Stummer Anrufer die Zweite :(

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barmeier

barmeier

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Hallo,

dank des Forums habe ich meinen Asterisk mit C2 AVM und Anlagenanschluss
zum Laufen gebracht alles läuft Bestens... bis auf folgendes Problem:

Wenn ich interne Gespräche führe, also SIP->SIP kann der Anrufer vom
angerufenen nicht gehört werden. Drückt einer der beiden zweimal nacheinander
auf die HOLD Taste des Telefons können sich beide hören. Das passiert häufig,
aber nicht immer und ist absoluter Terror, wenn man ein Gespräch von Außen
vermittelt bekommt.

Ich habe keine Firewalls im internen Netz und ich habe keine Protokolle auf einem der Switches oder Router geblockt.

Was mich wundert, ist das im Telefon Setup als RTP Port 1722 angegeben wird, in der rtp.conf aber

rtpstart=10000
rtpend=20000

steht. WIe funktioniert das überhaupt ??

Hat jemand eine Idee wie ich mein Problem lösen kann ??

Danke im voraus.

Ciao
Matze
 
Hast Du denn die Ports 10000 bis 20000 UDP an den Asterisk weitergeleitet?
 
Hi madiehl,

was genau verstehst du unter weitergeleitet ?
Telefone und Asterisk sind im internen Netz, keine Firewall, kein NAT, kein Portforwarding.

Was ich nicht verstehe ist, wenn auf dem Telefon RTP Port 1722 angegeben wird, wie
kann der Asterisk überhaupt antworten ??

Nimmt er jeden Datenverkehr per UDP auf, der das richtige Format hat ???

Ich werde weiter probieren, aber vielleicht hast du oder jemand anderes noch einen Hinweis
für mich.

Ciao
Matze
 
Hallo zusammen,

ich konnte einen Workaround erarbeiten. Wie es aussieht reicht es ein SIP DEBUG auf der Konsole einzugeben. Solange der Debugmodus aktiv ist läuft alles perfekt.

Sobald ich SIP NO DEBUG eingebe geht es nicht mehr :confused: :confused:

Hat jemand so etwas schon mal gesehen und kann mir erklären was das bedeutet ??

Ciao
Matze
 
ist für das telefon nat=no gesetzt?
sonst kannst du ja mal das debug posten
 
Hi,

ich habe nat=no und nat=never versucht. Hier ist meine sip.conf

Code: 
[general]
context=default                 ; Default context for incoming calls
bindport=5060                   ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0                ; IP address to bind to (0.0.0.0 binds to all)
callevents=yes
;srvlookup=yes                  ; Enable DNS SRV lookups on outbound calls
                                ; Note: Asterisk only uses the first host
                                ; in SRV records
                                ; Disabling DNS SRV lookups disables the
                                ; ability to place SIP calls based on domain
                                ; names to some other SIP users on the Internet

nat=never
tos=184

disallow=all                    ; First disallow all codecs
allow=ulaw
allow=alaw                      ; Allow codecs in order of preference
allow=gsm
allow=g723
allow=g729
musicclass=default

[12]
type=friend
host=192.168.10.12
username=12
authname=12
secret=1234
context=default

[13]
type=friend
host=192.168.10.13
callerid=398068313
username=13
authname=13
secret=1234
context=default

[14]
type=friend
host=192.168.10.14
username=14
authname=14
secret=1234
context=default

[15]
type=friend
host=192.168.10.15
username=15
authname=15
context=default

[16]
type=friend
;host=192.168.10.16
host=dynamic
username=16
authname=16
secret=1234
context=default

[20]
type=friend
;host=dynamic
;fromuser=398068320
host=192.168.10.20
username=20
authname=20
secret=1234
context=default

Ein SIP Debug kann ich nur erzeugen, in dem ich es ein paar mal an und abschalte (das SIP Debug, dann tritt das Problem auch mit aktiviertem Debug auf :confused:

Das Log enthält den Versuch einer Verbindung, in der der Anrufer keinen Audio vom angerufenen bekommen hat, der Angerufene aber alles wunder bar hören konnte.

Code: 
<-- SIP read from 192.168.10.16:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.10.16;rport;branch=z9hG4bKckbkvbbp
Max-Forwards: 70
To: <sip:[email protected]>
From: "16" <sip:[email protected]>;tag=nhzez
Call-ID: [email protected]
CSeq: 850 INVITE
Contact: <sip:[email protected]>
Content-Type: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, REFER, NOTIFY, SUBSCRIBE
Supported: 100rel
User-Agent: Twinkle/0.5
Content-Length: 250

v=0
o=16 1711186251 423106286 IN IP4 192.168.10.16
s=-
c=IN IP4 192.168.10.16
t=0 0
m=audio 8000 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

--- (13 headers 12 lines)---
Using INVITE request as basis request - [email protected]
Sending to 192.168.10.16 : 5060 (non-NAT)
Reliably Transmitting (no NAT) to 192.168.10.16:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.10.16;rport;branch=z9hG4bKckbkvbbp;received=192.168.10.16
From: "16" <sip:[email protected]>;tag=nhzez
To: <sip:[email protected]>;tag=as4531458b
Call-ID: [email protected]
CSeq: 850 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:[email protected]>
Proxy-Authenticate: Digest realm="asterisk", nonce="1f61e569"
Content-Length: 0


---
Scheduling destruction of call '[email protected]' in 15000 ms
Found user '16'

<-- SIP read from 192.168.10.16:5060:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.10.16;rport;branch=z9hG4bKckbkvbbp
Max-Forwards: 70
To: <sip:[email protected]>;tag=as4531458b
From: "16" <sip:[email protected]>;tag=nhzez
Call-ID: [email protected]
CSeq: 850 ACK
User-Agent: Twinkle/0.5
Content-Length: 0


--- (9 headers 0 lines)---

<-- SIP read from 192.168.10.16:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.10.16;rport;branch=z9hG4bKyqyssgnh
Max-Forwards: 70
Proxy-Authorization: Digest username="16",realm="asterisk",nonce="1f61e569",uri="sip:[email protected]",response="08649ec6931be6e927f7c8023dc591b5",algorithm=MD5
To: <sip:[email protected]>
From: "16" <sip:[email protected]>;tag=nhzez
Call-ID: [email protected]
CSeq: 851 INVITE
Contact: <sip:[email protected]>
Content-Type: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, REFER, NOTIFY, SUBSCRIBE
Supported: 100rel
User-Agent: Twinkle/0.5
Content-Length: 250

v=0
o=16 1711186251 423106286 IN IP4 192.168.10.16
s=-
c=IN IP4 192.168.10.16
t=0 0
m=audio 8000 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

--- (14 headers 12 lines)---
Using INVITE request as basis request - [email protected]
Sending to 192.168.10.16 : 5060 (non-NAT)
Found user '16'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 101
Peer audio RTP is at port 192.168.10.16:8000
Found description format PCMU
Found description format PCMA
Found description format GSM
Found description format telephone-event
Capabilities: us - 0x10f (g723|gsm|ulaw|alaw|g729), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 20 in default (domain 192.168.10.1)
list_route: hop: <sip:[email protected]>
Transmitting (no NAT) to 192.168.10.16:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.10.16;rport;branch=z9hG4bKyqyssgnh;received=192.168.10.16
From: "16" <sip:[email protected]>;tag=nhzez
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 851 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:[email protected]>
Content-Length: 0


---
We're at 192.168.10.1 port 18746
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x1 (g723) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 15 lines
Reliably Transmitting (no NAT) to 192.168.10.20:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK63919bb3
From: "16" <sip:[email protected]>;tag=as04d6e020
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 22 Feb 2006 15:46:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 332

v=0
o=root 5499 5499 IN IP4 192.168.10.1
s=session
c=IN IP4 192.168.10.1
t=0 0
m=audio 18746 RTP/AVP 0 8 3 4 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---

<-- SIP read from 192.168.10.20:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK63919bb3
Call-ID: [email protected]
CSeq: 102 INVITE
From: "16" <sip:[email protected]>;tag=as04d6e020
To: <sip:[email protected]>;tag=IPfP7FSeAp07cigB
Contact: <sip:[email protected]:5060>
Content-Length: 0


--- (8 headers 0 lines)---
Transmitting (no NAT) to 192.168.10.16:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.10.16;rport;branch=z9hG4bKyqyssgnh;received=192.168.10.16
From: "16" <sip:[email protected]>;tag=nhzez
To: <sip:[email protected]>;tag=as3f36a918
Call-ID: [email protected]
CSeq: 851 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:[email protected]>
Content-Length: 0


---

<-- SIP read from 192.168.10.20:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK63919bb3
Call-ID: [email protected]
CSeq: 102 INVITE
From: "16" <sip:[email protected]>;tag=as04d6e020
To: <sip:[email protected]>;tag=IPfP7FSeAp07cigB
Contact: <sip:[email protected]:5060>
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
Supported: replaces
Content-Type: application/sdp
Content-Length: 139

v=0
o=- 91792934 19784135 IN IP4 192.168.10.20
s=SIP CALL
c=IN IP4 192.168.10.20
t=0 0
m=audio 17220 RTP/AVP 0
a=rtpmap:0 PCMU/8000

--- (11 headers 7 lines)---
Found RTP audio format 0
Peer audio RTP is at port 192.168.10.20:17220
Found description format PCMU
Capabilities: us - 0x10f (g723|gsm|ulaw|alaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
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 192.168.10.20, port 5060
Transmitting (no NAT) to 192.168.10.20:5060:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK13254a83
From: "16" <sip:[email protected]>;tag=as04d6e020
To: <sip:[email protected]>;tag=IPfP7FSeAp07cigB
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
We're at 192.168.10.1 port 11514
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x1 (g723) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.10.16:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.16;rport;branch=z9hG4bKyqyssgnh;received=192.168.10.16
From: "16" <sip:[email protected]>;tag=nhzez
To: <sip:[email protected]>;tag=as3f36a918
Call-ID: [email protected]
CSeq: 851 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 332

v=0
o=root 5499 5499 IN IP4 192.168.10.1
s=session
c=IN IP4 192.168.10.1
t=0 0
m=audio 11514 RTP/AVP 0 8 3 4 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
set_destination: Parsing <sip:[email protected]:5060> for address/port to send to
set_destination: set destination to 192.168.10.20, port 5060
We're at 192.168.10.1 port 18746
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
13 headers, 10 lines
Reliably Transmitting (no NAT) to 192.168.10.20:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK33e07126
From: "16" <sip:[email protected]>;tag=as04d6e020
To: <sip:[email protected]>;tag=IPfP7FSeAp07cigB
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
X-asterisk-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 206

v=0
o=root 5499 5500 IN IP4 192.168.10.16
s=session
c=IN IP4 192.168.10.16
t=0 0
m=audio 8000 RTP/AVP 0 8 3
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=silenceSupp:off - - - -

---

<-- SIP read from 192.168.10.16:5060:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.10.16;rport;branch=z9hG4bKnlokqyna
Max-Forwards: 70
Proxy-Authorization: Digest username="16",realm="asterisk",nonce="1f61e569",uri="sip:[email protected]",response="08649ec6931be6e927f7c8023dc591b5",algorithm=MD5
To: <sip:[email protected]>;tag=as3f36a918
From: "16" <sip:[email protected]>;tag=nhzez
Call-ID: [email protected]
CSeq: 851 ACK
User-Agent: Twinkle/0.5
Content-Length: 0


--- (10 headers 0 lines)---
set_destination: Parsing <sip:[email protected]> for address/port to send to
set_destination: set destination to 192.168.10.16, port 5060
We're at 192.168.10.1 port 11514
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 10 lines
Reliably Transmitting (no NAT) to 192.168.10.16:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK757884bd
From: <sip:[email protected]>;tag=as3f36a918
To: "16" <sip:[email protected]>;tag=nhzez
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
X-asterisk-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 216

v=0
o=root 5499 5500 IN IP4 192.168.10.20
s=session
c=IN IP4 192.168.10.20
t=0 0
m=audio 17220 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---

<-- SIP read from 192.168.10.20:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK33e07126
Call-ID: [email protected]
CSeq: 103 INVITE
From: "16" <sip:[email protected]>;tag=as04d6e020
To: <sip:[email protected]>;tag=IPfP7FSeAp07cigB
Contact: <sip:[email protected]:5060>
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
Supported: replaces
Content-Type: application/sdp
Content-Length: 139

v=0
o=- 99407365 84190217 IN IP4 192.168.10.20
s=SIP CALL
c=IN IP4 192.168.10.20
t=0 0
m=audio 17220 RTP/AVP 0
a=rtpmap:0 PCMU/8000

--- (11 headers 7 lines)---
Found RTP audio format 0
Peer audio RTP is at port 192.168.10.20:17220
Found description format PCMU
Capabilities: us - 0x10f (g723|gsm|ulaw|alaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
set_destination: Parsing <sip:[email protected]:5060> for address/port to send to
set_destination: set destination to 192.168.10.20, port 5060
Transmitting (no NAT) to 192.168.10.20:5060:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK3458c279
From: "16" <sip:[email protected]>;tag=as04d6e020
To: <sip:[email protected]>;tag=IPfP7FSeAp07cigB
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---

<-- SIP read from 192.168.10.16:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK757884bd
To: "16" <sip:[email protected]>;tag=nhzez
From: <sip:[email protected]>;tag=as3f36a918
Call-ID: [email protected]
CSeq: 102 INVITE
Contact: <sip:[email protected]>
Content-Type: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, REFER, NOTIFY, SUBSCRIBE
Server: Twinkle/0.5
Supported:
Content-Length: 203

v=0
o=16 1711186251 423106286 IN IP4 192.168.10.16
s=-
c=IN IP4 192.168.10.16
t=0 0
m=audio 8000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

--- (12 headers 10 lines)---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 192.168.10.16:8000
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0x10f (g723|gsm|ulaw|alaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:[email protected]>
set_destination: Parsing <sip:[email protected]> for address/port to send to
set_destination: set destination to 192.168.10.16, port 5060
Transmitting (no NAT) to 192.168.10.16:5060:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK7b2fc0cf
From: <sip:[email protected]>;tag=as3f36a918
To: "16" <sip:[email protected]>;tag=nhzez
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---

<-- SIP read from 192.168.10.16:5060:
BYE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.10.16;rport;branch=z9hG4bKrrqpuppk
Max-Forwards: 70
To: <sip:[email protected]>;tag=as3f36a918
From: "16" <sip:[email protected]>;tag=nhzez
Call-ID: [email protected]
CSeq: 852 BYE
User-Agent: Twinkle/0.5
Content-Length: 0


--- (9 headers 0 lines)---
Sending to 192.168.10.16 : 5060 (non-NAT)
Transmitting (no NAT) to 192.168.10.16:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.16;rport;branch=z9hG4bKrrqpuppk;received=192.168.10.16
From: "16" <sip:[email protected]>;tag=nhzez
To: <sip:[email protected]>;tag=as3f36a918
Call-ID: [email protected]
CSeq: 852 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:[email protected]>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing


---
set_destination: Parsing <sip:[email protected]:5060> for address/port to send to
set_destination: set destination to 192.168.10.20, port 5060
We're at 192.168.10.1 port 18746
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x1 (g723) to SDP
Adding codec 0x100 (g729) to SDP
13 headers, 13 lines
Reliably Transmitting (no NAT) to 192.168.10.20:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK1d9fdd72
From: "16" <sip:[email protected]>;tag=as04d6e020
To: <sip:[email protected]>;tag=IPfP7FSeAp07cigB
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 104 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-asterisk-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 276

v=0
o=root 5499 5501 IN IP4 192.168.10.1
s=session
c=IN IP4 192.168.10.1
t=0 0
m=audio 18746 RTP/AVP 0 8 3 4 18
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=silenceSupp:off - - - -

---

<-- SIP read from 192.168.10.20:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK1d9fdd72
Call-ID: [email protected]
CSeq: 104 INVITE
From: "16" <sip:[email protected]>;tag=as04d6e020
To: <sip:[email protected]>;tag=IPfP7FSeAp07cigB
Contact: <sip:[email protected]:5060>
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
Supported: replaces
Content-Type: application/sdp
Content-Length: 139

v=0
o=- 81192499 69731988 IN IP4 192.168.10.20
s=SIP CALL
c=IN IP4 192.168.10.20
t=0 0
m=audio 17220 RTP/AVP 0
a=rtpmap:0 PCMU/8000

--- (11 headers 7 lines)---
Found RTP audio format 0
Peer audio RTP is at port 192.168.10.20:17220
Found description format PCMU
Capabilities: us - 0x10f (g723|gsm|ulaw|alaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
set_destination: Parsing <sip:[email protected]:5060> for address/port to send to
set_destination: set destination to 192.168.10.20, port 5060
Transmitting (no NAT) to 192.168.10.20:5060:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK3683ba3e
From: "16" <sip:[email protected]>;tag=as04d6e020
To: <sip:[email protected]>;tag=IPfP7FSeAp07cigB
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 104 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
set_destination: Parsing <sip:[email protected]:5060> for address/port to send to
set_destination: set destination to 192.168.10.20, port 5060
Reliably Transmitting (no NAT) to 192.168.10.20:5060:
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK2906a3f4
From: "16" <sip:[email protected]>;tag=as04d6e020
To: <sip:[email protected]>;tag=IPfP7FSeAp07cigB
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 105 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Destroying call '[email protected]'

<-- SIP read from 192.168.10.20:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK2906a3f4
Call-ID: [email protected]
CSeq: 105 BYE
From: "16" <sip:[email protected]>;tag=as04d6e020
To: <sip:[email protected]>;tag=IPfP7FSeAp07cigB
Contact: <sip:[email protected]:5060>
Content-Length: 0


--- (8 headers 0 lines)---
Destroying call '[email protected]'
 
Hi zusammen,

ich habe das jetzt so gelöst, dass ich meine Telefone mit IAX2 Images bestückt habe und siehe da schon gehts.

Warum das vorher nicht ging weiss ich allerdings immer noch nicht :(

Vielen Dank.

Ciao
Matze
 
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