[Problem] Asterisk an bestehende VOIP Telefonanlage

[Frankyi]

Hallo zusammen,

ich habe folgendes Problem.

Wir haben bereits eine bestehende VOIP Telefonanlage mit 30 Teilnehmern.
Da wir diese zwecks Handyanbindung (Voip Client) nicht direkt in das Internet stellen wollen, wollte ich eine Asterisk davor schalten.

Im Moment sieht es folgendermaßen aus.

sip.conf

[general]
context=default                 ; Default context for incoming calls
allowoverlap=no                 ; Disable overlap dialing support. (Default is yes)
bindport=5060                   ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=188.65.206.9                ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
                                ; Note: Asterisk only uses the first host

register => 513:[email protected]/513


[513]
type=peer
username=513
secret=513
fromuser=513
host=dynamic
nat=yes
insecure=very
canreinvite=no
qualify=yes
dtmfmode=inband
allow=alaw
call-limit=4
context=513

Die 513 ist die Rufnummer bzw. auch das Kennwort in unserer bestehenden Voip Anlage (Tevitel).

Meine extensions.conf sieht folgendermaßen aus.

[513]
exten => _X.,1,Dial(SIP/01${EXTEN}@513,60,r)

Das Problem ist nun, das ich mein Handy mit dem Eintrag host=dynamic anmelden kann, jedoch nicht extern Telefonieren.
Ist mein Handy nun also mit host=dynamic mit der TK Anlage verbunden, und ich wechsele host=10.221.253.6 (IP unserer TK Anlage) funktioniert das externe rausrufen. ABER nur für ca 20 Sekunden, danach kommt Benutzername Kennwort falsch.

Hat jemand eine Idee?, bzw. was für Infos benötigt ihr?

 

[wildzero]

host=dynamic lassen und den Dial so:
exten => _X.,1,Dial(SIP/01${EXTEN}@10.221.253.6,60,r)

SIP trace anfügen ist auch nie falsch.

 

[Frankyi]

Habe ich angepasst. Funktioniert leider nicht. Ich höre in der Voip App ein Freizeichen, aber unser Monitor der TK Anlage zeigt kein Gespräch an (Es klingelt auch nicht).
Nochmal zur übersicht.

Ich habe eine bestehende Telefonanlage mit der IP 10.221.253.6
Habe Voip Agenten
513
512
511
.
.

Dort ist jeweils Rufnummer=Kennwort.
Nun habe ich eine Asterisk davor geschaltet mit der IP 188.65.206.9 (beide Netze sind verbunden).
Hier ist nun das Sip Trace wenn ich versuche, mit dem Account 513 nach außen zu telefonieren.
(Habe einfach mal die 021611858599 angerufen)

<--- SIP read from UDP:188.65.200.5:12596 --->
INVITE sip:[email protected] SIP/2.0
Accept: application/reginfo+xml, application/sdp, application/simple-message-summary, message/sipfrag, multipart/mixed, multipart/related
Via: SIP/2.0/UDP 188.65.200.5:18040;branch=z9hG4bK0526ec32e7b4a8a78
Max-Forwards: 70
From: "513" <sip:[email protected]:5060>;tag=9fa58f6e18
To: <sip:[email protected]>
Call-ID: d68d7c8161173fee
CSeq: 946047181 INVITE
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, SUBSCRIBE, UPDATE
Allow-Events: refer
Contact: <sip:[email protected]:18040>
Supported: replaces
User-Agent: Media5-fone/3.7.1.1118
Content-Disposition: session
Content-Type: application/sdp
Content-Length: 294

v=0
o=MxSIP 2274883285 2274883286 IN IP4 188.65.200.5
s=SIP Call
c=IN IP4 188.65.200.5
t=0 0
a=sendrecv
m=audio 64920 RTP/AVP 0 8 96 125
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 iLBC/8000
a=rtpmap:125 telephone-event/8000
a=fmtp:96 mode=30
a=fmtp:125 0-15
a=sendrecv
<------------->
--- (16 headers 14 lines) ---
Sending to 188.65.200.5:12596 (NAT)
Using INVITE request as basis request - d68d7c8161173fee
Found peer '513' for '513' from 188.65.200.5:12596

<--- Reliably Transmitting (NAT) to 188.65.200.5:12596 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 188.65.200.5:18040;branch=z9hG4bK0526ec32e7b4a8a78;received=188.65.200.5;rport=12596
From: "513" <sip:[email protected]:5060>;tag=9fa58f6e18
To: <sip:[email protected]>;tag=as1e5256e1
Call-ID: d68d7c8161173fee
CSeq: 946047181 INVITE
Server: Asterisk PBX 1.8.13.1~dfsg-3ubuntu3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7c3e9d38"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'd68d7c8161173fee' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:188.65.200.5:12596 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 188.65.200.5:18040;branch=z9hG4bK0526ec32e7b4a8a78
Max-Forwards: 70
From: "513" <sip:[email protected]:5060>;tag=9fa58f6e18
To: <sip:[email protected]>;tag=as1e5256e1
Call-ID: d68d7c8161173fee
CSeq: 946047181 ACK
User-Agent: Media5-fone/3.7.1.1118
Content-Length: 0

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

<--- SIP read from UDP:188.65.200.5:12596 --->
INVITE sip:[email protected] SIP/2.0
Accept: application/reginfo+xml, application/sdp, application/simple-message-summary, message/sipfrag, multipart/mixed, multipart/related
Via: SIP/2.0/UDP 188.65.200.5:18040;branch=z9hG4bKe0242452b6cbeed5e
Max-Forwards: 70
From: "513" <sip:[email protected]:5060>;tag=9fa58f6e18
To: <sip:[email protected]>
Call-ID: d68d7c8161173fee
CSeq: 946047182 INVITE
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, SUBSCRIBE, UPDATE
Allow-Events: refer
Authorization: Digest username="513",realm="asterisk",nonce="7c3e9d38",uri="sip:[email protected]",response="37753d4c79c13049ed43a82eb7e6cd95",algorithm=MD5
Contact: <sip:[email protected]:18040>
Supported: replaces
User-Agent: Media5-fone/3.7.1.1118
Content-Disposition: session
Content-Type: application/sdp
Content-Length: 294

v=0
o=MxSIP 2274883285 2274883286 IN IP4 188.65.200.5
s=SIP Call
c=IN IP4 188.65.200.5
t=0 0
a=sendrecv
m=audio 64920 RTP/AVP 0 8 96 125
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 iLBC/8000
a=rtpmap:125 telephone-event/8000
a=fmtp:96 mode=30
a=fmtp:125 0-15
a=sendrecv
<------------->
--- (17 headers 14 lines) ---
Sending to 188.65.200.5:12596 (NAT)
Using INVITE request as basis request - d68d7c8161173fee
Found peer '513' for '513' from 188.65.200.5:12596
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 96
Found RTP audio format 125
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format iLBC for ID 96
Found audio description format telephone-event for ID 125
Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x40c (ulaw|alaw|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event|), combined - 0x0 (nothing)
Peer audio RTP is at port 188.65.200.5:64920
Looking for 021611858599 in 513 (domain 188.65.206.9)
list_route: hop: <sip:[email protected]:18040>

<--- Transmitting (NAT) to 188.65.200.5:12596 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 188.65.200.5:18040;branch=z9hG4bKe0242452b6cbeed5e;received=188.65.200.5;rport=12596
From: "513" <sip:[email protected]:5060>;tag=9fa58f6e18
To: <sip:[email protected]>
Call-ID: d68d7c8161173fee
CSeq: 946047182 INVITE
Server: Asterisk PBX 1.8.13.1~dfsg-3ubuntu3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0


<------------>
Audio is at 12400
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x800000000000 (testlaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 10.221.253.6:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 188.65.206.9:5060;branch=z9hG4bK22146f75;rport
Max-Forwards: 70
From: "513" <sip:[email protected]>;tag=as681b2126
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.13.1~dfsg-3ubuntu3
Date: Mon, 20 Jan 2014 13:31:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 298

v=0
o=root 1782897719 1782897719 IN IP4 188.65.206.9
s=Asterisk PBX 1.8.13.1~dfsg-3ubuntu3
c=IN IP4 188.65.206.9
t=0 0
m=audio 12400 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- Transmitting (NAT) to 188.65.200.5:12596 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 188.65.200.5:18040;branch=z9hG4bKe0242452b6cbeed5e;received=188.65.200.5;rport=12596
From: "513" <sip:[email protected]:5060>;tag=9fa58f6e18
To: <sip:[email protected]>;tag=as40f0bc40
Call-ID: d68d7c8161173fee
CSeq: 946047182 INVITE
Server: Asterisk PBX 1.8.13.1~dfsg-3ubuntu3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0


<------------>

<--- SIP read from UDP:10.221.253.6:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 188.65.206.9:5060;branch=z9hG4bK22146f75;rport=5060
From: "513" <sip:[email protected]>;tag=as681b2126
To: <sip:[email protected]>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Server: SER at EcotelVoIP
Content-Length: 0
Warning: 392 10.221.253.6:5060 "Noisy feedback tells: pid=1247 req_src_ip=188.65.206.9 req_src_port=5060 in_uri=sip:[email protected] out_uri=sip:[email protected] via_cnt==1"

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

<--- SIP read from UDP:10.221.253.6:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 188.65.206.9:5060;branch=z9hG4bK22146f75;rport=5060
From: "513" <sip:[email protected]>;tag=as681b2126
To: <sip:[email protected]>;tag=0d2f260f9d186a09b7d5a78d9751033b.e544
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm="188.65.206.9", nonce="52dd266b70d48df32316b059aed56e53b6482bc5"
Server: SER at EcotelVoIP
Content-Length: 0
Warning: 392 10.221.253.6:5060 "Noisy feedback tells: pid=1247 req_src_ip=188.65.206.9 req_src_port=5060 in_uri=sip:[email protected] out_uri=sip:[email protected] via_cnt==1"

<------------->
--- (10 headers 0 lines) ---
set_destination: Parsing <sip:[email protected]> for address/port to send to
set_destination: set destination to 10.221.253.6:5060
Transmitting (NAT) to 10.221.253.6:5060:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 188.65.206.9:5060;branch=z9hG4bK22146f75;rport
Max-Forwards: 70
From: "513" <sip:[email protected]>;tag=as681b2126
To: <sip:[email protected]>;tag=0d2f260f9d186a09b7d5a78d9751033b.e544
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.13.1~dfsg-3ubuntu3
Content-Length: 0


---
[Jan 20 14:31:56] NOTICE[905]: chan_sip.c:20354 handle_response_invite: Failed to authenticate on INVITE to '"513" <sip:[email protected]>;tag=as681b2126'
Really destroying SIP dialog '[email protected]:5060' Method: INVITE

<--- SIP read from UDP:188.65.200.5:12596 --->
CANCEL sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 188.65.200.5:18040;branch=z9hG4bKe0242452b6cbeed5e
Max-Forwards: 70
From: "513" <sip:[email protected]:5060>;tag=9fa58f6e18
To: <sip:[email protected]>
Call-ID: d68d7c8161173fee
CSeq: 946047182 CANCEL
User-Agent: Media5-fone/3.7.1.1118
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 188.65.200.5:12596 (NAT)

<--- Reliably Transmitting (NAT) to 188.65.200.5:12596 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 188.65.200.5:18040;branch=z9hG4bKe0242452b6cbeed5e;received=188.65.200.5;rport=12596
From: "513" <sip:[email protected]:5060>;tag=9fa58f6e18
To: <sip:[email protected]>;tag=as40f0bc40
Call-ID: d68d7c8161173fee
CSeq: 946047182 INVITE
Server: Asterisk PBX 1.8.13.1~dfsg-3ubuntu3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- Transmitting (NAT) to 188.65.200.5:12596 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 188.65.200.5:18040;branch=z9hG4bKe0242452b6cbeed5e;received=188.65.200.5;rport=12596
From: "513" <sip:[email protected]:5060>;tag=9fa58f6e18
To: <sip:[email protected]>;tag=as40f0bc40
Call-ID: d68d7c8161173fee
CSeq: 946047182 CANCEL
Server: Asterisk PBX 1.8.13.1~dfsg-3ubuntu3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[Jan 20 14:32:01] WARNING[2677]: pbx.c:5203 __ast_pbx_run: Don't know what to do with 'SIP/513-00000020'

<--- SIP read from UDP:188.65.200.5:12596 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 188.65.200.5:18040;branch=z9hG4bKe0242452b6cbeed5e
Max-Forwards: 70
From: "513" <sip:[email protected]:5060>;tag=9fa58f6e18
To: <sip:[email protected]>;tag=as40f0bc40
Call-ID: d68d7c8161173fee
CSeq: 946047182 ACK
User-Agent: Media5-fone/3.7.1.1118
Content-Length: 0

<------------->