IP-Phone --> Asterisk --> SER --> SIP Proxy (Carrie

Toggle sidebar Toggle sidebar
Upgrade 3CX to v18 and get it hosted free!
Status
Für weitere Antworten geschlossen.
A

Abarta

Neuer User
Mitglied seit
2 Nov 2005
Beiträge
2
Punkte für Reaktionen
0
Punkte
0
Hallo,

ich habe ein Problem mit dem Zusammenspiel von Asterisk / SER. Ich habe einen Server mit 2 Netzwerkkarten. Eine Netzwerkkarte ist für die Anbindung der Endgeräte an den Asterisk.
Die andere Netzwerkkarte ist für die In- und Outbound-Telefonie vom und ins Festnetz über einen SIP-Provider.

Für die Konfigurationsbeispiele habe ich folgende IP-Adressen verwendet:

IP der 1. Netzwerkkarte (Asterisk): 192.168.0.1
IP der 2. Netzwerkkarte (SER): 192.168.1.1
IP des SIP-Proxys vom Carrier: 192.168.1.2


Der Asterisk läuft nur auf 192.168.0.1:5060 und SER auf 192.168.1.1:5060. Der SIP-Proxy des Carrier akzeptiert alle Gespräche von 192.168.1.1:5060 und leitet alle ankommenden Rufe
an 192.168.1.1:5060.

Gespräche vom Festnetz auf eine SIP-Rufnummer funktioniert. Der Anruf wird korrekt weitergeleitet (Proxy --> SER --> Asterisk --> VoIP-Endgerät).

Gespräche von einem VoIP-Endgerät ins Festnetz funktionieren nicht. Es sieht so aus, wie wenn der SER Proxy die INVITE Anfragen nicht an die 192.168.1.2 weiterleitet sondern wieder an den Asterisk zurückschickt...

Code: 
asterisk-log:

    -- Executing GotoIf("SIP/+4930123456789-9f9d", "0?2:4") in new stack
    -- Goto (default,050123456789,4)
    -- Executing GotoIf("SIP/+4930123456789-9f9d", "1?5:7") in new stack
    -- Goto (default,050123456789,5)
    -- Executing Dial("SIP/+4930123456789-9f9d", "SIP/[email protected]") in new stack
    -- Called [email protected]
    -- Got SIP response 482 "Loop Detected" back from 192.168.1.1
    -- Now forwarding SIP/+4930123456789-9f9d to 'Local/+4950123456789@default' (thanks to SIP/192.168.1.1-f880)
    -- Executing Dial("Local/+4950123456789@default-239e,2", "SIP/[email protected]") in new stack
    -- Called [email protected]
    -- Got SIP response 482 "Loop Detected" back from 192.168.1.1
    -- Now forwarding Local/+4950123456789@default-239e,2 to 'Local/+4950123456789@default' (thanks to SIP/192.168.1.1-b736)
    -- Executing Dial("Local/+4950123456789@default-99bd,2", "SIP/[email protected]") in new stack
    -- Called [email protected]
    -- Got SIP response 482 "Loop Detected" back from 192.168.1.1
    .......



ngrep auf 2. Netzwerkkarte:

#
U 192.168.1.1:5060 -> 192.168.1.2:5060
  INVITE sip:[email protected]:5060 SIP/2.0..Max-Forwards: 10..Record-Route: <sip:192.168.1.1;ftag=as5da54c15;lr=on>..Via: SIP/2.0/UDP 192.168.1.1;branch=0..Via: SIP/2.0/UDP 213.146.230.50:5060;branch=z9hG4bK7b301edf..Fro
  m: "+4930123456789" <sip:[email protected]>;tag=as5da54c15..To: <sip:[email protected]>..Contact: <sip:[email protected]>..Call-ID: [email][email protected][/email]: 102 INVITE..User-
  Agent: Asterisk PBX..Date: Fri, 11 Nov 2005 15:40:12 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY..Content-Type: application/sdp..Content-Length: 267....v=0..o=root 29657 29657 IN IP4 213.146.230.50..s=session..c=IN I
  P4 213.146.230.50..t=0 0..m=audio 14138 RTP/AVP 0 3 8 101..a=rtpmap:0 PCMU/8000..a=rtpmap:3 GSM/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..
#
U 192.168.1.2:5060 -> 192.168.1.1:5060
  SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP 192.168.1.1;branch=0..Via: SIP/2.0/UDP 213.146.230.50:5060;branch=z9hG4bK7b301edf..From: "+4930123456789" <sip:[email protected]>;tag=as5da54c15..To: <si
  p:[email protected]>..Call-ID: [email][email protected][/email]: 102 INVITE..Content-Length: 0....
#
U 192.168.1.2:5060 -> 192.168.1.1:5060
  INVITE sip:[email protected]:5060 SIP/2.0..Record-Route: <sip:[email protected];ftag=as5da54c15;lr=on>..Max-Forwards:  9..Record-Route: <sip:192.168.1.1;ftag=as5da54c15;lr=on>..Via: SIP/2.0/UDP 192.168.1.2;b
  ranch=z9hG4bKe44b.50306125.0..Via: SIP/2.0/UDP 192.168.1.1;branch=0..Via: SIP/2.0/UDP 213.146.230.50:5060;branch=z9hG4bK7b301edf..From: "+4930123456789" <sip:[email protected]>;tag=as5da54c15..To: <sip:[email protected]
  6.231.2>..Contact: <sip:[email protected]>..Call-ID: [email][email protected][/email]: 102 INVITE..User-Agent: Asterisk PBX..Date: Fri, 11 Nov 2005 15:40:12 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE
  , REFER, NOTIFY..Content-Type: application/sdp..Content-Length: 267....v=0..o=root 29657 29657 IN IP4 213.146.230.50..s=session..c=IN IP4 213.146.230.50..t=0 0..m=audio 14138 RTP/AVP 0 3 8 101..a=rtpmap:0 PCMU/8000..a=rtpmap:3 GSM/80
  00..a=rtpmap:8 PCMA/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..
#
U 192.168.1.1:5060 -> 192.168.1.2:5060
  SIP/2.0 482 Loop Detected..Via: SIP/2.0/UDP 192.168.1.2;branch=z9hG4bKe44b.50306125.0..Via: SIP/2.0/UDP 192.168.1.1;branch=0..Via: SIP/2.0/UDP 213.146.230.50:5060;branch=z9hG4bK7b301edf..From: "+4930123456789" <sip:+4930123456789@
  192.168.1.1>;tag=as5da54c15..To: <sip:[email protected]>;tag=as5da54c15..Call-ID: [email][email protected][/email]: 102 INVITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
  , NOTIFY..Contact: <sip:[email protected]>..Content-Length: 0..X-Asterisk-HangupCause:: Normal Clearing....
#
U 192.168.1.2:5060 -> 192.168.1.1:5060
  ACK sip:[email protected]:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.1.2;branch=z9hG4bKe44b.50306125.0..From: "+4930123456789" <sip:[email protected]>;tag=as5da54c15..Call-ID: [email protected]
  6.231.2..To: <sip:[email protected]>;tag=as5da54c15..CSeq: 102 ACK..Content-Length: 0....
#
U 192.168.1.2:5060 -> 192.168.1.1:5060
  SIP/2.0 482 Loop Detected..Via: SIP/2.0/UDP 192.168.1.1;branch=0..Via: SIP/2.0/UDP 213.146.230.50:5060;branch=z9hG4bK7b301edf..From: "+4930123456789" <sip:[email protected]>;tag=as5da54c15..To: <sip:[email protected]
  31.2>;tag=as5da54c15..Call-ID: [email][email protected][/email]: 102 INVITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY..Contact: <sip:[email protected]>..Content-
  Length: 0..X-Asterisk-HangupCause:: Normal Clearing....
#
U 192.168.1.1:5060 -> 192.168.1.2:5060
  ACK sip:[email protected]:5060 SIP/2.0..Max-Forwards: 10..Record-Route: <sip:192.168.1.1;ftag=as5da54c15;lr=on>..Via: SIP/2.0/UDP 192.168.1.1;branch=0..Via: SIP/2.0/UDP 213.146.230.50:5060;branch=z9hG4bK7b301edf..From:
  "+4930123456789" <sip:[email protected]>;tag=as5da54c15..To: <sip:[email protected]>;tag=as5da54c15..Contact: <sip:[email protected]>..Call-ID: [email][email protected][/email]: 102 A
  CK..User-Agent: Asterisk PBX..Content-Length: 0....
#
U 192.168.1.1:5060 -> 192.168.1.2:5060
  INVITE sip:[email protected]:5060 SIP/2.0..Max-Forwards: 10..Record-Route: <sip:192.168.1.1;ftag=as051dada6;lr=on>..Via: SIP/2.0/UDP 192.168.1.1;branch=0..Via: SIP/2.0/UDP 213.146.230.50:5060;branch=z9hG4bK7519deee..Fro
  m: "+4930123456789" <sip:[email protected]>;tag=as051dada6..To: <sip:[email protected]>..Contact: <sip:[email protected]>..Call-ID: [email][email protected][/email]: 102 INVITE..User-
  Agent: Asterisk PBX..Date: Fri, 11 Nov 2005 15:40:12 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY..Content-Type: application/sdp..Content-Length: 267....v=0..o=root 29658 29658 IN IP4 213.146.230.50..s=session..c=IN I
  P4 213.146.230.50..t=0 0..m=audio 18576 RTP/AVP 0 3 8 101..a=rtpmap:0 PCMU/8000..a=rtpmap:3 GSM/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..
#
U 192.168.1.2:5060 -> 192.168.1.1:5060
  SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP 192.168.1.1;branch=0..Via: SIP/2.0/UDP 213.146.230.50:5060;branch=z9hG4bK7519deee..From: "+4930123456789" <sip:[email protected]>;tag=as051dada6..To: <si
  p:[email protected]>..Call-ID: [email][email protected][/email]: 102 INVITE..Content-Length: 0....
#
U 192.168.1.2:5060 -> 192.168.1.1:5060
  INVITE sip:[email protected]:5060 SIP/2.0..Record-Route: <sip:[email protected];ftag=as051dada6;lr=on>..Max-Forwards:  9..Record-Route: <sip:192.168.1.1;ftag=as051dada6;lr=on>..Via: SIP/2.0/UDP 192.168.1.2;b
  ranch=z9hG4bK7079.a3655fa7.0..Via: SIP/2.0/UDP 192.168.1.1;branch=0..Via: SIP/2.0/UDP 213.146.230.50:5060;branch=z9hG4bK7519deee..From: "+4930123456789" <sip:[email protected]>;tag=as051dada6..To: <sip:[email protected]
  6.231.2>..Contact: <sip:[email protected]>..Call-ID: [email][email protected][/email]: 102 INVITE..User-Agent: Asterisk PBX..Date: Fri, 11 Nov 2005 15:40:12 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE
  , REFER, NOTIFY..Content-Type: application/sdp..Content-Length: 267....v=0..o=root 29658 29658 IN IP4 213.146.230.50..s=session..c=IN IP4 213.146.230.50..t=0 0..m=audio 18576 RTP/AVP 0 3 8 101..a=rtpmap:0 PCMU/8000..a=rtpmap:3 GSM/80
  00..a=rtpmap:8 PCMA/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..
#
U 192.168.1.1:5060 -> 192.168.1.2:5060
  SIP/2.0 482 Loop Detected..Via: SIP/2.0/UDP 192.168.1.2;branch=z9hG4bK7079.a3655fa7.0..Via: SIP/2.0/UDP 192.168.1.1;branch=0..Via: SIP/2.0/UDP 213.146.230.50:5060;branch=z9hG4bK7519deee..From: "+4930123456789" <sip:+4930123456789@
  192.168.1.1>;tag=as051dada6..To: <sip:[email protected]>;tag=as051dada6..Call-ID: [email][email protected][/email]: 102 INVITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
  , NOTIFY..Contact: <sip:[email protected]>..Content-Length: 0..X-Asterisk-HangupCause:: Normal Clearing....
#
U 192.168.1.2:5060 -> 192.168.1.1:5060
  ACK sip:[email protected]:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.1.2;branch=z9hG4bK7079.a3655fa7.0..From: "+4930123456789" <sip:[email protected]>;tag=as051dada6..Call-ID: [email protected]
  6.231.2..To: <sip:[email protected]>;tag=as051dada6..CSeq: 102 ACK..Content-Length: 0....
#
U 192.168.1.2:5060 -> 192.168.1.1:5060
  SIP/2.0 482 Loop Detected..Via: SIP/2.0/UDP 192.168.1.1;branch=0..Via: SIP/2.0/UDP 213.146.230.50:5060;branch=z9hG4bK7519deee..From: "+4930123456789" <sip:[email protected]>;tag=as051dada6..To: <sip:[email protected]
  31.2>;tag=as051dada6..Call-ID: [email][email protected][/email]: 102 INVITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY..Contact: <sip:[email protected]>..Content-
  Length: 0..X-Asterisk-HangupCause:: Normal Clearing....
#
U 192.168.1.1:5060 -> 192.168.1.2:5060
  ACK sip:[email protected]:5060 SIP/2.0..Max-Forwards: 10..Record-Route: <sip:192.168.1.1;ftag=as051dada6;lr=on>..Via: SIP/2.0/UDP 192.168.1.1;branch=0..Via: SIP/2.0/UDP 213.146.230.50:5060;branch=z9hG4bK7519deee..From:
  "+4930123456789" <sip:[email protected]>;tag=as051dada6..To: <sip:[email protected]>;tag=as051dada6..Contact: <sip:[email protected]>..Call-ID: [email][email protected][/email]: 102 A
  CK..User-Agent: Asterisk PBX..Content-Length: 0....
#
.....


ngrep auf 1. Netzwerkkarte:

U 192.168.0.3:5060 -> 192.168.0.1:5060
  INVITE sip:[email protected];user=phone SIP/2.0..Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bK14dd21ae9e1f3539..From: "+4930123456789" <sip:[email protected];user=phone>;tag=0598e21a55e999a9..To: <sip:050123456789@
  192.168.0.1;user=phone>..Contact: <sip:[email protected];user=phone>..Supported: replaces, timer..Call-ID: [email][email protected][/email]: 37544 INVITE..User-Agent: Grandstream GXP2000 1.0.1.8..Max-Forwards: 70..
  Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK..Content-Type: application/sdp..Content-Length: 192....v=0..o=+4930123456789 8000 8000 IN IP4 192.168.0.3..s=SIP Call..c=IN IP4 192.168.0.3..t=0 0..m=
  audio 5004 RTP/AVP 0 8..a=sendrecv..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=ptime:20..
#
U 192.168.0.1:5060 -> 192.168.0.3:5060
  SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bK14dd21ae9e1f3539..From: "+4930123456789" <sip:[email protected];user=phone>;tag=0598e21a55e999a9..To: <sip:[email protected];
  user=phone>;tag=as65ef3a34..Call-ID: [email][email protected][/email]: 37544 INVITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY..Contact: <sip:[email protected]>..Proxy-Authenticat
  e: Digest realm="asterisk", nonce="38500f22" ..Content-Length: 0....
#
U 192.168.0.3:5060 -> 192.168.0.1:5060
  ACK sip:[email protected];user=phone SIP/2.0..Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bK14dd21ae9e1f3539..From: "+4930123456789" <sip:[email protected];user=phone>;tag=0598e21a55e999a9..To: <sip:050123456789@213
  .146.230.50;user=phone>;tag=as65ef3a34..Contact: <sip:[email protected];user=phone>..Call-ID: [email][email protected][/email]: 37544 ACK..User-Agent: Grandstream GXP2000 1.0.1.8..Max-Forwards: 70..Allow: INVITE,ACK,C
  ANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK..Content-Length: 0....
#
U 192.168.0.3:5060 -> 192.168.0.1:5060
  INVITE sip:[email protected];user=phone SIP/2.0..Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKb5b641f12d2acb9c..From: "+4930123456789" <sip:[email protected];user=phone>;tag=0598e21a55e999a9..To: <sip:050123456789@
  192.168.0.1;user=phone>..Contact: <sip:[email protected];user=phone>..Supported: replaces, timer..Proxy-Authorization: Digest username="+4930123456789", realm="asterisk", algorithm=MD5, uri="sip:[email protected]
  0;user=phone", nonce="38500f22", response="eab48deaf044a2aceb93e3d53062d34e"..Call-ID: [email][email protected][/email]: 37545 INVITE..User-Agent: Grandstream GXP2000 1.0.1.8..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIF
  Y,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK..Content-Type: application/sdp..Content-Length: 192....v=0..o=+4930123456789 8000 8001 IN IP4 192.168.0.3..s=SIP Call..c=IN IP4 192.168.0.3..t=0 0..m=audio 5004 RTP/AVP 0 8..a=sendrecv
  ..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=ptime:20..
#
U 192.168.0.1:5060 -> 192.168.0.3:5060
  SIP/2.0 100 Trying..Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKb5b641f12d2acb9c..From: "+4930123456789" <sip:[email protected];user=phone>;tag=0598e21a55e999a9..To: <sip:[email protected];user=phone>..Call-ID: 8
  [email][email protected][/email]: 37545 INVITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY..Contact: <sip:[email protected]>..Content-Length: 0....
#
.....



ser.cfg
=======

check_via=no
dns=no
rev_dns=no
listen=192.168.1.1
alias="sip.domainname.de"
alias="192.168.1.1"
fifo="/tmp/ser_fifo"
loadmodule "/usr/lib/ser/modules/sl.so"
loadmodule "/usr/lib/ser/modules/tm.so"
loadmodule "/usr/lib/ser/modules/rr.so"
loadmodule "/usr/lib/ser/modules/maxfwd.so"
loadmodule "/usr/lib/ser/modules/usrloc.so"
loadmodule "/usr/lib/ser/modules/registrar.so"
loadmodule "/usr/lib/ser/modules/exec.so"
modparam("usrloc", "db_mode",   0)
modparam("rr", "enable_full_lr", 1)
route{
        if (!mf_process_maxfwd_header("10")) {
                sl_send_reply("483","Too Many Hops");
                break;
        };
        if ( msg:len > max_len ) {
                sl_send_reply("513", "Message too big");
                break;
        };
        record_route();
        if (loose_route()) {
                t_relay();
                break;
        };


        if (src_ip==192.168.1.2)
                {
                route(1);
                break;
                }

        if (!src_ip==192.168.1.2){
                route(2);
                break;
                }

        # send not found error for any other URI
        sl_send_reply( "404", "Not Found" );

}

route[1]
        {
        if (uri=~"sip:+494012345678[0-9]@(sip\.domainname\.de)|(192\.168\.1\.1)" )
                {
                forward( 192.168.0.1, 5060 );
                break;
                };
        if (uri=~"sip:+493012345678[0-9]@(sip\.domainname\.de)|(192\.168\.1\.1)" )
                {
                forward( 192.168.0.1, 5060 );
                break;
                };
        }

route[2]
        {
        forward( 192.168.1.2, 5060 );
        }









extensions.conf
===============

[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no

[default]
include => inbound-telephony
include => outbound-telephony

[inbound-telephony]
exten=>+4930123456780,1,Dial(SIP/+4930123456781,120)
exten=>+4930123456780,2,Hangup
exten=>+4930123456781,1,Dial(SIP/+4930123456781,120)
exten=>+4930123456781,2,Hangup
exten=>+4930123456782,1,Dial(SIP/+4930123456782,120)
exten=>+4930123456782,2,Hangup
exten=>+4930123456783,1,Dial(SIP/+4930123456783,120)
exten=>+4930123456783,2,Hangup
exten=>+4930123456784,1,Dial(SIP/+4930123456784,120)
exten=>+4930123456784,2,Hangup
exten=>+4930123456785,1,Dial(SIP/+4930123456785,120)
exten=>+4930123456785,2,Hangup
exten=>+4930123456786,1,Dial(SIP/+4930123456786,120)
exten=>+4930123456786,2,Hangup
exten=>+4930123456787,1,Dial(SIP/+4930123456787,120)
exten=>+4930123456787,2,Hangup
exten=>+4930123456788,1,Dial(SIP/+4930123456788,120)
exten=>+4930123456788,2,Hangup
exten=>+4930123456789,1,Dial(SIP/+4930123456789,120)
exten=>+4930123456789,2,Hangup
exten=>+4940123456780,1,Dial(SIP/+4940123456780,120)
exten=>+4940123456780,2,Hangup
exten=>+4940123456781,1,Dial(SIP/+4940123456781,120)
exten=>+4940123456781,2,Hangup
exten=>+4940123456782,1,Dial(SIP/+4940123456782,120)
exten=>+4940123456782,2,Hangup
exten=>+4940123456783,1,Dial(SIP/+4940123456783,120)
exten=>+4940123456783,2,Hangup
exten=>+4940123456784,1,Dial(SIP/+4940123456784,120)
exten=>+4940123456784,2,Hangup
exten=>+4940123456785,1,Dial(SIP/+4940123456785,120)
exten=>+4940123456785,2,Hangup
exten=>+4940123456786,1,Dial(SIP/+4940123456786,120)
exten=>+4940123456786,2,Hangup
exten=>+4940123456787,1,Dial(SIP/+4940123456787,120)
exten=>+4940123456787,2,Hangup
exten=>+4940123456788,1,Dial(SIP/+4940123456788,120)
exten=>+4940123456788,2,Hangup
exten=>+4940123456789,1,Dial(SIP/+4940123456789,120)
exten=>+4940123456789,2,Hangup

[outbound-telephony]
;;; Rufnummer unterdruecken
;   exten => _.,1,SetCallerID( <>)
;;; Rufnummer anzeigen
;   exten => _.,1,SetCallerID(+4930123456780 <+4930123456780>)
;   exten => _.,2,Dial(SIP/${EXTEN}@192.168.1.2)
;   exten => _.,3,Hangup

exten => _+.,1,Dial(SIP/${EXTEN}@192.168.1.2)
exten => _+.,2,Hangup
exten => _X.,1,GotoIF($["${EXTEN:0:2}" = "00"]?2:4)
exten => _X.,2,Dial(SIP/+${EXTEN:2}@192.168.1.2)
exten => _X.,3,Goto(7)
exten => _X.,4,GotoIF($["${EXTEN:0:1}" = "0"]?5:7)
exten => _X.,5,Dial(SIP/+49${EXTEN:1}@192.168.1.2)
exten => _X.,6,Goto(10)
exten => _X.,7,Hangup






sip.conf
========

[general]
context=default
allowguest=yes
bindport=5060
bindaddr=213.146.230.50
language=de
dtmfmode = rfc2833
autocreatepeer=yes
canreinvite=no

[192.168.1.1]
type=peer       
context=default
fromdomain=192.168.1.1
host=192.168.1.1
qualify=yes
insecure=very

[+4930123456780]
type=friend
context=default
username=+4930123456780
secret=+4930123456780
host=dynamic

[+4930123456781]
type=friend
context=default
username=+4930123456781
secret=+4930123456781
host=dynamic

[+4930123456782]
type=friend
context=default
username=+4930123456782
secret=+4930123456782
host=dynamic

[+4930123456783]
type=friend
context=default
username=+4930123456783
secret=+4930123456783
host=dynamic

[+4930123456784]
type=friend
context=default
username=+4930123456784
secret=+4930123456784
host=dynamic

[+4930123456785]
type=friend
context=default
username=+4930123456785
secret=+4930123456785
host=dynamic

[+4930123456786]
type=friend
context=default
username=+4930123456786
secret=+4930123456786
host=dynamic

[+4930123456787]
type=friend
context=default
username=+4930123456787
secret=+4930123456787
host=dynamic

[+4930123456788]
type=friend
context=default
username=+4930123456788
secret=+4930123456788
host=dynamic

[+4930123456789]
type=friend
context=default
username=+4930123456789
secret=+4930123456789
host=dynamic

[+4940123456780]
type=friend
context=default
username=+4940123456780
secret=+4940123456780
host=dynamic

[+4940123456781]
type=friend
context=default
username=+4940123456781
secret=+4940123456781
host=dynamic

[+4940123456782]
type=friend
context=default
username=+4940123456782
secret=+4940123456782
host=dynamic

[+4940123456783]
type=friend
context=default
username=+4940123456783
secret=+4940123456783
host=dynamic

[+4940123456784]
type=friend
context=default
username=+4940123456784
secret=+4940123456784
host=dynamic

[+4940123456785]
type=friend
context=default
username=+4940123456785
secret=+4940123456785
host=dynamic

[+4940123456786]
type=friend
context=default
username=+4940123456786
secret=+4940123456786
host=dynamic

[+4940123456787]
type=friend
context=default
username=+4940123456787
secret=+4940123456787
host=dynamic

[+4940123456788]
type=friend
context=default
username=+4940123456788
secret=+4940123456788
host=dynamic

Ich sehr dankbar, wenn ich Lösungen für das Problem bekommen würde :-)

Gruß,

Abarta
 
Problem behoben?

Hallo

hast du dein Problem schon behoben oder steht es noch an?!?

mfg
Rolf
 
Status
Für weitere Antworten geschlossen.
Kostenlos!
Jetzt holen

Neueste Beiträge

Kategorien

Wissenswertes

Statistik des Forums

Themen
248,886
Beiträge
2,303,960
Mitglieder
378,564
Neuestes Mitglied
warumdas

3CX jetzt einen Monat kostenlos testen

Die Verbindung, Zusammenarbeit und Kommunikation mit Ihrem Team und Ihren Kunden war noch nie so einfach. Nutzen Sie unsere kostenlose 30-Tage-Testversion und entdecken Sie die richtige Lösung für Ihr Unternehmen.

3CX start your free trial guy
Für diese E-Mail-Adresse besteht bereits ein 3CX-Konto. Sie werden zum Kundenportal weitergeleitet, wo Sie sich anmelden oder Ihr Passwort zurücksetzen können, falls Sie dieses vergessen haben.
| Style by ThemeHouse
XenPorta 2 PRO © Jason Axelrod of 8WAYRUN
Oben Unten