Problem mit Dialog-Info

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matthias2525

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Hallo,

ich habe Asterisk 1.6.2.1, Snom 320/360

Hallo ich habe ein Problem mit den Dialog Info.
Wenn ein Anruf zu übernehmen signalisiert wird, erscheint am Display "von" und "Zu" mit Jeweiliger Nummer.

Bei uns steht aber bei "Von" die angerufene Nebenstelle (bsp. Von-120) und bei "Zu" korrekterweise auch die angerufene Nebenstelle (bsp. Zu-120)

Snom config:

Dialog-Info Call Pickup:Yes
Tonsignal als Pickup-Info:Yes

Ich weiß nicht mehr woch ich suchen soll, bitte un hinweise.

mfg
 
Zuletzt bearbeitet:
Hallo Matthias,
Wenn mich nicht alles täuscht, ist das beim 1.6er-Pickup (mindestens) in gewissen Fällen so... (Müsste kurz im Code nachschauen.)
Ist also (wahrscheinlich) ein Asterisk-"Problem".
Update: Wäre in channels/chan_sip.c ab Zeile 11171, aber ich habe gerade keine Lust mehr, das zu analysieren bzw. nach Änderungen in späteren Versionen zu schauen (meine aber, da wäre in späteren 1.6.2er oder im 1.8er noch dran rumgeschraubt worden), sorry!
 
Zuletzt bearbeitet:
Danke für den Tipp,

habe leider keine guten Programmierkenntnisse.

Ich habe meinen Server auf Asterisk 1.6.2.6 upgedatet aber es brachte auch nichts.
Snoms sind alle 7.3.30.

Das ist ein lestiger Fehler, ich werde in meiner Firma schon täglich gelöchet wann das entlich funktioniert.

Es muss ja irgend eine lösung geben.
 
Hi,

du kannst dem Telefon nach dem Pickup ein NOTIFY schicken mit dem Caller und Called URI und Displayname geändert werden können.

Wie genau das in der Asterisk geht kann ich leider nicht sagen aber das würde dein Problem lösen :)

Das NOTIFY muss dann im Body

From: Caller Displayname <sip:[email protected]>
To: Callee Displayname <sip:[email protected]>

stehen haben.

HTH

Filip

matthias2525 schrieb:
Danke für den Tipp,

habe leider keine guten Programmierkenntnisse.

Ich habe meinen Server auf Asterisk 1.6.2.6 upgedatet aber es brachte auch nichts.
Snoms sind alle 7.3.30.

Das ist ein lestiger Fehler, ich werde in meiner Firma schon täglich gelöchet wann das entlich funktioniert.

Es muss ja irgend eine lösung geben.
Zum Vergrößern anklicken....
 
Hallo Leute!

Nun habe ich etwas gewarten, wegen den oben genannten Thema, aktelle updates haben nicht gebrach!

So weit ich Informiert bin bekommt das Snom die notwendigen Informationen per "dialog-info+xml".

Kommt auch beim Snom an:

Code: 
Sent to udp:192.168.0.2:5060 at 6/5/2010 09:28:41:373 (620 bytes):

SUBSCRIBE sip:*[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.21:2054;branch=z9hG4bK-347klabnauey;rport
From: <sip:[email protected]>;tag=cnoxu83gow
To: <sip:*[email protected];user=phone>
Call-ID: 3c6d9c095096-7igi7nt8ka4s
CSeq: 31081 SUBSCRIBE
Max-Forwards: 70
Contact: <sip:[email protected]:2054;line=dnc907u4>;reg-id=1
Event: dialog
Accept: application/dialog-info+xml
User-Agent: snom320/7.3.30
Authorization: Digest username="121",realm="asterisk",nonce="1785b39e",uri="sip:*[email protected];user=phone",response="53bc72c08a9c3ce192fc38698eeb8264",algorithm=MD5
Expires: 3600
Content-Length: 0


Nun möchte ich mal klären wo der fehler liegt, beim snom oder Asterisk.
Hierfür möchte ich irgendwie das gesendete XML ansehen.

Musste ungefähr so aussehen:

Code: 
"<dialog id=\"%s\" call-id=\"pickup-%s\" direction=\"recipient\">\n"
11227 	"<remote>\n"
11228 	/* See the limitations of this above. Luckily the phone seems to still be
11229 	happy when these values are not correct. */
11230 	"<identity display=\"%s\">%s</identity>\n"
11231 	"<target uri=\"%s\"/>\n"
11232 	"</remote>\n"
11233 	"<local>\n"
11234 	"<identity>%s</identity>\n"
11235 	"<target uri=\"%s\"/>\n"
11236 	"</local>\n",
11237 	p->exten, p->callid, local_display, local_target, local_target, mto, mto);
11238 	} else {
11239 	ast_str_append(&tmp, 0, "<dialog id=\"%s\">\n", p->exten);

Mich würde Interessiere ob Asterisk diese auch korrekt ausfüllt.

Kann mir bitte jemand einen Tip geben wie ich das gesendete XML ansehen kann.

DANKE!

mfg
Matthias
 
matthias2525 schrieb:
Kann mir bitte jemand einen Tip geben wie ich das gesendete XML ansehen kann.
Zum Vergrößern anklicken....
Entweder im Asterisk-Log (sip debug), auf dem snom-Telefon (SIP trace) oder mit wireshark.

Edit - Ach, das war ja das subscribe, das DU reinkopiert hast. Als subscribe auf *8 ist wohl falsch... Überprüfe doch mal bitte Deine snom-Konfiguration.
 
Zuletzt bearbeitet:
So jetzt hab ichs ausprobier:

Anruf von meinen Handy 0676 xxx auf intern 03xxx xxxxx 110

Sieht am snom 121 so aus:

Code: 
Received from udp:192.168.0.2:5060 at 6/5/2010 10:07:54:158 (551 bytes):

OPTIONS sip:[email protected]:2054;line=dnc907u4 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK3a569e97;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as6a835897
To: <sip:[email protected]:2054;line=dnc907u4>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.6
Date: Thu, 06 May 2010 08:07:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

Sent to udp:192.168.0.2:5060 at 6/5/2010 10:07:54:167 (605 bytes):

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK3a569e97;rport=5060
From: "asterisk" <sip:[email protected]>;tag=as6a835897
To: <sip:[email protected]:2054;line=dnc907u4>
Call-ID: [email protected]
CSeq: 102 OPTIONS
Contact: <sip:[email protected]:2054;line=dnc907u4>;reg-id=1
User-Agent: snom320/7.3.30
Accept-Language: en
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Content-Length: 0

Received from udp:192.168.0.2:5060 at 6/5/2010 10:08:31:219 (670 bytes):

NOTIFY sip:[email protected]:2054;line=dnc907u4 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK3c279fde;rport
Max-Forwards: 70
From: <sip:[email protected];user=phone>;tag=as28e0c86a
To: <sip:[email protected]>;tag=go9pijrt9m
Contact: <sip:[email protected]>
Call-ID: 3c55a68d5cfa-q7d8ocz2pob6
CSeq: 1741 NOTIFY
User-Agent: Asterisk PBX 1.6.2.6
Event: dialog
Content-Type: application/dialog-info+xml
Subscription-State: active
Content-Length: 205

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="1639" state="full" entity="sip:[email protected]">
<dialog id="110">
<state>terminated</state>
</dialog>
</dialog-info>

Sent to udp:192.168.0.2:5060 at 6/5/2010 10:08:31:236 (258 bytes):

SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK3c279fde;rport=5060
From: <sip:[email protected];user=phone>;tag=as28e0c86a
To: <sip:[email protected]>;tag=go9pijrt9m
Call-ID: 3c55a68d5cfa-q7d8ocz2pob6
CSeq: 1741 NOTIFY
Content-Length: 0

Sent to udp:192.168.0.2:5060 at 6/5/2010 10:08:41:833 (456 bytes):

SUBSCRIBE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.21:2054;branch=z9hG4bK-5lj1g26rbwsm;rport
From: <sip:[email protected]>;tag=37u19ck4nk
To: <sip:[email protected];user=phone>
Call-ID: 3c6da569c836-5gi4yxacu4sf
CSeq: 27420 SUBSCRIBE
Max-Forwards: 70
Contact: <sip:[email protected]:2054;line=dnc907u4>;reg-id=1
Event: dialog
Accept: application/dialog-info+xml
User-Agent: snom320/7.3.30
Expires: 3600
Content-Length: 0

Received from udp:192.168.0.2:5060 at 6/5/2010 10:08:41:840 (508 bytes):

SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.21:2054;branch=z9hG4bK-5lj1g26rbwsm;received=192.168.0.21;rport=2054
From: <sip:[email protected]>;tag=37u19ck4nk
To: <sip:[email protected];user=phone>;tag=as2cf2ca55
Call-ID: 3c6da569c836-5gi4yxacu4sf
CSeq: 27420 SUBSCRIBE
Server: Asterisk PBX 1.6.2.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="66cac3d6"
Content-Length: 0

Sent to udp:192.168.0.2:5060 at 6/5/2010 10:08:41:852 (623 bytes):

SUBSCRIBE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.21:2054;branch=z9hG4bK-mzkbo4gpn4nk;rport
From: <sip:[email protected]>;tag=37u19ck4nk
To: <sip:[email protected];user=phone>
Call-ID: 3c6da569c836-5gi4yxacu4sf
CSeq: 27421 SUBSCRIBE
Max-Forwards: 70
Contact: <sip:[email protected]:2054;line=dnc907u4>;reg-id=1
Event: dialog
Accept: application/dialog-info+xml
User-Agent: snom320/7.3.30
Authorization: Digest username="121",realm="asterisk",nonce="66cac3d6",uri="sip:[email protected];user=phone",response="36923116cf5bfc0749471ceddc2e93e7",algorithm=MD5
Expires: 3600
Content-Length: 0

Received from udp:192.168.0.2:5060 at 6/5/2010 10:08:41:881 (429 bytes):

SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.21:2054;branch=z9hG4bK-mzkbo4gpn4nk;received=192.168.0.21;rport=2054
From: <sip:[email protected]>;tag=37u19ck4nk
To: <sip:[email protected];user=phone>;tag=as2cf2ca55
Call-ID: 3c6da569c836-5gi4yxacu4sf
CSeq: 27421 SUBSCRIBE
Server: Asterisk PBX 1.6.2.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

Sent to udp:192.168.0.2:5060 at 6/5/2010 10:08:41:899 (454 bytes):

SUBSCRIBE sip:*[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.21:2054;branch=z9hG4bK-0oeg10l70pio;rport
From: <sip:[email protected]>;tag=cnoxu83gow
To: <sip:*[email protected];user=phone>
Call-ID: 3c6da569d97e-ejfjbe8gxuxj
CSeq: 31096 SUBSCRIBE
Max-Forwards: 70
Contact: <sip:[email protected]:2054;line=dnc907u4>;reg-id=1
Event: dialog
Accept: application/dialog-info+xml
User-Agent: snom320/7.3.30
Expires: 3600
Content-Length: 0

Received from udp:192.168.0.2:5060 at 6/5/2010 10:08:41:921 (507 bytes):

SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.21:2054;branch=z9hG4bK-0oeg10l70pio;received=192.168.0.21;rport=2054
From: <sip:[email protected]>;tag=cnoxu83gow
To: <sip:*[email protected];user=phone>;tag=as3477feab
Call-ID: 3c6da569d97e-ejfjbe8gxuxj
CSeq: 31096 SUBSCRIBE
Server: Asterisk PBX 1.6.2.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="485d9d85"
Content-Length: 0

Sent to udp:192.168.0.2:5060 at 6/5/2010 10:08:41:933 (620 bytes):

SUBSCRIBE sip:*[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.21:2054;branch=z9hG4bK-id16llkokn0i;rport
From: <sip:[email protected]>;tag=cnoxu83gow
To: <sip:*[email protected];user=phone>
Call-ID: 3c6da569d97e-ejfjbe8gxuxj
CSeq: 31097 SUBSCRIBE
Max-Forwards: 70
Contact: <sip:[email protected]:2054;line=dnc907u4>;reg-id=1
Event: dialog
Accept: application/dialog-info+xml
User-Agent: snom320/7.3.30
Authorization: Digest username="121",realm="asterisk",nonce="485d9d85",uri="sip:*[email protected];user=phone",response="3f8198db20bcdfe5ca2a0b9279bd1947",algorithm=MD5
Expires: 3600
Content-Length: 0

Received from udp:192.168.0.2:5060 at 6/5/2010 10:08:41:975 (428 bytes):

SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.21:2054;branch=z9hG4bK-id16llkokn0i;received=192.168.0.21;rport=2054
From: <sip:[email protected]>;tag=cnoxu83gow
To: <sip:*[email protected];user=phone>;tag=as3477feab
Call-ID: 3c6da569d97e-ejfjbe8gxuxj
CSeq: 31097 SUBSCRIBE
Server: Asterisk PBX 1.6.2.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

Received from udp:192.168.0.2:5060 at 6/5/2010 10:08:54:205 (551 bytes):

OPTIONS sip:[email protected]:2054;line=dnc907u4 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK1d4db4e6;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as05028be9
To: <sip:[email protected]:2054;line=dnc907u4>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.6
Date: Thu, 06 May 2010 08:08:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

Sent to udp:192.168.0.2:5060 at 6/5/2010 10:08:54:213 (605 bytes):

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK1d4db4e6;rport=5060
From: "asterisk" <sip:[email protected]>;tag=as05028be9
To: <sip:[email protected]:2054;line=dnc907u4>
Call-ID: [email protected]
CSeq: 102 OPTIONS
Contact: <sip:[email protected]:2054;line=dnc907u4>;reg-id=1
User-Agent: snom320/7.3.30
Accept-Language: en
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Content-Length: 0

Received from udp:192.168.0.2:5060 at 6/5/2010 10:09:54:236 (551 bytes):

OPTIONS sip:[email protected]:2054;line=dnc907u4 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK234b9cf0;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as70f854ae
To: <sip:[email protected]:2054;line=dnc907u4>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.6
Date: Thu, 06 May 2010 08:09:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

Sent to udp:192.168.0.2:5060 at 6/5/2010 10:09:54:245 (605 bytes):

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK234b9cf0;rport=5060
From: "asterisk" <sip:[email protected]>;tag=as70f854ae
To: <sip:[email protected]:2054;line=dnc907u4>
Call-ID: [email protected]
CSeq: 102 OPTIONS
Contact: <sip:[email protected]:2054;line=dnc907u4>;reg-id=1
User-Agent: snom320/7.3.30
Accept-Language: en
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Content-Length: 0

Received from udp:192.168.0.2:5060 at 6/5/2010 10:10:54:287 (551 bytes):

OPTIONS sip:[email protected]:2054;line=dnc907u4 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK09874a89;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as1a9b4eb5
To: <sip:[email protected]:2054;line=dnc907u4>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.6
Date: Thu, 06 May 2010 08:10:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

Sent to udp:192.168.0.2:5060 at 6/5/2010 10:10:54:297 (605 bytes):

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK09874a89;rport=5060
From: "asterisk" <sip:[email protected]>;tag=as1a9b4eb5
To: <sip:[email protected]:2054;line=dnc907u4>
Call-ID: [email protected]
CSeq: 102 OPTIONS
Contact: <sip:[email protected]:2054;line=dnc907u4>;reg-id=1
User-Agent: snom320/7.3.30
Accept-Language: en
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Content-Length: 0

Received from udp:192.168.0.2:5060 at 6/5/2010 10:11:54:318 (551 bytes):

OPTIONS sip:[email protected]:2054;line=dnc907u4 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK45c86968;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as3aebab96
To: <sip:[email protected]:2054;line=dnc907u4>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.6
Date: Thu, 06 May 2010 08:11:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

Sent to udp:192.168.0.2:5060 at 6/5/2010 10:11:54:328 (605 bytes):

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK45c86968;rport=5060
From: "asterisk" <sip:[email protected]>;tag=as3aebab96
To: <sip:[email protected]:2054;line=dnc907u4>
Call-ID: [email protected]
CSeq: 102 OPTIONS
Contact: <sip:[email protected]:2054;line=dnc907u4>;reg-id=1
User-Agent: snom320/7.3.30
Accept-Language: en
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Content-Length: 0

Received from udp:192.168.0.2:5060 at 6/5/2010 10:12:54:366 (551 bytes):

OPTIONS sip:[email protected]:2054;line=dnc907u4 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK6d30e997;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as09be05e1
To: <sip:[email protected]:2054;line=dnc907u4>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.6
Date: Thu, 06 May 2010 08:12:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

Sent to udp:192.168.0.2:5060 at 6/5/2010 10:12:54:376 (605 bytes):

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK6d30e997;rport=5060
From: "asterisk" <sip:[email protected]>;tag=as09be05e1
To: <sip:[email protected]:2054;line=dnc907u4>
Call-ID: [email protected]
CSeq: 102 OPTIONS
Contact: <sip:[email protected]:2054;line=dnc907u4>;reg-id=1
User-Agent: snom320/7.3.30
Accept-Language: en
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Content-Length: 0

Sent to udp:192.168.0.2:5060 at 6/5/2010 10:13:41:893 (456 bytes):

SUBSCRIBE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.21:2054;branch=z9hG4bK-wyhl0pz2n088;rport
From: <sip:[email protected]>;tag=37u19ck4nk
To: <sip:[email protected];user=phone>
Call-ID: 3c6da695d6de-g0zaxia6wqrh
CSeq: 27422 SUBSCRIBE
Max-Forwards: 70
Contact: <sip:[email protected]:2054;line=dnc907u4>;reg-id=1
Event: dialog
Accept: application/dialog-info+xml
User-Agent: snom320/7.3.30
Expires: 3600
Content-Length: 0

Received from udp:192.168.0.2:5060 at 6/5/2010 10:13:41:900 (508 bytes):

SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.21:2054;branch=z9hG4bK-wyhl0pz2n088;received=192.168.0.21;rport=2054
From: <sip:[email protected]>;tag=37u19ck4nk
To: <sip:[email protected];user=phone>;tag=as29113b0d
Call-ID: 3c6da695d6de-g0zaxia6wqrh
CSeq: 27422 SUBSCRIBE
Server: Asterisk PBX 1.6.2.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5e4d2b85"
Content-Length: 0

Sent to udp:192.168.0.2:5060 at 6/5/2010 10:13:41:912 (623 bytes):

SUBSCRIBE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.21:2054;branch=z9hG4bK-cwa0gt0hcaev;rport
From: <sip:[email protected]>;tag=37u19ck4nk
To: <sip:[email protected];user=phone>
Call-ID: 3c6da695d6de-g0zaxia6wqrh
CSeq: 27423 SUBSCRIBE
Max-Forwards: 70
Contact: <sip:[email protected]:2054;line=dnc907u4>;reg-id=1
Event: dialog
Accept: application/dialog-info+xml
User-Agent: snom320/7.3.30
Authorization: Digest username="121",realm="asterisk",nonce="5e4d2b85",uri="sip:[email protected];user=phone",response="f702b322c4902197fe7f294b03e5ec32",algorithm=MD5
Expires: 3600
Content-Length: 0

Received from udp:192.168.0.2:5060 at 6/5/2010 10:13:41:941 (429 bytes):

SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.21:2054;branch=z9hG4bK-cwa0gt0hcaev;received=192.168.0.21;rport=2054
From: <sip:[email protected]>;tag=37u19ck4nk
To: <sip:[email protected];user=phone>;tag=as29113b0d
Call-ID: 3c6da695d6de-g0zaxia6wqrh
CSeq: 27423 SUBSCRIBE
Server: Asterisk PBX 1.6.2.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

Sent to udp:192.168.0.2:5060 at 6/5/2010 10:13:41:996 (454 bytes):

SUBSCRIBE sip:*[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.21:2054;branch=z9hG4bK-ff607y6kug4v;rport
From: <sip:[email protected]>;tag=cnoxu83gow
To: <sip:*[email protected];user=phone>
Call-ID: 3c6da695f01d-44buvsadtgdj
CSeq: 31098 SUBSCRIBE
Max-Forwards: 70
Contact: <sip:[email protected]:2054;line=dnc907u4>;reg-id=1
Event: dialog
Accept: application/dialog-info+xml
User-Agent: snom320/7.3.30
Expires: 3600
Content-Length: 0

Received from udp:192.168.0.2:5060 at 6/5/2010 10:13:42:003 (507 bytes):

SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.21:2054;branch=z9hG4bK-ff607y6kug4v;received=192.168.0.21;rport=2054
From: <sip:[email protected]>;tag=cnoxu83gow
To: <sip:*[email protected];user=phone>;tag=as6cd558fe
Call-ID: 3c6da695f01d-44buvsadtgdj
CSeq: 31098 SUBSCRIBE
Server: Asterisk PBX 1.6.2.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7356f25d"
Content-Length: 0

Sent to udp:192.168.0.2:5060 at 6/5/2010 10:13:42:015 (620 bytes):

SUBSCRIBE sip:*[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.21:2054;branch=z9hG4bK-4c6yal2zggu9;rport
From: <sip:[email protected]>;tag=cnoxu83gow
To: <sip:*[email protected];user=phone>
Call-ID: 3c6da695f01d-44buvsadtgdj
CSeq: 31099 SUBSCRIBE
Max-Forwards: 70
Contact: <sip:[email protected]:2054;line=dnc907u4>;reg-id=1
Event: dialog
Accept: application/dialog-info+xml
User-Agent: snom320/7.3.30
Authorization: Digest username="121",realm="asterisk",nonce="7356f25d",uri="sip:*[email protected];user=phone",response="b450b40989546e92206043baacf7c862",algorithm=MD5
Expires: 3600
Content-Length: 0

Received from udp:192.168.0.2:5060 at 6/5/2010 10:13:42:041 (428 bytes):

SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.21:2054;branch=z9hG4bK-4c6yal2zggu9;received=192.168.0.21;rport=2054
From: <sip:[email protected]>;tag=cnoxu83gow
To: <sip:*[email protected];user=phone>;tag=as6cd558fe
Call-ID: 3c6da695f01d-44buvsadtgdj
CSeq: 31099 SUBSCRIBE
Server: Asterisk PBX 1.6.2.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

Received from udp:192.168.0.2:5060 at 6/5/2010 10:13:54:405 (551 bytes):

OPTIONS sip:[email protected]:2054;line=dnc907u4 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK4934dcc8;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as55e3dc48
To: <sip:[email protected]:2054;line=dnc907u4>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.6
Date: Thu, 06 May 2010 08:13:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

Sent to udp:192.168.0.2:5060 at 6/5/2010 10:13:54:414 (605 bytes):

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK4934dcc8;rport=5060
From: "asterisk" <sip:[email protected]>;tag=as55e3dc48
To: <sip:[email protected]:2054;line=dnc907u4>
Call-ID: [email protected]
CSeq: 102 OPTIONS
Contact: <sip:[email protected]:2054;line=dnc907u4>;reg-id=1
User-Agent: snom320/7.3.30
Accept-Language: en
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Content-Length: 0

Received from udp:192.168.0.2:5060 at 6/5/2010 10:14:16:741 (669 bytes):

NOTIFY sip:[email protected]:2054;line=dnc907u4 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK3636549a;rport
Max-Forwards: 70
From: <sip:[email protected];user=phone>;tag=as28e0c86a
To: <sip:[email protected]>;tag=go9pijrt9m
Contact: <sip:[email protected]>
Call-ID: 3c55a68d5cfa-q7d8ocz2pob6
CSeq: 1742 NOTIFY
User-Agent: Asterisk PBX 1.6.2.6
Event: dialog
Content-Type: application/dialog-info+xml
Subscription-State: active
Content-Length: 204

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="1640" state="full" entity="sip:[email protected]">
<dialog id="110">
<state>confirmed</state>
</dialog>
</dialog-info>

Sent to udp:192.168.0.2:5060 at 6/5/2010 10:14:16:760 (258 bytes):

SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK3636549a;rport=5060
From: <sip:[email protected];user=phone>;tag=as28e0c86a
To: <sip:[email protected]>;tag=go9pijrt9m
Call-ID: 3c55a68d5cfa-q7d8ocz2pob6
CSeq: 1742 NOTIFY
Content-Length: 0

Received from udp:192.168.0.2:5060 at 6/5/2010 10:14:54:444 (551 bytes):

OPTIONS sip:[email protected]:2054;line=dnc907u4 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK1abe95a9;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as32b756f8
To: <sip:[email protected]:2054;line=dnc907u4>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.6
Date: Thu, 06 May 2010 08:14:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

Sent to udp:192.168.0.2:5060 at 6/5/2010 10:14:54:453 (605 bytes):

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK1abe95a9;rport=5060
From: "asterisk" <sip:[email protected]>;tag=as32b756f8
To: <sip:[email protected]:2054;line=dnc907u4>
Call-ID: [email protected]
CSeq: 102 OPTIONS
Contact: <sip:[email protected]:2054;line=dnc907u4>;reg-id=1
User-Agent: snom320/7.3.30
Accept-Language: en
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Content-Length: 0

Received from udp:192.168.0.2:5060 at 6/5/2010 10:14:59:310 (934 bytes):

NOTIFY sip:[email protected]:2054;line=dnc907u4 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK4f3fe3bc;rport
Max-Forwards: 70
From: <sip:[email protected];user=phone>;tag=as28e0c86a
To: <sip:[email protected]>;tag=go9pijrt9m
Contact: <sip:[email protected]>
Call-ID: 3c55a68d5cfa-q7d8ocz2pob6
CSeq: 1743 NOTIFY
User-Agent: Asterisk PBX 1.6.2.6
Event: dialog
Content-Type: application/dialog-info+xml
Subscription-State: active
Content-Length: 469

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="1641" state="full" entity="sip:[email protected]">
<dialog id="110" call-id="pickup-3c55a68d5cfa-q7d8ocz2pob6" direction="recipient">
<remote>
<identity display="110">sip:[email protected]</identity>
<target uri="sip:[email protected]"/>
</remote>
<local>
<identity>sip:[email protected]</identity>
<target uri="sip:[email protected]"/>
</local>
<state>early</state>
</dialog>
</dialog-info>


ich finden da nirgens meine Handy nummer.

Ist das ein Asterisk Proplem (kann fast nicht sein) oder ein config fehler?

lg
 
Hallo!? Ist ein bisschen viel Log... Mindestens den OPTIONS-Mist könntest Du uns ersparen...
Das hier ist eigentlich der Teil, der interessiert (mit dem gesuchten XML):
Code: 
NOTIFY sip:[email protected]:2054;line=dnc907u4 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK3636549a;rport
Max-Forwards: 70
From: <sip:[email protected];user=phone>;tag=as28e0c86a
To: <sip:[email protected]>;tag=go9pijrt9m
Contact: <sip:[email protected]>
Call-ID: 3c55a68d5cfa-q7d8ocz2pob6
CSeq: 1742 NOTIFY
User-Agent: Asterisk PBX 1.6.2.6
Event: dialog
Content-Type: application/dialog-info+xml
Subscription-State: active
Content-Length: 204

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="1640" state="full" entity="sip:[email protected]">
<dialog id="110">
<state>confirmed</state>
</dialog>
</dialog-info>

Received from udp:192.168.0.2:5060 at 6/5/2010 10:14:59:310 (934 bytes):

NOTIFY sip:[email protected]:2054;line=dnc907u4 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK4f3fe3bc;rport
Max-Forwards: 70
From: <sip:[email protected];user=phone>;tag=as28e0c86a
To: <sip:[email protected]>;tag=go9pijrt9m
Contact: <sip:[email protected]>
Call-ID: 3c55a68d5cfa-q7d8ocz2pob6
CSeq: 1743 NOTIFY
User-Agent: Asterisk PBX 1.6.2.6
Event: dialog
Content-Type: application/dialog-info+xml
Subscription-State: active
Content-Length: 469

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="1641" state="full" entity="sip:[email protected]">
<dialog id="110" call-id="pickup-3c55a68d5cfa-q7d8ocz2pob6" direction="recipient">
<remote>
<identity display="110">sip:[email protected]</identity>
<target uri="sip:[email protected]"/>
</remote>
<local>
<identity>sip:[email protected]</identity>
<target uri="sip:[email protected]"/>
</local>
<state>early</state>
</dialog>
</dialog-info>
Zum Vergrößern anklicken....
...da siehst Du, was Dein snom vom Asterisk kriegt. 2x die 110, sowohl als remote als auch als local. Wie bereits erwähnt, bin ich der Meinung, "it's not a bug, it's a feature", d.h. der Asterisk 1.6 macht das (mindestens) in gewissen Fällen so, wenn er den "Absender" nicht kennt bzw. nicht ermitteln kann. Aber ich habe im Moment gerade keine Lust, mich durch den Asterisk-1.6-Code durchzulesen, habe gerade ein paar Hundert Zeilen SIP-Trace lesen hinter mir...
-> also eigentlich Asterisk-"Fehler".


matthias2525 schrieb:
So jetzt hab ichs ausprobier:
ich finden da nirgens meine Handy nummer.
Zum Vergrößern anklicken....
...und ich finde da mindestens 2x "SIP/2.0 404 Not Found":
Code: 
Sent to udp:192.168.0.2:5060 at 6/5/2010 10:08:41:852 (623 bytes):

SUBSCRIBE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.21:2054;branch=z9hG4bK-mzkbo4gpn4nk;rport
From: <sip:[email protected]>;tag=37u19ck4nk
To: <sip:[email protected];user=phone>
Call-ID: 3c6da569c836-5gi4yxacu4sf
CSeq: 27421 SUBSCRIBE

Received from udp:192.168.0.2:5060 at 6/5/2010 10:08:41:881 (429 bytes):

SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.21:2054;branch=z9hG4bK-mzkbo4gpn4nk;received=192.168.0.21;rport=2054
From: <sip:[email protected]>;tag=37u19ck4nk
To: <sip:[email protected];user=phone>;tag=as2cf2ca55
Call-ID: 3c6da569c836-5gi4yxacu4sf
CSeq: 27421 SUBSCRIBE
Server: Asterisk PBX 1.6.2.6

Sent to udp:192.168.0.2:5060 at 6/5/2010 10:08:41:933 (620 bytes):

SUBSCRIBE sip:*[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.21:2054;branch=z9hG4bK-id16llkokn0i;rport
From: <sip:[email protected]>;tag=cnoxu83gow
To: <sip:*[email protected];user=phone>
Call-ID: 3c6da569d97e-ejfjbe8gxuxj

Received from udp:192.168.0.2:5060 at 6/5/2010 10:08:41:975 (428 bytes):

SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.21:2054;branch=z9hG4bK-id16llkokn0i;received=192.168.0.21;rport=2054
From: <sip:[email protected]>;tag=cnoxu83gow
To: <sip:*[email protected];user=phone>;tag=as3477feab
Call-ID: 3c6da569d97e-ejfjbe8gxuxj
CSeq: 31097 SUBSCRIBE
Server: Asterisk PBX 1.6.2.6

Sent to udp:192.168.0.2:5060 at 6/5/2010 10:13:41:912 (623 bytes):

SUBSCRIBE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.21:2054;branch=z9hG4bK-cwa0gt0hcaev;rport
From: <sip:[email protected]>;tag=37u19ck4nk
To: <sip:[email protected];user=phone>

Received from udp:192.168.0.2:5060 at 6/5/2010 10:13:41:941 (429 bytes):

SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.21:2054;branch=z9hG4bK-cwa0gt0hcaev;received=192.168.0.21;rport=2054
From: <sip:[email protected]>;tag=37u19ck4nk
To: <sip:[email protected];user=phone>;tag=as29113b0d
Call-ID: 3c6da695d6de-g0zaxia6wqrh
CSeq: 27423 SUBSCRIBE
Server: Asterisk PBX 1.6.2.6

Sent to udp:192.168.0.2:5060 at 6/5/2010 10:13:42:015 (620 bytes):

SUBSCRIBE sip:*[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.21:2054;branch=z9hG4bK-4c6yal2zggu9;rport
From: <sip:[email protected]>;tag=cnoxu83gow
To: <sip:*[email protected];user=phone>

Received from udp:192.168.0.2:5060 at 6/5/2010 10:13:42:041 (428 bytes):

SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.21:2054;branch=z9hG4bK-4c6yal2zggu9;received=192.168.0.21;rport=2054
From: <sip:[email protected]>;tag=cnoxu83gow
To: <sip:*[email protected];user=phone>;tag=as6cd558fe
Call-ID: 3c6da695f01d-44buvsadtgdj
CSeq: 31099 SUBSCRIBE
Server: Asterisk PBX 1.6.2.6
Zum Vergrößern anklicken....
-> Konfigurations-Fehler?
 
Hallo,

sorry für den vielen code!

Leider hab ich keine c kenntnisse.

Das ist der code aus chan_sip.c

Code: 
ast_str_append(&tmp, 0,
11226 	"<dialog id=\"%s\" call-id=\"pickup-%s\" direction=\"recipient\">\n"
11227 	"<remote>\n"
11228 	/* See the limitations of this above. Luckily the phone seems to still be
11229 	happy when these values are not correct. */
11230 	"<identity display=\"%s\">%s</identity>\n"
11231 	"<target uri=\"%s\"/>\n"
11232 	"</remote>\n"
11233 	"<local>\n"
11234 	"<identity>%s</identity>\n"
11235 	"<target uri=\"%s\"/>\n"
11236 	"</local>\n",
11237 	p->exten, p->callid, local_display, local_target, local_target, mto, mto);
11238 	}

p->exten, p->callid: ist logisch

Code: 
const char *local_display = p->exten;
char *local_target = mto;

ist für mich nicht logisch!!
 
Ich hab den Code nur ganz kurz überflogen auch nicht verfolgt, wo die einzelnen Werte herkommen.

Aber: ist notifycid gesetzt und ist kein Subscribecontext definiert?
 
Hallo,

danke für den Hinweis, also

meine sip.conf

(ausschnitt)
Code: 
[general]

allowsubscribe=yes


[121]
subscribecontext=sip-hints


notifycid ist nirges gesetzt, wie genau mus ich das setzten, was bewirkt dieses?

mfg
 
Ich zitiere mal die Beispielkonfiguration für 1.6.2

http://svnview.digium.com/svn/aster...s/sip.conf.sample?revision=260618&view=markup

Code: 
430 	;notifycid = yes ; Control whether caller ID information is sent along with
431 	; dialog-info+xml notifications (supported by snom phones).
432 	; Note that this feature will only work properly when the
433 	; incoming call is using the same extension and context that
434 	; is being used as the hint for the called extension. This means
435 	; that it won't work when using subscribecontext for your sip
436 	; user or peer (if subscribecontext is different than context).
437 	; This is also limited to a single caller, meaning that if an
438 	; extension is ringing because multiple calls are incoming,
439 	; only one will be used as the source of caller ID. Specify
440 	; 'ignore-context' to ignore the called context when looking
441 	; for the caller's channel. The default value is 'no.' Setting
442 	; notifycid to 'ignore-context' also causes call-pickups attempted
443 	; via SNOM's NOTIFY mechanism to set the context for the call pickup
444 	; to PICKUPMARK.
 
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