[Gelöst] C430A GO Anrufer hört mich nicht

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StefanBr

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Hallo,

folgendes Problem:
Wenn jemand bei mir anruft und ich den Anruf mit dem Gigaset C430A GO annehme, hört mich der Anrufer nicht. Ich höre ihn jedoch schon.

Das Problem habe ich mit SIP Clients am PC (Bria4) nicht. Hier funktioniert die Telefonie ohne Probleme.

Anrufe nach draußen mit dem Gigaset funktionieren ohne Probleme.

Folgende Einstellungen sind in der TK betreffend des Gigasets hinterlegt:
sip.conf
Code: 
[2006]type = friend
secret = xxx
host = dynamic
qualify = yes


extensions.conf
Code: 
[AlleKlingeln]
exten => _X.,1,NoOp(Alle Klingeln)
exten => _X.,2,Dial(SIP/2000&SIP/2001&SIP/2002&SIP/2003&SIP/2005&SIP/2006,30,tTr)

RTP-Port Range ist von 10000 bis 20000.
Asterisk: 192.168.1.230
Gigaset: 192.168.1.231

Gigaset und Asterisk befinden sich im selben Netz.

Anbei Asterisk Konsole Auszug mit aktivierten SIP und RTP Debug während eines Anrufes:
Code: 
<------------->
--- (8 headers 0 lines) ---
[Jun  1 08:36:32] NOTICE[15785]: chan_sip.c:23665 handle_response_peerpoke: Peer '2006' is now Reachable. (35ms / 2000ms)
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
[Jun  1 08:36:52] NOTICE[15785]: chan_sip.c:15104 sip_reregister:    -- Re-registration for  [email protected]
[Jun  1 08:36:52] NOTICE[15785]: chan_sip.c:23615 handle_response_register: Outbound Registration: Expiry for 213.172.xxx.xxx is 30 sec (Scheduling reregistration in 24 s)
  == Using SIP RTP CoS mark 5
    -- Executing [xxxx@from-sip:1] NoOp("SIP/chiemgau-0000000f", "<sip:[email protected]>;tag=SDmvm1101-00E0F510089D11D1448BE8F87002") in new stack
    -- Executing [xxxx@from-sip:2] Goto("SIP/chiemgau-0000000f", "split-dw,xxx") in new stack
    -- Goto (split-dw,xxx)
    -- Executing [xxxx@split-dw:1] Dial("SIP/chiemgau-0000000f", "SIP/2000&SIP/2002&SIP/2003&SIP/2005&SIP/2006") in new stack
  == Using SIP RTP CoS mark 5
  == Using SIP RTP CoS mark 5
  == Using SIP RTP CoS mark 5
[Jun  1 08:36:53] WARNING[16659][C-00000003]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
  == Using SIP RTP CoS mark 5
    -- Called SIP/2000
    -- Called SIP/2002
    -- Called SIP/2003
Audio is at 11516
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100017 (testlaw) to SDP
Reliably Transmitting (NAT) to 192.168.1.231:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.230:5060;branch=z9hG4bK290700a5;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as349f94d8
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.10.2
Date: Wed, 01 Jun 2016 06:36:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 227




v=0
o=root 826803036 826803036 IN IP4 192.168.1.230
s=Asterisk PBX 11.10.2
c=IN IP4 192.168.1.230
t=0 0
m=audio 11516 RTP/AVP 0 8 3
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=ptime:20
a=sendrecv




---
    -- Called SIP/2006
  == Extension Changed 2000[lokal] new state Ringing for Notify User 2003
  == Extension Changed 2002[lokal] new state Ringing for Notify User 2000
  == Extension Changed 2002[lokal] new state Ringing for Notify User 2003
  == Extension Changed 2003[lokal] new state Ringing for Notify User 2000
  == Extension Changed 2006[lokal] new state Ringing for Notify User 2000




<--- SIP read from UDP:192.168.1.231:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.230:5060;branch=z9hG4bK290700a5;rport=5060
From: <sip:[email protected]>;tag=as349f94d8
To: <sip:[email protected]:5060>;tag=675870565
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Contact: <sip:[email protected]:5060>
User-Agent: C430A GO/42.231.00.000.000
Content-Length: 0




<------------->
--- (9 headers 0 lines) ---
    -- SIP/2002-00000011 is ringing
  == Extension Changed 2002[lokal] new state Ringing for Notify User 2000
  == Extension Changed 2002[lokal] new state Ringing for Notify User 2003
    -- SIP/2000-00000010 is ringing
    -- SIP/2000-00000010 is ringing
  == Extension Changed 2000[lokal] new state Ringing for Notify User 2003
    -- SIP/2003-00000012 is ringing
  == Extension Changed 2003[lokal] new state Ringing for Notify User 2000




<--- SIP read from UDP:192.168.1.231:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.230:5060;branch=z9hG4bK290700a5;rport=5060
From: <sip:[email protected]>;tag=as349f94d8
To: <sip:[email protected]:5060>;tag=675870565
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Contact: <sip:[email protected]:5060>
Allow-Events: message-summary, refer, ua-profile, talk, check-sync
User-Agent: C430A GO/42.231.00.000.000
Content-Length: 0




<------------->
--- (10 headers 0 lines) ---
list_route: hop: <sip:[email protected]:5060>
    -- SIP/2006-00000013 is ringing
  == Extension Changed 2006[lokal] new state Ringing for Notify User 2000




<--- SIP read from UDP:192.168.1.231:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.230:5060;branch=z9hG4bK290700a5;rport=5060
From: <sip:[email protected]>;tag=as349f94d8
To: <sip:[email protected]:5060>;tag=675870565
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Contact: <sip:[email protected]:5060>
Supported: replaces
Allow-Events: message-summary, refer, ua-profile, talk, check-sync
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY, REFER
Content-Type: application/sdp
Content-Length: 212




v=0
o=2006 10006 45 IN IP4 192.168.1.231
s=Mapping
c=IN IP4 192.168.1.231
t=0 0
m=audio 10006 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=ptime:20
<------------->
--- (12 headers 11 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event|), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.1.231:10006
Peer doesn't provide T.140
list_route: hop: <sip:[email protected]:5060>
set_destination: Parsing <sip:[email protected]:5060> for address/port to send to
set_destination: set destination to 192.168.1.231:5060
Transmitting (NAT) to 192.168.1.231:5060:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.230:5060;branch=z9hG4bK31ededfe;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as349f94d8
To: <sip:[email protected]:5060>;tag=675870565
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.10.2
Content-Length: 0








---
    -- SIP/2006-00000013 answered SIP/chiemgau-0000000f
    -- Remotely bridging SIP/chiemgau-0000000f and SIP/2006-00000013
set_destination: Parsing <sip:[email protected]:5060> for address/port to send to
set_destination: set destination to 192.168.1.231:5060
Audio is at 11516
Adding codec 100003 (ulaw) to SDP
Reliably Transmitting (NAT) to 192.168.1.231:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.230:5060;branch=z9hG4bK4eff8314;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as349f94d8
To: <sip:[email protected]:5060>;tag=675870565
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 11.10.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 182




v=0
o=root 826803036 826803037 IN IP4 213.172.xxx.xxx
s=Asterisk PBX 11.10.2
c=IN IP4 213.172.xxx.xxx
t=0 0
m=audio 60150 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv




---
  == Extension Changed 2006[lokal] new state InUse for Notify User 2000
  == Extension Changed 2000[lokal] new state Idle for Notify User 2003
  == Extension Changed 2002[lokal] new state Idle for Notify User 2000
  == Extension Changed 2002[lokal] new state Idle for Notify User 2003
  == Extension Changed 2003[lokal] new state Idle for Notify User 2000
       > 0x278b0f0 -- Probation passed - setting RTP source address to 192.168.1.231:10006
Sent RTP P2P packet to 213.172.xxx.xxx:60150 (type 00, len 000160)
Retransmitting #1 (NAT) to 192.168.1.231:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.230:5060;branch=z9hG4bK4eff8314;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as349f94d8
To: <sip:[email protected]:5060>;tag=675870565
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 11.10.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 182




v=0
o=root 826803036 826803037 IN IP4 213.172.xxx.xxx
s=Asterisk PBX 11.10.2
c=IN IP4 213.172.xxx.xxx
t=0 0
m=audio 60150 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv




---
       > 0xb4e0cda8 -- Probation passed - setting RTP source address to 213.172.xxx.xxx:60150
Sent RTP P2P packet to 192.168.1.231:10006 (type 00, len 000160)
Sent RTP P2P packet to 192.168.1.231:10006 (type 00, len 000160)
Sent RTP P2P packet to 192.168.1.231:10006 (type 00, len 000160)
Sent RTP P2P packet to 192.168.1.231:10006 (type 00, len 000160)
Sent RTP P2P packet to 192.168.1.231:10006 (type 00, len 000160)




<--- SIP read from UDP:192.168.1.231:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.230:5060;branch=z9hG4bK4eff8314;rport=5060
From: <sip:[email protected]>;tag=as349f94d8
To: <sip:[email protected]:5060>;tag=675870565
Call-ID: [email protected]:5060
CSeq: 103 INVITE
Contact: <sip:[email protected]:5060>
User-Agent: C430A GO/42.231.00.000.000
Content-Length: 0




<------------->
--- (9 headers 0 lines) ---
Sent RTP P2P packet to 192.168.1.231:10006 (type 00, len 000160)




<--- SIP read from UDP:192.168.1.231:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.230:5060;branch=z9hG4bK4eff8314;rport=5060
From: <sip:[email protected]>;tag=as349f94d8
To: <sip:[email protected]:5060>;tag=675870565
Call-ID: [email protected]:5060
CSeq: 103 INVITE
Contact: <sip:[email protected]:5060>
User-Agent: C430A GO/42.231.00.000.000
Content-Length: 0




<------------->
--- (9 headers 0 lines) ---
Sent RTP P2P packet to 192.168.1.231:10006 (type 00, len 000160)




<--- SIP read from UDP:192.168.1.231:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.230:5060;branch=z9hG4bK4eff8314;rport=5060
From: <sip:[email protected]>;tag=as349f94d8
To: <sip:[email protected]:5060>;tag=675870565
Call-ID: [email protected]:5060
CSeq: 103 INVITE
Contact: <sip:[email protected]:5060>
Supported: replaces
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY, REFER
Content-Type: application/sdp
Content-Length: 212




v=0
o=2006 10006 45 IN IP4 192.168.1.231
s=Mapping
c=IN IP4 192.168.1.231
t=0 0
m=audio 10006 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=ptime:20
<------------->
--- (11 headers 11 lines) ---
set_destination: Parsing <sip:[email protected]:5060> for address/port to send to
set_destination: set destination to 192.168.1.231:5060
Transmitting (NAT) to 192.168.1.231:5060:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.230:5060;branch=z9hG4bK297ab2f1;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as349f94d8
To: <sip:[email protected]:5060>;tag=675870565
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX 11.10.2
Content-Length: 0








---
set_destination: Parsing <sip:[email protected]:5060> for address/port to send to
set_destination: set destination to 192.168.1.231:5060
Audio is at 11516
Adding codec 100003 (ulaw) to SDP
Reliably Transmitting (NAT) to 192.168.1.231:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.230:5060;branch=z9hG4bK1ae62d45;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as349f94d8
To: <sip:[email protected]:5060>;tag=675870565
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 104 INVITE
User-Agent: Asterisk PBX 11.10.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 182




v=0
o=root 826803036 826803038 IN IP4 213.172.xxx.xxx
s=Asterisk PBX 11.10.2
c=IN IP4 213.172.xxx.xxx
t=0 0
m=audio 60150 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv




---
Sent RTP P2P packet to 192.168.1.231:10006 (type 00, len 000160)
Sent RTP P2P packet to 192.168.1.231:10006 (type 00, len 000160)
Sent RTP P2P packet to 192.168.1.231:10006 (type 00, len 000160)
Sent RTP P2P packet to 192.168.1.231:10006 (type 00, len 000160)
Sent RTP P2P packet to 192.168.1.231:10006 (type 00, len 000160)
Retransmitting #1 (NAT) to 192.168.1.231:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.230:5060;branch=z9hG4bK1ae62d45;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as349f94d8
To: <sip:[email protected]:5060>;tag=675870565
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 104 INVITE
User-Agent: Asterisk PBX 11.10.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 182




v=0
o=root 826803036 826803038 IN IP4 213.172.xxx.xxx
s=Asterisk PBX 11.10.2
c=IN IP4 213.172.xxx.xxx
t=0 0
m=audio 60150 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv




---
Sent RTP P2P packet to 192.168.1.231:10006 (type 00, len 000160)
Sent RTP P2P packet to 192.168.1.231:10006 (type 00, len 000160)




<--- SIP read from UDP:192.168.1.231:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.230:5060;branch=z9hG4bK1ae62d45;rport=5060
From: <sip:[email protected]>;tag=as349f94d8
To: <sip:[email protected]:5060>;tag=675870565
Call-ID: [email protected]:5060
CSeq: 104 INVITE
Contact: <sip:[email protected]:5060>
User-Agent: C430A GO/42.231.00.000.000
Content-Length: 0




<------------->
--- (9 headers 0 lines) ---
Sent RTP P2P packet to 192.168.1.231:10006 (type 00, len 000160)
Sent RTP P2P packet to 192.168.1.231:10006 (type 00, len 000160)




<--- SIP read from UDP:192.168.1.231:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.230:5060;branch=z9hG4bK1ae62d45;rport=5060
From: <sip:[email protected]>;tag=as349f94d8
To: <sip:[email protected]:5060>;tag=675870565
Call-ID: [email protected]:5060
CSeq: 104 INVITE
Contact: <sip:[email protected]:5060>
User-Agent: C430A GO/42.231.00.000.000
Content-Length: 0




<------------->
--- (9 headers 0 lines) ---




<--- SIP read from UDP:192.168.1.231:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.230:5060;branch=z9hG4bK1ae62d45;rport=5060
From: <sip:[email protected]>;tag=as349f94d8
To: <sip:[email protected]:5060>;tag=675870565
Call-ID: [email protected]:5060
CSeq: 104 INVITE
Contact: <sip:[email protected]:5060>
Supported: replaces
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY, REFER
Content-Type: application/sdp
Content-Length: 212




v=0
o=2006 10006 45 IN IP4 192.168.1.231
s=Mapping
c=IN IP4 192.168.1.231
t=0 0
m=audio 10006 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=ptime:20
<------------->
--- (11 headers 11 lines) ---
set_destination: Parsing <sip:[email protected]:5060> for address/port to send to
set_destination: set destination to 192.168.1.231:5060
Transmitting (NAT) to 192.168.1.231:5060:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.230:5060;branch=z9hG4bK37212407;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as349f94d8
To: <sip:[email protected]:5060>;tag=675870565
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 104 ACK
User-Agent: Asterisk PBX 11.10.2
Content-Length: 0








---
Sent RTP P2P packet to 192.168.1.231:10006 (type 00, len 000160)
Sent RTP P2P packet to 192.168.1.231:10006 (type 00, len 000160)
Sent RTP P2P packet to 192.168.1.231:10006 (type 00, len 000160)
Sent RTP P2P packet to 192.168.1.231:10006 (type 00, len 000160)
Sent RTP P2P packet to 192.168.1.231:10006 (type 00, len 000160)
Sent RTP P2P packet to 192.168.1.231:10006 (type 00, len 000160)
.... ca 300x folgend


<--- SIP read from UDP:192.168.1.231:5060 --->
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.231:5060;branch=z9hG4bK9d695de66a34f7bc521c6b4c7a07d37e;rport
From: <sip:[email protected]:5060>;tag=675870565
To: <sip:[email protected]>;tag=as349f94d8
Call-ID: [email protected]:5060
CSeq: 103 BYE
Contact: <sip:[email protected]:5060>
Max-Forwards: 70
User-Agent: C430A GO/42.231.00.000.000
Content-Length: 0




<------------->
--- (10 headers 0 lines) ---
Sending to 192.168.1.231:5060 (NAT)
Scheduling destruction of SIP dialog '[email protected]:5060' in 6400 ms (Method: BYE)




<--- Transmitting (NAT) to 192.168.1.231:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.231:5060;branch=z9hG4bK9d695de66a34f7bc521c6b4c7a07d37e;received=192.168.1.231;rport=5060
From: <sip:[email protected]:5060>;tag=675870565
To: <sip:[email protected]>;tag=as349f94d8
Call-ID: [email protected]:5060
CSeq: 103 BYE
Server: Asterisk PBX 11.10.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0








<------------>
  == Extension Changed 2006[lokal] new state Idle for Notify User 2000
  == Spawn extension (split-dw, xxx, 1) exited non-zero on 'SIP/chiemgau-0000000f'
Really destroying SIP dialog '1680697243@192_168_1_231' Method: REGISTER
Really destroying SIP dialog '[email protected]:5060' Method: BYE


 
Zuletzt bearbeitet:
Hallo StefanBr,

das Gigaset C430A ist ein analoges Telefon. Um dieses an einem SIP-Anschluss zu betreiben braucht es einen SIP-ATA. Oder benutzt du das Gigaset C430A GO?
Welche Einstellungen sind im Gigaset vorgenommen worden?


Gruss
Catalonia
 
Sorry, also ja es ist das C430A GO

Folgende Einstellungen:
Telefonie -> Verbindungen -> 1. VoIP-Verbindung
Code: 
Anmeldename: 2006
Passwort: xxxx
Benutzername: 2006
Domain: 192.168.1.230
Registration-Server: 192.168.1.230
Registration-Serverport: 5060
Anmelde-Refreshzeit: 180 Sek
Stun: Nein
Outbound-Proxy: Nie
Netzwerkprotokoll: Automatisch

Telefonie -> Audio
Code: 
Sprachqualität: Eigene Codec-Präferenz
Ausgewählte Codecs: G.711 a law
G.711 u law
G.722
G.729
Annex B für Codec G.729 aktivieren: Nein

Telefonie -> Weitere VoIP Einstellungen:
Code: 
Automatisches Aushandeln der MFV-Übertragung: Ja
R-Taste: Ja
Anruf übergeben durch Auflegen: Ja
Zieladresse automatisch ermitteln: Ja
Halten im Gerät: Beide Optionen
Zufällige Ports benutzen: Nein
SIP-Port: 5060
RTP-Port: 10000 - 20000
Wartemelodie: Ja
 
Perfekt - vielen herzlichen Dank - genau das war es. :)

Eine Frage noch aus Interesse - kannst du mir erklären warum beim Gigaset diese Option benötigt wird, bei den Softphones jedoch nicht?

Ganz so habe ich die Thematik was das betrifft wohl noch nicht verstanden...
 
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