[Problem] kann keine Verbindung asterisk 1.8.6 und sipgate herstellen.

tameritoke

Neuer User
Mitglied seit
31 Jan 2009
Beiträge
16
Punkte für Reaktionen
0
Punkte
0
Hi!
Ich schaffe es nicht, mit sipgate erfolgreich eine Verbindung zu einem anderem Teilnehmer aufzubauen.

Die Linux Maschine baut die Verbindung selbst per PPPoE auf, also es befindet sich auch keine Firewall oder sonstiger NAT Router dazwischen.

Ich komme jedenfalls nicht dahinter, wie ich dieses Problem gelöst kriege. Unter Asterisk 1.6.0 lief es, nur stehe ich jetzt total auf dem Schlauch.

Für jede Hilfe bin ich dankbar.

sip.conf:
Code:
[general]
port=5060
bindaddr=0.0.0.0
language=de
dtmfmode=rfc2833
srvlookup=yes
disallow=all
allow=alaw
allow=ulaw
nat=no

register = 334xxxxxxx:[email protected]/2000

[sipgate]
type=friend
insecure=invite
username=334xxxxxxx
fromuser=334xxxxxxx
fromdomain=sipgate.de
host=sipgate.de
secret=xxxxxxx
host=sipgate.de
qualify=yes
caninvite=no
canreinvite=no
disallow=all
allow=alaw
allow=ulaw
allow=speex
allow=g729
dtmfmode=rfc2833
context=tamertelein

und der sip.debug:

Code:
office*CLI> sip set debug on
SIP Debugging re-enabled
    -- Accepting overlap call from '73585120' to '0151xxxxxxxx' on channel 0/2, span 2
    -- Starting simple switch on 'DAHDI/i2/73585120-4'
[Sep  8 23:28:26] NOTICE[17120]: chan_sip.c:12593 sip_reregister:    -- Re-registration for  [email protected]
       > doing dnsmgr_lookup for 'sipgate.de'
       > ast_get_srv: SRV lookup for '_sip._udp.sipgate.de' mapped to host sipgate.de, port 5060
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 217.10.79.9:5060:
REGISTER sip:sipgate.de SIP/2.0
Via: SIP/2.0/UDP 78.51.16.192:5060;branch=z9hG4bK2fec2dd6
Max-Forwards: 70
From: <sip:[email protected]>;tag=as0f2d5b8d
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 125 REGISTER
User-Agent: Asterisk PBX 1.8.6.0
Authorization: Digest username="334xxxxxxx", realm="sipgate.de", algorithm=MD5, uri="sip:sipgate.de", nonce="4e6933d47deb3e2eb7989ee2e3c20a2f8462dc94", response="672e5f7f64c9936f974cd679bb6552e2"
Expires: 120
Contact: <sip:[email protected]:5060>
Content-Length: 0


---

<--- SIP read from UDP:217.10.79.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 78.51.16.192:5060;branch=z9hG4bK2fec2dd6
From: <sip:[email protected]>;tag=as0f2d5b8d
To: <sip:[email protected]>;tag=4fa8f7eb71cc68cca91a14abea886308.7c24
Call-ID: [email protected]
CSeq: 125 REGISTER
Contact: <sip:[email protected]:5060>;expires=120
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: REGISTER)
[Sep  8 23:28:26] NOTICE[17120]: chan_sip.c:20125 handle_response_register: Outbound Registration: Expiry for sipgate.de is 120 sec (Scheduling reregistration in 105 s)
    -- Executing [[email protected]:1] MSet("DAHDI/i2/73585120-4", "~~EXTEN~~=0151xxxxxxxx") in new stack
    -- Executing [[email protected]:2] Goto("DAHDI/i2/73585120-4", "sw_2_73585120,10") in new stack
    -- Goto (isdntelaus,sw_2_73585120,10)
    -- Executing [[email protected]:10] Dial("DAHDI/i2/73585120-4", "SIP/334xxxxxxx/0151xxxxxxxx,,r") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 5060
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 85.13.156.94:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 78.51.16.192:5060;branch=z9hG4bK6199021c
Max-Forwards: 70
From: "73585120" <sip:[email protected]>;tag=as3c3f2588
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.6.0
Date: Thu, 08 Sep 2011 21:28:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 1297421477 1297421477 IN IP4 78.51.16.192
s=Asterisk PBX 1.8.6.0
c=IN IP4 78.51.16.192
t=0 0
m=audio 28072 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called SIP/334xxxxxxx/0151xxxxxxxx
Retransmitting #1 (no NAT) to 85.13.156.94:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 78.51.16.192:5060;branch=z9hG4bK6199021c
Max-Forwards: 70
From: "73585120" <sip:[email protected]>;tag=as3c3f2588
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.6.0
Date: Thu, 08 Sep 2011 21:28:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 1297421477 1297421477 IN IP4 78.51.16.192
s=Asterisk PBX 1.8.6.0
c=IN IP4 78.51.16.192
t=0 0
m=audio 28072 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
Retransmitting #2 (no NAT) to 85.13.156.94:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 78.51.16.192:5060;branch=z9hG4bK6199021c
Max-Forwards: 70
From: "73585120" <sip:[email protected]>;tag=as3c3f2588
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.6.0
Date: Thu, 08 Sep 2011 21:28:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 1297421477 1297421477 IN IP4 78.51.16.192
s=Asterisk PBX 1.8.6.0
c=IN IP4 78.51.16.192
t=0 0
m=audio 28072 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
Retransmitting #3 (no NAT) to 85.13.156.94:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 78.51.16.192:5060;branch=z9hG4bK6199021c
Max-Forwards: 70
From: "73585120" <sip:[email protected]>;tag=as3c3f2588
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.6.0
Date: Thu, 08 Sep 2011 21:28:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 1297421477 1297421477 IN IP4 78.51.16.192
s=Asterisk PBX 1.8.6.0
c=IN IP4 78.51.16.192
t=0 0
m=audio 28072 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
Retransmitting #4 (no NAT) to 85.13.156.94:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 78.51.16.192:5060;branch=z9hG4bK6199021c
Max-Forwards: 70
From: "73585120" <sip:[email protected]>;tag=as3c3f2588
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.6.0
Date: Thu, 08 Sep 2011 21:28:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 1297421477 1297421477 IN IP4 78.51.16.192
s=Asterisk PBX 1.8.6.0
c=IN IP4 78.51.16.192
t=0 0
m=audio 28072 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Span 2: Channel 0/2 got hangup request, cause 16
Scheduling destruction of SIP dialog '[email protected]:5060' in 32000 ms (Method: INVITE)
  == Spawn extension (isdntelaus, sw_2_73585120, 10) exited non-zero on 'DAHDI/i2/73585120-4'
    -- Hungup 'DAHDI/i2/73585120-4'
Reliably Transmitting (no NAT) to 217.10.79.9:5060:
OPTIONS sip:sipgate.de SIP/2.0
Via: SIP/2.0/UDP 78.51.16.192:5060;branch=z9hG4bK76414742
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as65aec14a
To: <sip:sipgate.de>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.6.0
Date: Thu, 08 Sep 2011 21:28:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:217.10.79.9:5060 --->
SIP/2.0 200 OK
Record-Route: <sip:217.10.79.9;lr=on;ftag=as65aec14a>
Via: SIP/2.0/UDP 78.51.16.192:5060;branch=z9hG4bK76414742
From: "asterisk" <sip:[email protected]>;tag=as65aec14a
To: <sip:sipgate.de>;tag=4fa8f7eb71cc68cca91a14abea886308.7023
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Accept: */*
Accept-Encoding: 
Accept-Language: en
Supported: 
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
Retransmitting #5 (no NAT) to 85.13.156.94:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 78.51.16.192:5060;branch=z9hG4bK6199021c
Max-Forwards: 70
From: "73585120" <sip:[email protected]>;tag=as3c3f2588
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.6.0
Date: Thu, 08 Sep 2011 21:28:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 1297421477 1297421477 IN IP4 78.51.16.192
s=Asterisk PBX 1.8.6.0
c=IN IP4 78.51.16.192
t=0 0
m=audio 28072 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
Really destroying SIP dialog '[email protected]' Method: REGISTER
Retransmitting #6 (no NAT) to 85.13.156.94:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 78.51.16.192:5060;branch=z9hG4bK6199021c
Max-Forwards: 70
From: "73585120" <sip:[email protected]>;tag=as3c3f2588
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.6.0
Date: Thu, 08 Sep 2011 21:28:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 1297421477 1297421477 IN IP4 78.51.16.192
s=Asterisk PBX 1.8.6.0
c=IN IP4 78.51.16.192
t=0 0
m=audio 28072 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
[Sep  8 23:28:59] WARNING[17120]: chan_sip.c:3620 retrans_pkt: Retransmission timeout reached on transmission [email protected]:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
Really destroying SIP dialog '[email protected]:5060' Method: INVITE
office*CLI>
 

abw1oim

Aktives Mitglied
Mitglied seit
26 Mrz 2007
Beiträge
956
Punkte für Reaktionen
4
Punkte
18
Der Dialbefehl passt nicht zur Konfig:
Code:
Dial(SIP/sipgate/0151xxxxxxxx,,r)
wird funktionieren.
 

tameritoke

Neuer User
Mitglied seit
31 Jan 2009
Beiträge
16
Punkte für Reaktionen
0
Punkte
0
Der DIAL Befehl stimmt, du irsst dich. Ausserdem, wenn der nicht stimmen würde, käme die Meldung "extension not found" oder was ähnliches und ich hätte keinen sip debug. Du siehst ja selber im debug, dass asterisk die Pakete nicht weiterleiten kann.

Schaue dir den debug genau an! Morgen lese ich mir mal die SIP RFC komplett durch. Irgendwo muss ja der Fehler liegen.
 

abw1oim

Aktives Mitglied
Mitglied seit
26 Mrz 2007
Beiträge
956
Punkte für Reaktionen
4
Punkte
18
Sorry, aber das ist Unsinn! Das ist ein abgehender Call und ein Lesen des SIP-Debugs würde Dir aufzeigen, dass versucht wird, eine Verbindung zur IP 334xxxxxxx aufzubauen, die natürlich nicht existiert (da sie über die Konfig nicht aufgelöst werden kann) und daher permanant retransmitted wird.
Und "extension not found" ist nur bei eingehenden Calls relevant!
 

tameritoke

Neuer User
Mitglied seit
31 Jan 2009
Beiträge
16
Punkte für Reaktionen
0
Punkte
0
Ich habe den Fehler gefunden wie du mir gesagt hast! Fehler im Dialplan!
 
Zuletzt bearbeitet:

3CX PBX - GRATIS
Linux / Win / Cloud

Statistik des Forums

Themen
234,443
Beiträge
2,046,629
Mitglieder
354,202
Neuestes Mitglied
LukBar54