Auflegen waehrend dem Anrufen funktioniert nicht

RTDI · 31 Mai 2005

Hi,

ich verwende derzeit folgende Asterisk Version:
Asterisk CVS-HEAD-05/31/05-16:45:12
Ich habe eine Digiumkarte und sowohl ausgehende als auch eingehende Anrufe mit DDI funktionieren wunderbar.

Jedoch habe ich ein Problem:
Wenn man jetzt ueber einen SIP Client (entweder Sipura oder Xlite) eine Nummer waehlt und vor einem Verbindungsaufbau auflegt, laeutet das Telefon trotzdem auf der Gegenstelle weiter und ich sehe auch keinen Eintrag im Asterisk log (asterisk -vvvgc), dass aufgelgt wurde
Legt der SIP Client aber waehrend eines Gespraechs auf funktioniert das wunderbar.

Die SIP Clients sind hinter einem NAT Rechner und ommen ueber einen SIP Proxy daher.

Meine zapata.conf

Code:

[channels]
busydetect=no
callprogress=no
switchtype=euroisdn
immediate=no
overlapdial=yes
pridialplan=unknown
prilocaldialplan=national
echocancel=no
echocancelwhenbridged=no
echotraining=no
signalling=pri_cpe
context=default
internationalprefix=00
nationalprefix=0
usecallingpres=yes
callprogress=no
language=de
group=1
rxgain=0.0
txgain=0.0
channel => 1-15
channel => 17-31

hier der ausschnitt aus der extensions.conf fuers rauswaehlen:

[trunk-out]
exten => _X.,1,EnumLookup(${EXTEN:3})
exten => _X.,2,Goto(1000)
exten => _X.,52,Goto(${ENUM},2000)
exten => _X.,102,Goto(2000)
exten => _X.,1000,Dial(${ENUM})
exten => _X.,1001,Goto(2000)
exten => _X.,1100,Busy
exten => _X.,2000,SetCIDNum(XXXXXX{CALLERIDNUM})
exten => _X.,2001,Macro(dialout,${EXTEN})

[macro-dialout]
exten => s,1,Dial(Zap/g1/${ARG1}|60)
exten => s,2,Congestion
exten => s,102,Busy

Ideen?

Maik · 31 Mai 2005

Kannst du mal bitte 'sip debug' machen und das hier posten?

RTDI · 1 Jun 2005

XXX.XXX.XXX.XXX = oeffentliche ip adresse des routers/NAT servers
YYY.YYY.YYY.YYY = IP Adresse von SER
ZZZ.ZZZ.ZZZ.ZZZ = IP Adresse des siproxd
PPP.PPP.PPP.PPP = Private IP des SIP Clients hinter NAT server
realm = meine domain
telefonnummer = gewaehlte telefonnummer

also das ganze laeuft ueber einen siproxd ... hier mal der log auszug davon:

Code:

Jun  1 08:02:08 proxy siproxd[3907]: proxy.c:211 INFO:Outgoing Call from: [email][email protected][/email]
Jun  1 08:02:10 proxy siproxd[3907]: proxy.c:225 INFO:Ending Call from: [email][email protected][/email]

da scheint alles normal zu sein.. wenn ich im log direkt mitschaue sehe ich dann halt auch den logeintrag sobald ich auf auflegen druecke...

aber beim sip debug in asterisk bekomme ich nach dem auflegen keinen log eintrag:

asterisk*CLI>
<-- SIP read from YYY.YYY.YYY.YYY:5060:
INVITE sip:telefonnummer@realm SIP/2.0
Record-Route: <sip:[email protected];ftag=809628580;lr=on>
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=0
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:5060;branch=z9hG4bK93c4b19e10af5ba9443f1c0135f7cdad
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:62628;rport;branch=z9hG4bKDF8A45EDC7C44A78A062E9498D719499
From: aswin <sip:1010@realm>;tag=809628580
To: <sip:telefonnummer@realm>
Call-ID: [email][email protected][/email]
CSeq: 10233 INVITE
Contact: <sip:[email protected]>
Max-forwards: 68
User-agent: X-Lite release 1103m
Content-Type: application/sdp
Content-Length:   217

v=0
o=1010 1517015 1517062 IN IP4 ZZZ.ZZZ.ZZZ.ZZZ
s=X-Lite
c=IN IP4 ZZZ.ZZZ.ZZZ.ZZZ
t=0 0
m=audio 7077 RTP/AVP 0 8 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

--- (14 headers 10 lines)---
Using INVITE request as basis request - [email][email protected][/email]
Sending to YYY.YYY.YYY.YYY : 5060 (NAT)
Found no matching peer or user for 'YYY.YYY.YYY.YYY:5060'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port ZZZ.ZZZ.ZZZ.ZZZ:7077
Found description format pcmu
Found description format pcma
Found description format telephone-event
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for telefonnummer in trunk-out
list_route: hop: <sip:[email protected];ftag=809628580;lr=on>
list_route: hop: <sip:[email protected]>
Transmitting (NAT) to YYY.YYY.YYY.YYY:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=0;received=YYY.YYY.YYY.YYY;rport=5060
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:5060;branch=z9hG4bK93c4b19e10af5ba9443f1c0135f7cdad
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:62628;branch=z9hG4bKDF8A45EDC7C44A78A062E9498D719499
From: aswin <sip:1010@realm>;tag=809628580
To: <sip:telefonnummer@realm>
Call-ID: [email][email protected][/email]
CSeq: 10233 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:[email protected]>
Content-Length: 0


---
asterisk*CLI>
<-- SIP read from YYY.YYY.YYY.YYY:5060:
INVITE sip:telefonnummer@realm SIP/2.0
Record-Route: <sip:[email protected];ftag=809628580;lr=on>
Record-Route: <sip:[email protected];ftag=809628580;lr=on>
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=0
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=0
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bK66b2.5e7d3ec2.0
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:5060;branch=z9hG4bK93c4b19e10af5ba9443f1c0135f7cdad
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:62628;rport;branch=z9hG4bKDF8A45EDC7C44A78A062E9498D719499
From: aswin <sip:1010@realm>;tag=809628580
To: <sip:telefonnummer@realm>
Call-ID: [email][email protected][/email]
CSeq: 10233 INVITE
Contact: <sip:[email protected]>
Max-forwards: 67
User-agent: X-Lite release 1103m
Content-Type: application/sdp
Content-Length:   217

v=0
o=1010 1517015 1517062 IN IP4 ZZZ.ZZZ.ZZZ.ZZZ
s=X-Lite
c=IN IP4 ZZZ.ZZZ.ZZZ.ZZZ
t=0 0
m=audio 7077 RTP/AVP 0 8 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

--- (17 headers 10 lines)---
Ignoring this request
Transmitting (NAT) to YYY.YYY.YYY.YYY:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=0;received=YYY.YYY.YYY.YYY;rport=5060
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=0
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bK66b2.5e7d3ec2.0
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:5060;branch=z9hG4bK93c4b19e10af5ba9443f1c0135f7cdad
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:62628;branch=z9hG4bKDF8A45EDC7C44A78A062E9498D719499
From: aswin <sip:1010@realm>;tag=809628580
To: <sip:telefonnummer@realm>
Call-ID: [email][email protected][/email]
CSeq: 10233 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:[email protected]>
Content-Length: 0


---
asterisk*CLI>
<-- SIP read from YYY.YYY.YYY.YYY:5060:
INVITE sip:telefonnummer@realm SIP/2.0
Record-Route: <sip:[email protected];ftag=809628580;lr=on>
Record-Route: <sip:[email protected];ftag=809628580;lr=on>
Record-Route: <sip:[email protected];ftag=809628580;lr=on>
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=0
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=0
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bK66b2.6e7d3ec2.0
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=0
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bK66b2.5e7d3ec2.0
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:5060;branch=z9hG4bK93c4b19e10af5ba9443f1c0135f7cdad
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:62628;rport;branch=z9hG4bKDF8A45EDC7C44A78A062E9498D719499
From: aswin <sip:1010@realm>;tag=809628580
To: <sip:telefonnummer@realm>
Call-ID: [email][email protected][/email]
CSeq: 10233 INVITE
Contact: <sip:[email protected]>
Max-forwards: 66
User-agent: X-Lite release 1103m
Content-Type: application/sdp
Content-Length:   217

v=0
o=1010 1517015 1517062 IN IP4 ZZZ.ZZZ.ZZZ.ZZZ
s=X-Lite
c=IN IP4 ZZZ.ZZZ.ZZZ.ZZZ
t=0 0
m=audio 7077 RTP/AVP 0 8 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

--- (20 headers 10 lines)---
Ignoring this request
Transmitting (NAT) to YYY.YYY.YYY.YYY:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=0;received=YYY.YYY.YYY.YYY;rport=5060
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=0
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bK66b2.6e7d3ec2.0
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=0
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bK66b2.5e7d3ec2.0
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:5060;branch=z9hG4bK93c4b19e10af5ba9443f1c0135f7cdad
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:62628;branch=z9hG4bKDF8A45EDC7C44A78A062E9498D719499
From: aswin <sip:1010@realm>;tag=809628580
To: <sip:telefonnummer@realm>
Call-ID: [email][email protected][/email]
CSeq: 10233 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:[email protected]>
Content-Length: 0


---
We're at 85.90.131.59 port 13816
Answering with preferred capability 0x8 (alaw)
Answering with preferred capability 0x4 (ulaw)
Answering with preferred capability 0x2 (gsm)
Answering with non-codec capability 0x1 (telephone-event)
Transmitting (NAT) to YYY.YYY.YYY.YYY:5060:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=0;received=YYY.YYY.YYY.YYY;rport=5060
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:5060;branch=z9hG4bK93c4b19e10af5ba9443f1c0135f7cdad
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:62628;branch=z9hG4bKDF8A45EDC7C44A78A062E9498D719499
From: aswin <sip:1010@realm>;tag=809628580
To: <sip:telefonnummer@realm>;tag=as1d35229a
Call-ID: [email][email protected][/email]
CSeq: 10233 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 19780 19780 IN IP4 85.90.131.59
s=session
c=IN IP4 85.90.131.59
t=0 0
m=audio 13816 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---

ab diesem zeitpunkt habe ich aufgelegt und es ist kein weiterer eintrag im
logging vom asterisk gekommen...

nachdem ich aber auf der gegenstelle aufgelegt habe, kommt wie
erwartet eine busy meldung.

Transmitting (NAT) to YYY.YYY.YYY.YYY:5060:
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=0;received=YYY.YYY.YYY.YYY;rport=5060
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:5060;branch=z9hG4bK93c4b19e10af5ba9443f1c0135f7cdad
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:62628;branch=z9hG4bKDF8A45EDC7C44A78A062E9498D719499
From: aswin <sip:1010@realm>;tag=809628580
To: <sip:telefonnummer@realm>;tag=as1d35229a
Call-ID: [email][email protected][/email]
CSeq: 10233 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:[email protected]>
Content-Length: 0


---
Destroying call '[email protected]'

EDIT: rollo hat das mal für Dich in Code Tags gesetzt, bitte beim nächsten mal selbst daran denken.

:
edit:
@rollo: danke

hab noch etwas interessantes bei tcpdump herausgefunden und zwar,
dass wenn der sip client ueber den siproxd den CANCEL request schickt,
dass vom sip server die antwort 404 an den proxy geschickt wird...
woran kann das liegen?

Code:

SEND TIME: 8518968 SEND >> ZZZ.ZZZ.ZZZ.ZZZ:5060 CANCEL sip:telefonnummer@realm SIP/2.0 Via: SIP/2.0/UDP PPP.PPP.PPP.PPP:5060;rport;branch=z9hG4bK4E244D65DCF746F4A3524669F0788C3A From: aswin <sip:1010@realm>;tag=3820329180 To: <sip:telefonnummer@realm> Contact: <sip:[email protected]:5060> Call-ID: [email][email protected][/email] CSeq: 64292 CANCEL Max-Forwards: 70 User-Agent: X-Lite release 1103m Content-Length: 0 RECEIVE TIME: 8518968 RECEIVE << ZZZ.ZZZ.ZZZ.ZZZ:5060 SIP/2.0 404 Not Found Via: SIP/2.0/UDP PPP.PPP.PPP.PPP:5060;rport;branch=z9hG4bK4E244D65DCF746F4A3524669F0788C3A Record-Route: <sip:[email protected]:5060;lr> From: aswin <sip:1010@realm>;tag=3820329180 To: <sip:telefonnummer@realm>;tag=dad9b29f0a661df3dd943e817af01760.bee1 Call-ID: [email][email protected][/email] CSeq: 64292 CANCEL Server: Sip EXpress router (0.8.14 (i386/linux)) Warning: 392 YYY.YYY.YYY.YYY:5060 "Noisy feedback tells: pid=22343 req_src_ip=ZZZ.ZZZ.ZZZ.ZZZ req_src_port=5060 in_uri=sip:telefonnummer@realm out_uri=sip:telefonnummer@realm via_cnt==2" Content-Length: 0 SEND TIME: 8520671 SEND >> ZZZ.ZZZ.ZZZ.ZZZ:5060 CANCEL sip:telefonnummer@realm SIP/2.0 Via: SIP/2.0/UDP PPP.PPP.PPP.PPP:5060;rport;branch=z9hG4bK4E244D65DCF746F4A3524669F0788C3A From: aswin <sip:1010@realm>;tag=3820329180 To: <sip:telefonnummer@realm> Contact: <sip:[email protected]:5060> Call-ID: [email][email protected][/email] CSeq: 64292 CANCEL Max-Forwards: 70 User-Agent: X-Lite release 1103m Content-Length: 0 RECEIVE TIME: 8520687 RECEIVE << ZZZ.ZZZ.ZZZ.ZZZ:5060 SIP/2.0 404 Not Found Via: SIP/2.0/UDP PPP.PPP.PPP.PPP:5060;rport;branch=z9hG4bK4E244D65DCF746F4A3524669F0788C3A Record-Route: <sip:[email protected]:5060;lr> From: aswin <sip:1010@realm>;tag=3820329180 To: <sip:telefonnummer@realm>;tag=dad9b29f0a661df3dd943e817af01760.bee1 Call-ID: [email][email protected][/email] CSeq: 64292 CANCEL Server: Sip EXpress router (0.8.14 (i386/linux)) Warning: 392 YYY.YYY.YYY.YYY:5060 "Noisy feedback tells: pid=15813 req_src_ip=ZZZ.ZZZ.ZZZ.ZZZ req_src_port=5060 in_uri=sip:telefonnummer@realm out_uri=sip:telefonnummer@realm via_cnt==2" Content-Length: 0

Languages

Suche

Languages

Suche

Auflegen waehrend dem Anrufen funktioniert nicht

RTDI

Neuer User

Maik

Gesperrt

RTDI

Neuer User

Zurzeit aktive Besucher

Neueste Beiträge

Statistik des Forums