f-72bd2e97
To: <sip:[email protected]>;tag=as0cfc1e13
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[email protected]>
Content-Length: 0


 to 192.168.0.201:5060
Urgent handler


Sip read:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.201:5060;branch=z9hG4bK620a60c2
From: "102" <sip:[email protected]>;tag=0003e3a5175200221209b25f-72bd2e97
To: <sip:[email protected]>;tag=as0cfc1e13
Call-ID: [email protected]
CSeq: 102 ACK
Content-Length: 0


Urgent handler
7 headers, 0 lines
Urgent handler
Destroying call '[email protected]'
Urgent handler


Sip read:

Urgent handler
1 headers, 0 lines
Urgent handler
Es ist sicherlich kein gewaltiger Fehler aber ich sehe ihn nicht mehr :) Danke im Vorraus

LG
 

stefanwillmerot

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Mein Verdacht: Deine Telefone liegen im Kontext "home". Von dort können sie aber nicht rauswählen weil der Dial-Befehl im Kontext "abgehend" nicht erreichbar ist. Einige Fehlermeldungen würdest Du wahrscheinlich auf der Asterisk-Konsole sehen können.

Pack den Dial-Befehl doch mal ins "home".

Grüße
Stefan
 

ululu

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stefanwillmerot schrieb:
Mein Verdacht: Deine Telefone liegen im Kontext "home". Von dort können sie aber nicht rauswählen weil der Dial-Befehl im Kontext "abgehend" nicht erreichbar ist. Einige Fehlermeldungen würdest Du wahrscheinlich auf der Asterisk-Konsole sehen können.

Pack den Dial-Befehl doch mal ins "home".

Grüße
Stefan

Hey Cool :) Das war es! Danke!

LG
 

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