linphone-1.0.0pre4 läst sich nicht registrirern

lo4dro

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Hallo Leute.

Da ich sehr zufreiden mit linphone war, wollte ich auf die 1.0.0 umsteigen.
Ich habe dafür das neue Debian Packet genommen.

Wenn ich linphone nun starte, kann ich mich nicht anmelden.
Das steht im logfile von linphone:

Code:
| INFO1 | <eXosip.c: 273> eXosip: Reseting timer to 15s before waking up!
| INFO1 | <eXosip.c: 273> eXosip: Reseting timer to 15s before waking up!
| INFO1 | <eXosip.c: 273> eXosip: Reseting timer to 15s before waking up!
| INFO2 | <osip_transaction.c: 129> allocating transaction ressource 1 150653899
8
| INFO2 | <nict.c: 36> allocating NICT context
| INFO2 | <eXutils.c: 508> Not an IPv4 or IPv6 address: pbx
| INFO2 | <eXutils.c: 538> DNS resolution with pbx:5060
| INFO1 | <jcallback.c: 147> Message sent: 
REGISTER sip:pbx SIP/2.0
Via: SIP/2.0/UDP 217.197.84.178:5060;branch=z9hG4bK1557996354
From: <sip:[email protected]>;tag=597050837
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 1 REGISTER
Contact: <sip:[email protected]:5060>
Max-Forwards: 5
User-Agent: Linphone-1.0.0pre4/eXosip
Expires: 900
Content-Length: 0

(len=16 sizeof(addr)=128 28)
| INFO1 | <jcallback.c: 503> cb_sndregister (id=1)
| INFO1 | <eXosip.c: 280> eXosip: timer sec:0 usec:474848!
| INFO1 | <udp.c: 1992> Received message: 
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 217.197.84.178:5060;branch=z9hG4bK1557996354;received=217.197.8
4.178;rport=5060
From: <sip:[email protected]>;tag=597050837
To: <sip:[email protected]>;tag=as34d9eaa7
Call-ID: [email protected]
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
| INFO1 | <jcallback.c: 579> cb_rcv1xx (id=1)
| INFO1 | <eXosip.c: 280> eXosip: timer sec:0 usec:463177!
| INFO1 | <udp.c: 1992> Received message: 
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 217.197.84.178:5060;branch=z9hG4bK1557996354;received=217.197.8
4.178;rport=5060
From: <sip:[email protected]>;tag=597050837
To: <sip:[email protected]>;tag=as34d9eaa7
Call-ID: [email protected]
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[email protected]>
WWW-Authenticate: Digest realm="asterisk", nonce="3f2f94a1"
Content-Length: 0


| INFO1 | <jcallback.c: 1364> cb_rcv4xx (id=1)
| INFO1 | <eXosip.c: 280> eXosip: timer sec:4 usec:999985!
| INFO1 | <eXosip.c: 280> eXosip: timer sec:0 usec:1454!
| INFO1 | <jcallback.c: 216> cb_nict_kill_transaction (id=1)
| INFO1 | <eXosip.c: 273> eXosip: Reseting timer to 15s before waking up!
|  BUG  | <osip_transaction.c: 263> transaction already removed from list 1!
| INFO2 | <nict.c: 109> free nict ressource
| INFO2 | <eXosip.c: 1888> INFO: authinfo: "asterisk" "asterisk"
| INFO2 | <eXosip.c: 1888> INFO: authinfo: "asterisk" atosc.org
| INFO2 | <osip_transaction.c: 129> allocating transaction ressource 2 150653899
8
| INFO2 | <nict.c: 36> allocating NICT context
| INFO2 | <eXutils.c: 508> Not an IPv4 or IPv6 address: pbx
| INFO2 | <eXutils.c: 538> DNS resolution with pbx:5060
| INFO1 | <jcallback.c: 147> Message sent: 

REGISTER sip:pbx SIP/2.0
Via: SIP/2.0/UDP 217.197.84.178:5060;branch=z9hG4bK1170141626
From: <sip:[email protected]>;tag=597050837
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 2 REGISTER
Contact: <sip:[email protected]:5060>
Max-Forwards: 5
User-Agent: Linphone-1.0.0pre4/eXosip
Expires: 900
Content-Length: 0

 (len=16 sizeof(addr)=128 28)
| INFO1 | <jcallback.c: 503> cb_sndregister (id=2)
| INFO1 | <eXosip.c: 280> eXosip: timer sec:0 usec:494030!
| INFO1 | <udp.c: 1992> Received message: 
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 217.197.84.178:5060;branch=z9hG4bK1170141626;received=217.197.8
4.178;rport=5060
From: <sip:[email protected]>;tag=597050837
To: <sip:[email protected]>;tag=as34d9eaa7
Call-ID: [email protected]
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[email protected]>
Content-Length: 0


| INFO1 | <jcallback.c: 579> cb_rcv1xx (id=2)
| INFO1 | <eXosip.c: 280> eXosip: timer sec:0 usec:488203!
| INFO1 | <udp.c: 1992> Received message: 
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 217.197.84.178:5060;branch=z9hG4bK1170141626;received=217.197.8
4.178;rport=5060
From: <sip:[email protected]>;tag=597050837
To: <sip:[email protected]>;tag=as34d9eaa7
Call-ID: [email protected]
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[email protected]>
WWW-Authenticate: Digest realm="asterisk", nonce="6da274ee"
Content-Length: 0

| INFO1 | <jcallback.c: 1364> cb_rcv4xx (id=2)
| INFO1 | <eXosip.c: 280> eXosip: timer sec:4 usec:999986!
| INFO1 | <eXosip.c: 280> eXosip: timer sec:0 usec:1168!
| INFO1 | <jcallback.c: 216> cb_nict_kill_transaction (id=2)
| INFO1 | <eXosip.c: 273> eXosip: Reseting timer to 15s before waking up!
Wenn ich in der sip.conf die registrierung abschalte, dann kann ich ein ISDN Telefon anrufen. Die Verbindung wird aufgebaut aber man hört nichts.

Wenn ich die Sip-Testnummer von Sipgate anrufe dann geht nichts.

Wie gesagt, mit kphone und dem alten linphone geht alles.
Vielleicht hat jemand eine Idee.
 

Hupe

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Hmm, hast Du mal im Changelog nachgesehen, was so alles verändert würde? Vielleicht findet sich da ja ein Hinweis? Wird vielleicht das Passwort jetzt verschlüsselt, oder so?
Was sagt denn Asterisk selbst zu Deinen Versuchen dich zu resistrieren?
Wie sieht das mit den Rtp-Ports aus? Hat sich da was verändert?
 

lo4dro

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@Hupe
Schön wider von dir zu hören.
Das sind gute Fragen.

Werd mal schauen ob die das passwort verschlüsseln.

Hier mal die Ausgabe von "sip debug"
Code:
10.10.10.10 = Asterisk
11.10.10.10 = linphone
1.2.3.4        = weisnet

Sip read: 
REGISTER sip:10.10.10.10 SIP/2.0
Via: SIP/2.0/UDP 11.10.10.10:5060;branch=z9hG4bK927836100
From: "200" <sip:[email protected]>;tag=681957313
To: "200" <sip:[email protected]>
Call-ID: [email protected]
CSeq: 1 REGISTER
Contact: <sip:[email protected]:5060>
Max-Forwards: 5
User-Agent: Linphone-1.0.0pre4/eXosip
Expires: 900
Content-Length: 0
11 headers, 0 lines
Using latest request as basis request
Sending to 11.10.10.10 : 5060 (NAT)
Transmitting (NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 11.10.10.10:5060;branch=z9hG4bK927836100;received=11.10.10.10;rport=5060
From: "200" <sip:[email protected]>;tag=681957313
To: "200" <sip:[email protected]>;tag=as279f91da
Call-ID: [email protected]
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER

Contact: <sip:[email protected]>
Content-Length: 0


 to 11.10.10.10:5060
Transmitting (NAT):
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 11.10.10.10:5060;branch=z9hG4bK927836100;received=11.10.10.10;rport=5060
From: "200" <sip:[email protected]>;tag=681957313
To: "200" <sip:[email protected]>;tag=as279f91da
Call-ID: [email protected]
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[email protected]>
WWW-Authenticate: Digest realm="asterisk", nonce="5af26552"
Content-Length: 0


 to 11.10.10.10:5060
Scheduling destruction of call '[email protected]' in 15000 ms
11 headers, 0 lines
Reliably Transmitting:
OPTIONS sip:1.2.3.4 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.10:5060;branch=z9hG4bK302fc390
From: "asterisk" <sip:[email protected]>;tag=as308698d7
To: <sip:1.2.3.4>
Contact: <sip:[email protected]>
Call-ID: [email protected]

CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Mon, 08 Nov 2004 06:59:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Length: 0

 (no NAT) to 1.2.3.4:5060
pbx*CLI> 

Sip read: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.10:5060;branch=z9hG4bK302fc390;received=10.10.10.10;rport=5060
From: "asterisk" <sip:[email protected]>;tag=as308698d7
To: <sip:1.2.3.4>;tag=as7be67ca5
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: PURtel
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1.2.3.4>
Accept: application/sdp
Content-Length: 0


11 headers, 0 lines
Destroying call '[email protected]'
11 headers, 0 lines
Reliably Transmitting:
REGISTER sip:sipgate.de SIP/2.0
Via: SIP/2.0/UDP 10.10.10.10:5060;branch=z9hG4bK78cbd2f1
edit otaku42: Bitte die Code-Tags fuer solche Sachen verwenden.
 

lo4dro

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Das sind die änderungen im Change.log:

Some date : linphone-1.0.0
- switch from osipua to eXosip/osip2 for improved sip functionnalities and compliance.
- support for presence (busy, online...) for everyone in the address book.
- support for configuring multiple proxies.
- add support for jackd (contributed)


January 2004 : linphone-0.12.2
- add enum support (see RFC3241 and RFC3026)
Thanks to Rene Bartsch < ml at bartschnet dot de > for its usefull
and precious help.
- interactive presence box (no more need to click OK to confirm)
- update spanish translation
- alsa interface: the user can choose precisely the pcm device to be used
by setting the sound/alsadev parameter of the config file.
- use 1 RTP socket instead of 2: this makes linphone NAT-friendly.
 

Hupe

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Bin da im Moment ach n bischen überfragt. Viellecht meldet sch ja hier noch jemand, der auch linphone mit Asterisk nutzt.
 

3CX PBX - GRATIS
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