[Gelöst] * hat plötzlich Problem mit Audioübertragung

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vwittich

vwittich

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Hallo zusammen,

bin hier leicht am verzweifeln da ich keine Idee für die Ursache habe. Das System lief die letzten 8 Wochen stabil. Seit heute wird bei Anrufen über VoIP Netz die Gegenstelle nicht mehr gehört... die hören mich jedoch! Normalerweise gab es bisher höchstens Probleme mit den Ports... hab schon den ganzen Tag getestet aber nicht die Ursache finden können.

Ein Anruf sieht so aus:
Code: 
*CLI>
    -- Executing [10005@telefone:1] Dial("SIP/12-0820d190", "SIP/10005@sipgate") in new stack
    -- Called 10005@sipgate
    -- SIP/sipgate-08211108 answered SIP/12-0820d190
    -- Native bridging SIP/12-0820d190 and SIP/sipgate-08211108
  == Spawn extension (telefone, 10005, 1) exited non-zero on 'SIP/12-0820d190'
    -- Executing [10005@telefone:1] Dial("SIP/12-081f8b50", "SIP/10005@sipgate") in new stack
    -- Called 10005@sipgate
    -- SIP/sipgate-0820bc00 answered SIP/12-081f8b50
    -- Native bridging SIP/12-081f8b50 and SIP/sipgate-0820bc00
  == Spawn extension (telefone, 10005, 1) exited non-zero on 'SIP/12-081f8b50'

Im Debug Modus werden folgende Infos angezeigt:
Code: 
v=0
o=12 8000 8002 IN IP4 192.168.1.9
s=SIP Call
c=IN IP4 192.168.1.9
t=0 0
m=audio 5004 RTP/AVP 8
a=sendrecv
a=rtpmap:8 PCMA/8000
a=ptime:20

<------------->
--- (12 headers 9 lines) ---
Found RTP audio format 8
Peer audio RTP is at port 192.168.1.9:5004
Found audio description format PCMA for ID 8
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.1.9:5004
set_destination: Parsing <sip:[email protected]:5060;transport=udp> for address/port to send to
set_destination: set destination to 192.168.1.9, port 5060
Transmitting (no NAT) to 192.168.1.9:5060:
ACK sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK1f4a4f73;rport
From: <sip:[email protected]>;tag=as554c6ed4
To: "..." <sip:[email protected]>;tag=376dc18d3fbed787
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Mir fällt da nix ungewöhnliches auf... eingehende Anrufe über ISDN geht übrigens auch ohne Problem.

Könnt ihr mir bitte weiterführende Tipps geben?

mfG Valentin
 
Zuletzt bearbeitet:
Hmm... hat niemand einen Tipp?

Vielleicht hilft noch das Schema meiner sip.conf
Code: 
[general]
port=5060
bindaddr=0.0.0.0
context=default        
qualitfy=no
; Auswahl der Codec [...]
srvlookup=yes
language=de
canreinvite=yes         ; versucht, den kürzesten Audio-Pfad zu verwenden

register => 12345678:[email protected]/12345678e0       ; e0
register => 12345678:[email protected]/12345678e1     ; e1
register => 12345678:[email protected]/12345678e2       ; e2
register => 23456789:[email protected]/zzz           ; d

; Telefone
[10]
callerid=e0 <10>
type=friend
context=telefone
secret=12345678
host=dynamic

; usw. 

[sipgate]       ;  e0
type=friend
context=von_sipgate
username=12345678e0
fromuser=12345678e0
secret=yyy
host=sipgate.de
fromdomain=sipgate.de
qualify=yes               ; Ping Geraet an
insecure=very           ; otherwise I get authentication errors
nat=yes
call-limit=10           ; Limit der gleichzeitigen Anrufe

; usw. für e1, e2 und d

aus der extensions.conf sind wohl folgende Abschnitte wichtig:
Code: 
; Eingehend

[von_sipgate]
; z.B. für e0
exten => 12345678e0,1,Dial(SIP/10)
exten => 12345678e0,n,Hangup()

; [...]
; Extern wird über ein verschiedene Vorwahlregeln aufgerufen... so das eine Nummer immer mit 00YX beginnt...
[national]
; e0
exten => _00ZX./_1[23],1,set(CALLERID(name)=496912345678)
exten => _00ZX./_1[23],2,Set(CALLERID(number)=12345678e0)
exten => _00ZX./12345678e0,3,Dial(SIP/${EXTEN}@sipgate_h,60,trg)
exten => _00ZX./12345678e0,n,Hangup()

Naja vielleicht kann das irgendwie weiterhelfen... habe momentan die Telefone über ISDN (mISDN) laufen... kann aber keine Dauerlösung bleiben...

Gruß Valentin
 
Heute gab es seit langem mal wieder ein Problem auf diesem Asterisk. Erst schien keine Verbindung zu Sipgate aufgebaut werden zu können:
Code: 
Got SIP response 476 "Unresolvable destination (476/TM)" back from 217.10.79.9

Dies ließ sich jedoch durch einen neustart beheben. Danach ergab sich das Problem das beim Vermitteln von einem Telefon (10) zum zweiten (12) nur die Stimme des externen Anrufs am zweiten Apparat ankam jedoch keine ausgehende Übertragung stattfand.
Code: 
    -- Called 12
    -- SIP/12-08202ce8 is ringing
    -- SIP/12-08202ce8 answered SIP/sipgate_h-0820b160
    -- Native bridging SIP/sipgate_h-0820b160 and SIP/12-08202ce8

in der sip.conf steht dazu folgendes...
Code: 
[sipgate_h]       ; e1
type=friend
context=von_sipgate
username=12345678e1
fromuser=12345678e1
secret=yyy
host=sipgate.de
fromdomain=sipgate.de
qualify=yes
insecure=very
nat=yes
call-limit=10

Kann mir nicht irgendjemand mal nen Tipp geben?

Gruß Valentin
 
Hallo,

Es wäre schön wenn im Fehlerfall ein "sip set debug peer sipgate" oder "sip set debug" auf der asterisk anschaltest. und die Ausgabe hier postest.
Dann kann man sehen was die asterisk Raus schickt.

MFG Daniel
 
So ein paar Jährchen später bin ich schon wieder über das Problem bei einer alter Asterisk Maschine gestolpert... Es war die ganze Zeit nicht so wichtig, aber jetzt wollte ich doch mal eine Lösung gefunden haben. Also mal den Debug für Sipgate gesetzt und das Protokoll etwas genauer untersucht nach dem ein Anruf über den Sipgate-Anschluss reinkam. Der Anruf kam von der Nummer 01734444444 und der Sip Empfänger ist 00496917300003.

Das einzige was mir beim durchsehen der Zeilen aufgefallen ist, ist das für die Audio-Übertragung ein Port mit der Nummer 19166 vergeben wird. Das finde ich etwas merkwürdig.

An welcher Stelle ist den sonst nach dem Problem zu suchen?

Gruß Valentin

Code: 
Asterisk 1.4.18, Copyright (C) 1999 - 2008 Digium, Inc. and others.
=========================================================================
Connected to Asterisk 1.4.18 currently running on server1 (pid = 2894)
server1*CLI> 
Verbosity is at least 3
<------------->
#--- (0 headers 0 lines) Nat keepalive ---
#
#[Kserver1*CLI> 
[Mar  8 13:05:32] #[1;33;40mNOTICE#[0;37;40m[2952]: #[1;37;40mchan_sip.c#[0;37;40m:#[1;37;40m14927#[0;37;40m #[1;37;40mhandle_request_subscribe#[0;37;40m: #Received SIP subscribe for peer without mailbox: 16
#
#[Kserver1*CLI> 
<--- SIP read from 217.10.79.9:5060 --->
INVITE sip:[email protected] SIP/2.0
Record-Route: <sip:217.10.79.9;lr;ftag=as702ff89a>
Record-Route: <sip:172.20.40.3;lr=on>
Record-Route: <sip:217.10.79.9;lr;ftag=as702ff89a>
Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK98e2.3a89b360ce4f511fab4b4f4b25ad6c56.0
Via: SIP/2.0/UDP 172.20.40.3;branch=z9hG4bK98e2.9dd8b2c4.0
Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.116.119.66;branch=z9hG4bK98e2.3be0aa288de6da63cede6736b3ddf0c9.0
Via: SIP/2.0/UDP 212.9.44.5:5060;branch=z9hG4bK62eb8767
Max-Forwards: 67
From: "01734444444" <sip:[email protected]>;tag=as702ff89a
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Content-Type: application/sdp
Content-Length: 464

v=0
o=root 1528325800 1528325800 IN IP4 212.9.44.5
s=sipgate VoIP GW
c=IN IP4 217.10.77.20
t=0 0
m=audio 45322 RTP/AVP 8 0 3 97 18 112 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=direction:active
a=nortpproxy:yes

<------------->
#--- (18 headers 21 lines) ---
#Sending to 217.10.79.9 : 5060 (no NAT)
#Using INVITE request as basis request - [email protected]
#Found peer 'sipgate'
#Found RTP audio format 8
#Found RTP audio format 0
#Found RTP audio format 3
#Found RTP audio format 97
#Found RTP audio format 18
#Found RTP audio format 112
#Found RTP audio format 101
#Peer audio RTP is at port 217.10.77.20:45322
#Found audio description format PCMA for ID 8
#Found audio description format PCMU for ID 0
#Found audio description format GSM for ID 3
#Found audio description format iLBC for ID 97
#Found audio description format G729 for ID 18
#Found audio description format G726-32 for ID 112
#Found audio description format telephone-event for ID 101
#Capabilities: us - 0x14e (gsm|ulaw|alaw|slin|g729), peer - audio=0xd0e (gsm|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x10e (gsm|ulaw|alaw|g729)
#Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
#Peer audio RTP is at port 217.10.77.20:45322
#[Mar  8 13:05:33] #[1;32;40mDEBUG#[0;37;40m[2952]: #[1;37;40mchan_sip.c#[0;37;40m:#[1;37;40m3250#[0;37;40m #[1;37;40mupdate_call_counter#[0;37;40m: #Call from peer 'sipgate' is 1 out of 10
#Looking for 2000002e0 in von_sipgate (domain 10.0.0.2)
#list_route: hop: <sip:217.10.79.9;lr;ftag=as702ff89a>
#list_route: hop: <sip:172.20.40.3;lr=on>
#list_route: hop: <sip:217.10.79.9;lr;ftag=as702ff89a>
#
<--- Transmitting (NAT) to 217.10.79.9:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK98e2.3a89b360ce4f511fab4b4f4b25ad6c56.0;received=217.10.79.9
Via: SIP/2.0/UDP 172.20.40.3;branch=z9hG4bK98e2.9dd8b2c4.0
Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.116.119.66;branch=z9hG4bK98e2.3be0aa288de6da63cede6736b3ddf0c9.0
Via: SIP/2.0/UDP 212.9.44.5:5060;branch=z9hG4bK62eb8767
Record-Route: <sip:217.10.79.9;lr;ftag=as702ff89a>
Record-Route: <sip:172.20.40.3;lr=on>
Record-Route: <sip:217.10.79.9;lr;ftag=as702ff89a>
From: "01734444444" <sip:[email protected]>;tag=as702ff89a
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:[email protected]>
Content-Length: 0


<------------>
#
#[Kserver1*CLI> 
    -- Executing [2000002e0@von_sipgate:1] #[1;36;40mDial#[0;37;40m("#[1;35;40mSIP/2000002e1-08814df0#[0;37;40m", "#[1;35;40mSIP/12#[0;37;40m") in new stack
#
#[Kserver1*CLI> 
[Mar  8 13:05:33] #[1;32;40mDEBUG#[0;37;40m[19591]: #[1;37;40mchan_sip.c#[0;37;40m:#[1;37;40m3250#[0;37;40m #[1;37;40mupdate_call_counter#[0;37;40m: #Call to peer '12' is 1 out of 10
#
#[Kserver1*CLI> 
    -- Called 12
#
#[Kserver1*CLI> 
    -- SIP/12-08819158 is ringing
#
#[Kserver1*CLI> 
<--- Transmitting (NAT) to 217.10.79.9:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK98e2.3a89b360ce4f511fab4b4f4b25ad6c56.0;received=217.10.79.9
Via: SIP/2.0/UDP 172.20.40.3;branch=z9hG4bK98e2.9dd8b2c4.0
Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.116.119.66;branch=z9hG4bK98e2.3be0aa288de6da63cede6736b3ddf0c9.0
Via: SIP/2.0/UDP 212.9.44.5:5060;branch=z9hG4bK62eb8767
Record-Route: <sip:217.10.79.9;lr;ftag=as702ff89a>
Record-Route: <sip:172.20.40.3;lr=on>
Record-Route: <sip:217.10.79.9;lr;ftag=as702ff89a>
From: "01734444444" <sip:[email protected]>;tag=as702ff89a
To: <sip:[email protected]>;tag=as410cd698
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:[email protected]>
Content-Length: 0


<------------>
#
#[Kserver1*CLI> 
[Mar  8 13:05:33] #[1;33;40mNOTICE#[0;37;40m[2952]: #[1;37;40mchan_sip.c#[0;37;40m:#[1;37;40m14927#[0;37;40m #[1;37;40mhandle_request_subscribe#[0;37;40m: #Received SIP subscribe for peer without mailbox: 16
#
#[Kserver1*CLI> 
<--- SIP read from 217.10.79.9:5060 --->

<------------->
#--- (0 headers 0 lines) Nat keepalive ---
#
#[Kserver1*CLI> 
[Mar  8 13:05:35] #[1;32;40mDEBUG#[0;37;40m[2952]: #[1;37;40mchan_sip.c#[0;37;40m:#[1;37;40m5873#[0;37;40m #[1;37;40mreqprep#[0;37;40m: #Strict routing enforced for session [email protected]
#    -- SIP/12-08819158 answered SIP/2000002e1-08814df0
#Audio is at 10.0.0.2 port 19166
#Adding codec 0x8 (alaw) to SDP
#Adding codec 0x4 (ulaw) to SDP
#Adding codec 0x100 (g729) to SDP
#Adding codec 0x2 (gsm) to SDP
#Adding non-codec 0x1 (telephone-event) to SDP
#
<--- Reliably Transmitting (NAT) to 217.10.79.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK98e2.3a89b360ce4f511fab4b4f4b25ad6c56.0;received=217.10.79.9
Via: SIP/2.0/UDP 172.20.40.3;branch=z9hG4bK98e2.9dd8b2c4.0
Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.116.119.66;branch=z9hG4bK98e2.3be0aa288de6da63cede6736b3ddf0c9.0
Via: SIP/2.0/UDP 212.9.44.5:5060;branch=z9hG4bK62eb8767
Record-Route: <sip:217.10.79.9;lr;ftag=as702ff89a>
Record-Route: <sip:172.20.40.3;lr=on>
Record-Route: <sip:217.10.79.9;lr;ftag=as702ff89a>
From: "01734444444" <sip:[email protected]>;tag=as702ff89a
To: <sip:[email protected]>;tag=as410cd698
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 330

v=0
o=root 2894 2894 IN IP4 10.0.0.2
s=session
c=IN IP4 10.0.0.2
t=0 0
m=audio 19166 RTP/AVP 8 0 18 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
#    -- Native bridging SIP/2000002e1-08814df0 and SIP/12-08819158
#[Mar  8 13:05:35] #[1;32;40mDEBUG#[0;37;40m[19591]: #[1;37;40mchan_sip.c#[0;37;40m:#[1;37;40m5873#[0;37;40m #[1;37;40mreqprep#[0;37;40m: #Strict routing enforced for session [email protected]
#
#[Kserver1*CLI> 
 Extension Changed 12[call_sips] new state InUse for Notify User 13 
# Extension Changed 12[call_sips] new state InUse for Notify User 16 
#
#[Kserver1*CLI> 
<--- SIP read from 217.10.79.9:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK98e2.4107216d7be81185bdfc671c80558fbb.0
Via: SIP/2.0/UDP 172.20.40.3;branch=z9hG4bK98e2.9dd8b2c4.2
Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.116.119.66;branch=z9hG4bK98e2.a516d45363824beb1f3495397fc65c22.0
Via: SIP/2.0/UDP 212.9.44.5:5060;branch=z9hG4bK45cde584
Max-Forwards: 67
From: "01734444444" <sip:[email protected]>;tag=as702ff89a
To: <sip:[email protected]>;tag=as410cd698
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 102 ACK
Content-Length: 0
X-hint: rr-enforced


<------------->
#--- (13 headers 0 lines) ---
#set_destination: Parsing <sip:217.10.79.9;lr;ftag=as702ff89a> for address/port to send to
#set_destination: set destination to 217.10.79.9, port 5060
#Audio is at 10.0.0.2 port 19166
#Adding codec 0x8 (alaw) to SDP
#Adding non-codec 0x1 (telephone-event) to SDP
#Reliably Transmitting (NAT) to 217.10.79.9:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK0c275d8a;rport
Route: <sip:217.10.79.9;lr;ftag=as702ff89a>,<sip:172.20.40.3;lr=on>,<sip:217.10.79.9;lr;ftag=as702ff89a>
From: <sip:[email protected]>;tag=as410cd698
To: "01734444444" <sip:[email protected]>;tag=as702ff89a
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 237

v=0
o=root 2894 2895 IN IP4 10.0.0.12
s=session
c=IN IP4 10.0.0.12
t=0 0
m=audio 5044 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
#
#[Kserver1*CLI> 
<--- SIP read from 217.10.79.9:5060 --->
SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP 10.0.0.2:5060;received=93.207.67.46;branch=z9hG4bK0c275d8a;rport=5060
From: <sip:[email protected]>;tag=as410cd698
To: "01734444444" <sip:[email protected]>;tag=as702ff89a
Call-ID: [email protected]
CSeq: 102 INVITE
Content-Length: 0


<------------->
#--- (7 headers 0 lines) ---
#
#[Kserver1*CLI> 
<--- SIP read from 217.10.79.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.2:5060;received=93.207.67.46;branch=z9hG4bK0c275d8a;rport=5060
From: <sip:[email protected]>;tag=as410cd698
To: "01734444444" <sip:[email protected]>;tag=as702ff89a
Call-ID: [email protected]
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Content-Length: 274

v=0
o=root 1528325800 1528325801 IN IP4 212.9.44.5
s=sipgate VoIP GW
c=IN IP4 217.10.77.20
t=0 0
m=audio 45322 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=nortpproxy:yes

<------------->
#--- (11 headers 13 lines) ---
#Found RTP audio format 8
#Found RTP audio format 101
#Peer audio RTP is at port 217.10.77.20:45322
#Found audio description format PCMA for ID 8
#Found audio description format telephone-event for ID 101
#Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
#Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
#Peer audio RTP is at port 217.10.77.20:45322
#set_destination: Parsing <sip:217.10.79.9;lr;ftag=as702ff89a> for address/port to send to
#set_destination: set destination to 217.10.79.9, port 5060
#Transmitting (NAT) to 217.10.79.9:5060:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK6422eeda;rport
Route: <sip:217.10.79.9;lr;ftag=as702ff89a>,<sip:172.20.40.3;lr=on>,<sip:217.10.79.9;lr;ftag=as702ff89a>
From: <sip:[email protected]>;tag=as410cd698
To: "01734444444" <sip:[email protected]>;tag=as702ff89a
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
#
#[Kserver1*CLI> 
[Mar  8 13:05:36] #[1;32;40mDEBUG#[0;37;40m[2952]: #[1;37;40mchan_sip.c#[0;37;40m:#[1;37;40m5873#[0;37;40m #[1;37;40mreqprep#[0;37;40m: #Strict routing enforced for session [email protected]
#
#[Kserver1*CLI> 
Really destroying SIP dialog '[email protected]' Method: REGISTER
#
#[Kserver1*CLI> 
Really destroying SIP dialog '[email protected]' Method: REGISTER
#
#[Kserver1*CLI> 
<--- SIP read from 217.10.79.9:5060 --->

<------------->
#--- (0 headers 0 lines) Nat keepalive ---
#
#[Kserver1*CLI> 
Really destroying SIP dialog '[email protected]' Method: REGISTER
#
#[Kserver1*CLI> 
set_destination: Parsing <sip:217.10.79.9;lr;ftag=as702ff89a> for address/port to send to
#set_destination: set destination to 217.10.79.9, port 5060
#Audio is at 10.0.0.2 port 19166
#Adding codec 0x8 (alaw) to SDP
#Adding non-codec 0x1 (telephone-event) to SDP
#Reliably Transmitting (NAT) to 217.10.79.9:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK3ddaeaa3;rport
Route: <sip:217.10.79.9;lr;ftag=as702ff89a>,<sip:172.20.40.3;lr=on>,<sip:217.10.79.9;lr;ftag=as702ff89a>
From: <sip:[email protected]>;tag=as410cd698
To: "01734444444" <sip:[email protected]>;tag=as702ff89a
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 236

v=0
o=root 2894 2896 IN IP4 10.0.0.2
s=session
c=IN IP4 10.0.0.2
t=0 0
m=audio 19166 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
#[Mar  8 13:05:40] #[1;32;40mDEBUG#[0;37;40m[19591]: #[1;37;40mchan_sip.c#[0;37;40m:#[1;37;40m3224#[0;37;40m #[1;37;40mupdate_call_counter#[0;37;40m: #Call to peer '12' removed from call limit 10
#  == Spawn extension (von_sipgate, 2000002e0, 1) exited non-zero on 'SIP/2000002e1-08814df0'
#
#[Kserver1*CLI> 
[Mar  8 13:05:40] #[1;32;40mDEBUG#[0;37;40m[19591]: #[1;37;40mchan_sip.c#[0;37;40m:#[1;37;40m3224#[0;37;40m #[1;37;40mupdate_call_counter#[0;37;40m: #Call from peer 'sipgate' removed from call limit 10
#Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: ACK)
#
#[Kserver1*CLI> 
<--- SIP read from 217.10.79.9:5060 --->
SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP 10.0.0.2:5060;received=93.207.67.46;branch=z9hG4bK3ddaeaa3;rport=5060
From: <sip:[email protected]>;tag=as410cd698
To: "01734444444" <sip:[email protected]>;tag=as702ff89a
Call-ID: [email protected]
CSeq: 103 INVITE
Content-Length: 0


<------------->
#--- (7 headers 0 lines) ---
#
#[Kserver1*CLI> 
<--- SIP read from 217.10.79.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.2:5060;received=93.207.67.46;branch=z9hG4bK3ddaeaa3;rport=5060
From: <sip:[email protected]>;tag=as410cd698
To: "01734444444" <sip:[email protected]>;tag=as702ff89a
Call-ID: [email protected]
CSeq: 103 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Content-Length: 274

v=0
o=root 1528325800 1528325802 IN IP4 212.9.44.5
s=sipgate VoIP GW
c=IN IP4 217.10.77.20
t=0 0
m=audio 45322 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=nortpproxy:yes

<------------->
#--- (11 headers 13 lines) ---
#Found RTP audio format 8
#Found RTP audio format 101
#Peer audio RTP is at port 217.10.77.20:45322
#Found audio description format PCMA for ID 8
#Found audio description format telephone-event for ID 101
#Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
#Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
#Peer audio RTP is at port 217.10.77.20:45322
#list_route: hop: <sip:[email protected]:5060>
#[Mar  8 13:05:40] #[1;32;40mDEBUG#[0;37;40m[2952]: #[1;37;40mchan_sip.c#[0;37;40m:#[1;37;40m5873#[0;37;40m #[1;37;40mreqprep#[0;37;40m: #Strict routing enforced for session [email protected]
#set_destination: Parsing <sip:[email protected]:5060> for address/port to send to
#set_destination: set destination to 212.9.44.5, port 5060
#Transmitting (NAT) to 217.10.79.9:5060:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK4fac50e3;rport
From: <sip:[email protected]>;tag=as410cd698
To: "01734444444" <sip:[email protected]>;tag=as702ff89a
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
#[Mar  8 13:05:40] #[1;32;40mDEBUG#[0;37;40m[2952]: #[1;37;40mchan_sip.c#[0;37;40m:#[1;37;40m5873#[0;37;40m #[1;37;40mreqprep#[0;37;40m: #Strict routing enforced for session [email protected]
#set_destination: Parsing <sip:[email protected]:5060> for address/port to send to
#set_destination: set destination to 212.9.44.5, port 5060
#Reliably Transmitting (NAT) to 217.10.79.9:5060:
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK143bb15b;rport
From: <sip:[email protected]>;tag=as410cd698
To: "01734444444" <sip:[email protected]>;tag=as702ff89a
Call-ID: [email protected]
CSeq: 104 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
#Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: ACK)
#
#[Kserver1*CLI> 
<--- SIP read from 217.10.79.9:5060 --->
SIP/2.0 404 not found (unknown domain)
Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK143bb15b;rport=5060;received=93.207.67.46
From: <sip:[email protected]>;tag=as410cd698
To: "01734444444" <sip:[email protected]>;tag=as702ff89a
Call-ID: [email protected]
CSeq: 104 BYE
Content-Length: 0


<------------->
#--- (7 headers 0 lines) ---
#
#[Kserver1*CLI> 
<--- SIP read from 217.10.79.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.2:5060;received=93.207.67.46;branch=z9hG4bK3ddaeaa3;rport=5060
From: <sip:[email protected]>;tag=as410cd698
To: "01734444444" <sip:[email protected]>;tag=as702ff89a
Call-ID: [email protected]
CSeq: 103 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Content-Length: 274

v=0
o=root 1528325800 1528325802 IN IP4 212.9.44.5
s=sipgate VoIP GW
c=IN IP4 217.10.77.20
t=0 0
m=audio 45322 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=nortpproxy:yes

<------------->
#--- (11 headers 13 lines) ---
#
#[Kserver1*CLI> 
Really destroying SIP dialog '[email protected]' Method: REGISTER
#
#[Kserver1*CLI> 
<--- SIP read from 217.10.79.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.2:5060;received=93.207.67.46;branch=z9hG4bK3ddaeaa3;rport=5060
From: <sip:[email protected]>;tag=as410cd698
To: "01734444444" <sip:[email protected]>;tag=as702ff89a
Call-ID: [email protected]
CSeq: 103 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Content-Length: 274

v=0
o=root 1528325800 1528325802 IN IP4 212.9.44.5
s=sipgate VoIP GW
c=IN IP4 217.10.77.20
t=0 0
m=audio 45322 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=nortpproxy:yes

<------------->
#--- (11 headers 13 lines) ---
#
#[Kserver1*CLI> 
<--- SIP read from 217.10.79.9:5060 --->

<------------->
#--- (0 headers 0 lines) Nat keepalive ---
#
#[Kserver1*CLI> exit
<--- SIP read from 217.10.79.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.2:5060;received=93.207.67.46;branch=z9hG4bK3ddaeaa3;rport=5060
From: <sip:[email protected]>;tag=as410cd698
To: "01734444444" <sip:[email protected]>;tag=as702ff89a
Call-ID: [email protected]
CSeq: 103 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Content-Length: 274

v=0
o=root 1528325800 1528325802 IN IP4 212.9.44.5
s=sipgate VoIP GW
c=IN IP4 217.10.77.20
t=0 0
m=audio 45322 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=nortpproxy:yes

<------------->
#--- (11 headers 13 lines) ---
#
#[Kserver1*CLI> exit
 
Setze canreinvite=no in der sip.conf. Sonst wir ein neues INVITE an sipgate gesendet mit dem RTP Port und der internen IP des Telefon und Asterisk klinkt sich aus dem RTP stream raus.

...das für die Audio-Übertragung ein Port mit der Nummer 19166 vergeben wird.
Zum Vergrößern anklicken....
Die RTP-Portrage welche Asterisk vergibt kannst du in der rtp.conf einstellen.
 
wildzero schrieb:
Setze canreinvite=no in der sip.conf. Sonst wir ein neues INVITE an sipgate gesendet mit dem RTP Port und der internen IP des Telefon und Asterisk klinkt sich aus dem RTP stream raus.
Zum Vergrößern anklicken....

Danke @wildzero genau das canreinvite fehlte im Sipgate-context.
 
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