[Problem] Sipcli - Text bricht nach 2 Sekunden ab

[leseratte10]

Hallo,

ich möchte mir ein Script erstellen, was mich anruft, wenn eine neue Mail eingegangen ist, und mir die dann vorliest.
Dazu habe ich das Tool "sipcli" (http://www.yasinkaplan.com/SipCli/) gefunden.
Theoretisch wäre das gut geeignet, aber das Vorlesen bricht immer nach ca. 2 Sekunden ab (egal welcher Text).

D:\sipcli>sipcli **9 -u 620 -p 620 -n 620 -d 10.0.1.1:5060 -g 80 -l 3

17.06.2013 17:08:28.790 - sipcli is being started on : 10.0.1.102:5070

17.06.2013 17:08:28.790 - Windows NT 6.1.7601 Service Pack 1 [F] v 2.4

17.06.2013 17:08:29.104 - Audio encoding has been started...

17.06.2013 17:08:29.513 - Encoding of announce file(s) is finished (2)...

17.06.2013 17:08:29.518 - Calling **9.

17.06.2013 17:08:29.527 - SIP req. to : 10.0.1.1:5060 (UDP) [Send]

INVITE sip:**[email protected] SIP/2.0
To: "**9" <sip:**[email protected]>
From: "620" <sip:[email protected]>;tag=ba4be14c
Via: SIP/2.0/UDP 10.0.1.102:5070;branch=z9hG4bK-ca0e2f3675a0857b1784d027889331de;rport
Call-ID: ca0e2f3675a0857b1784d027889331de
CSeq: 1 INVITE
Contact: <sip:[email protected]:5070>
Max-Forwards: 70
Allow: INVITE, MESSAGE, ACK, CANCEL, BYE
User-Agent: sipcli/v2.4
Content-Type: application/sdp
Content-Length: 238

v=0
o=sipcli-Session 658668839 1853934229 IN IP4 10.0.1.102
s=sipcli
c=IN IP4 10.0.1.102
t=0 0
m=audio 5630 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
a=sendrecv


17.06.2013 17:08:29.608 - SIP req. from : 10.0.1.1:5060 (UDP)

SIP/2.0 401 Unauthorized
[Via] = SIP/2.0/UDP 10.0.1.102:5070;branch=z9hG4bK-ca0e2f3675a0857b1784d027889331de;rport=5070
[From] = "620" <sip:[email protected]>;tag=ba4be14c
[To] = "**9" <sip:**[email protected]>;tag=EC6A8CFCADEA24F0
[Call-ID] = ca0e2f3675a0857b1784d027889331de
[CSeq] = 1 INVITE
[WWW-Authenticate] = Digest realm="fritz.box", nonce="861890C46D5DD844"
[User-Agent] = AVM FRITZ!Box Fon WLAN 7270 v3 74.05.50 (Jan 14 2013)
[Content-Length] = 0

17.06.2013 17:08:29.611 - SIP req. to : 10.0.1.1:5060 (UDP) [SendAuth ACK]

ACK sip:**[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.0.1.102:5070;branch=z9hG4bK-ca0e2f3675a0857b1784d027889331de;rport
From: "620" <sip:[email protected]>;tag=ba4be14c
To: "**9" <sip:**[email protected]>;tag=EC6A8CFCADEA24F0
Call-ID: ca0e2f3675a0857b1784d027889331de
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0


17.06.2013 17:08:29.614 - SIP req. to : 10.0.1.1:5060 (UDP) [SendAuth]

INVITE sip:**[email protected] SIP/2.0
To: "**9" <sip:**[email protected]>
From: "620" <sip:[email protected]>;tag=ba4be14c
Via: SIP/2.0/UDP 10.0.1.102:5070;branch=z9hG4bK-6d7acb5d4e8e2bd3344dc7eb45e9ae1e;rport
Call-ID: ca0e2f3675a0857b1784d027889331de
CSeq: 2 INVITE
Contact: <sip:[email protected]:5070>
Max-Forwards: 70
Allow: INVITE, MESSAGE, ACK, CANCEL, BYE
User-Agent: sipcli/v2.4
Content-Type: application/sdp
Authorization: Digest username="620",realm="fritz.box",nonce="861890C46D5DD844",uri="sip:**[email protected]",response="01
663a67c495d96f1b9f234cc51cb568",algorithm=MD5
Content-Length: 238

v=0
o=sipcli-Session 658668839 1853934229 IN IP4 10.0.1.102
s=sipcli
c=IN IP4 10.0.1.102
t=0 0
m=audio 5630 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
a=sendrecv


17.06.2013 17:08:29.640 - SIP req. from : 10.0.1.1:5060 (UDP)

SIP/2.0 100 Trying
[Via] = SIP/2.0/UDP 10.0.1.102:5070;branch=z9hG4bK-6d7acb5d4e8e2bd3344dc7eb45e9ae1e;rport=5070
[From] = "620" <sip:[email protected]>;tag=ba4be14c
[To] = "**9" <sip:**[email protected]>
[Call-ID] = ca0e2f3675a0857b1784d027889331de
[CSeq] = 2 INVITE
[User-Agent] = AVM FRITZ!Box Fon WLAN 7270 v3 74.05.50 (Jan 14 2013)
[Content-Length] = 0


17.06.2013 17:08:29.664 - SIP req. from : 10.0.1.1:5060 (UDP)

SIP/2.0 183 Session Progress
[Via] = SIP/2.0/UDP 10.0.1.102:5070;branch=z9hG4bK-6d7acb5d4e8e2bd3344dc7eb45e9ae1e;rport=5070
[From] = "620" <sip:[email protected]>;tag=ba4be14c
[To] = "**9" <sip:**[email protected]>;tag=B5618E52C77F370A
[Call-ID] = ca0e2f3675a0857b1784d027889331de
[CSeq] = 2 INVITE
[Contact] = <sip:[email protected]>
[User-Agent] = AVM FRITZ!Box Fon WLAN 7270 v3 74.05.50 (Jan 14 2013)
[Content-Type] = application/sdp
[Content-Length] = 232

v=0
o=user 3292997 3292997 IN IP4 10.0.1.1
s=sipcli
c=IN IP4 10.0.1.1
t=0 0
m=audio 7078 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
a=sendrecv
a=rtcp:7079


17.06.2013 17:08:31.053 - SIP req. from : 10.0.1.1:5060 (UDP)

SIP/2.0 200 OK
[Via] = SIP/2.0/UDP 10.0.1.102:5070;branch=z9hG4bK-6d7acb5d4e8e2bd3344dc7eb45e9ae1e;rport=5070
[From] = "620" <sip:[email protected]>;tag=ba4be14c
[To] = "**9" <sip:**[email protected]>;tag=B5618E52C77F370A
[Call-ID] = ca0e2f3675a0857b1784d027889331de
[CSeq] = 2 INVITE
[Contact] = <sip:[email protected]>
[User-Agent] = AVM FRITZ!Box Fon WLAN 7270 v3 74.05.50 (Jan 14 2013)
[Supported] = 100rel,replaces,timer
[Allow-Events] = telephone-event,refer
[Allow] = INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE,PUBLISH
[Content-Type] = application/sdp
[Accept] = application/sdp, multipart/mixed
[Accept-Encoding] = identity
[Content-Length] = 232

v=0
o=user 3292997 3292997 IN IP4 10.0.1.1
s=sipcli
c=IN IP4 10.0.1.1
t=0 0
m=audio 7078 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
a=sendrecv
a=rtcp:7079

17.06.2013 17:08:31.056 - SIP req. to : 10.0.1.1:5060 (UDP) [Successful]

ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.0.1.102:5070;branch=z9hG4bK-2ac451de8ba7423d3ca036592ca8bb09;rport
From: "620" <sip:[email protected]>;tag=ba4be14c
To: "**9" <sip:**[email protected]>;tag=B5618E52C77F370A
Call-ID: ca0e2f3675a0857b1784d027889331de
CSeq: 2 ACK
Contact: <sip:[email protected]:5070>
Max-Forwards: 70
Authorization: Digest username="620",realm="fritz.box",nonce="861890C46D5DD844",uri="sip:**[email protected]",response="01
663a67c495d96f1b9f234cc51cb568",algorithm=MD5
Content-Length: 0


17.06.2013 17:08:31.058 - Call to **9 successful.


Reading "langer langer sehr sehr langer text" (1) (habe den Text hier mal gekürzt)


17.06.2013 17:08:34.133 - Closing RTP connection...

 Der Index war außerhalb des Arraybereichs.

17.06.2013 17:08:34.133 - SIP req. to : 10.0.1.1:5060 (UDP)

BYE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.0.1.102:5070;branch=z9hG4bK-655ed035324c878e55176ad85706499b;rport
From: "620" <sip:[email protected]>;tag=ba4be14c
To: "**9" <sip:**[email protected]>;tag=B5618E52C77F370A
Call-ID: ca0e2f3675a0857b1784d027889331de
CSeq: 3 BYE
Contact: <sip:[email protected]:5070>
Max-Forwards: 70
Content-Length: 0



17.06.2013 17:08:34.145 - SIP req. from : 10.0.1.1:5060 (UDP)

SIP/2.0 200 OK
[Via] = SIP/2.0/UDP 10.0.1.102:5070;branch=z9hG4bK-655ed035324c878e55176ad85706499b;rport=5070
[From] = "620" <sip:[email protected]>;tag=ba4be14c
[To] = "**9" <sip:**[email protected]>;tag=B5618E52C77F370A
[Call-ID] = ca0e2f3675a0857b1784d027889331de
[CSeq] = 3 BYE
[X-RTP-Stat] = CS=0;PS=124;ES=224;OS=19840;SP=0/0;SO=0;QS=-;PR=150;ER=224;OR=24000;CR=0;SR=0;QR=-;PL=0,0;BL=0;LS=0;
RB=0/0;SB=-/-;EN=PCMU;DE=PCMU;JI=11,0;DL=0,0,0;IP=10.0.1.1:7078,10.0.1.102:5630
[X-RTP-Stat-Add] = DQ=4;DSS=0;DS=0;PLCS=0;JS=0
[User-Agent] = AVM FRITZ!Box Fon WLAN 7270 v3 74.05.50 (Jan 14 2013)
[Supported] = 100rel,replaces,timer
[Allow-Events] = telephone-event,refer
[Content-Length] = 0



17.06.2013 17:08:34.147 - Call to **9 hanged up.

17.06.2013 17:08:35.057 - sipcli Service is being stopped.

Habt ihr ne Idee warum das so ist? Oder kennt ihr andere Softphones, die sich einfach per cli bedienen lassen?

Leseratte10