Hallo Zusammen!
Ich bin gerade dabei meinen Asterisk per AstTAPI anzusprechen. Beim Klick in Outlook auf "Anruf beginnen" mit einer Rufnummer im Format +49 01234512345 passiert folgendes.
Der Sip Client 10 klingelt. Ich nehme ab.
Nach einer weile klingelt die angerufene nummer. Ich nehme ab.
In keine der Richtungen kann man was hören/sprechen
Sip read:
SIP/2.0 200 OK
CSeq: 103 INVITE
Call-ID: [email protected]
Contact: sip:217.115.134.46:5060
Content-Length: 234
Content-Type: application/sdp
From: "asterisk" <sip:[email protected]>;tag=as0e1b7cbf
Record-Route: <sip:[email protected];ftag=as0e1b7cbf;lr=on>
To: <sip:[email protected]>;tag=2594856171214291648677
Via: SIP/2.0/UDP 192.168.1.192:5060;branch=z9hG4bK7e8282d9;received=212.37.49.98;rport=25157
v=0
o=root 20664 20665 IN IP4 217.175.252.226
s=session
c=IN IP4 217.115.141.46
t=0 0
m=audio 13158 RTP/AVP 0 18
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=silenceSuppff - - - -
10 headers, 11 lines
Found RTP audio format 0
Found RTP audio format 18
Peer audio RTP is at port 217.115.141.46:13158
Found description format GSM
Found description format PCMA
Found description format PCMU
Found description format G729
Capabilities: us - 0x2(GSM), peer - audio=0x10e(GSM|ULAW|ALAW|G729A)/video=0x0(EMPTY), combined - 0x2(GSM)
Non-codec capabilities: us - 0x1(G723), peer - 0x0(EMPTY), combined - 0x0(EMPTY)
list_route: hop: <sip:[email protected];ftag=as0e1b7cbf;lr=on>
list_route: hop: <sip:217.115.11.46:5060>
set_destination: Parsing <sip:[email protected];ftag=as0e1b7cbf;lr=on> for address/port to send to
-- SIP/sipsnipout-65f5 answered SIP/10:-d2b1
set_destination: set destination to 212.227.232.145, port 5060
Transmitting:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.192:5060;branch=z9hG4bK6ba91049
Route: <sip:217.115.141.xx:5060>
From: "asterisk" <sip:[email protected]>;tag=as0e1b7cbf
To: <sip:[email protected]>;tag=2594856171214291648677
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 ACK
User-Agent: Asterisk PBX
Content-Length: 0
(no NAT) to 212.227.22.145:5060
-- Attempting native bridge of SIP/10:-d2b1 and SIP/sipsnipout-65f5
-- Attempting native bridge of SIP/10:-d2b1 and SIP/sipsnipout-65f5
______________
Soweit sieht das ja alles ganz gut aus, aber wie gesagt, man hört nichts
manager.conf:
sip.conf:
teil der extensions.conf
Ich habe unter Configure Asterisk TAPI driver alles wie folgt ausgefüllt:
Host: 192.168.1.192
Port: 5038
User: ab
Password: geheim
User Channel: SIP/10
Inbound Chan: leer
Dial by context: enabled
Context: TAPIwahl
Caller ID: 560
Attempt to set outgoing ID: disabled
Dial out by using the "Dial"applicatior: disabled
Outgouing Chan: sip/sipsnipout
Hat wer ideen, damit der Anruf klappt?
Falls noch Infos (Confs etc.) benötigt werden, kurz bescheid sagen
Ich bin gerade dabei meinen Asterisk per AstTAPI anzusprechen. Beim Klick in Outlook auf "Anruf beginnen" mit einer Rufnummer im Format +49 01234512345 passiert folgendes.
Der Sip Client 10 klingelt. Ich nehme ab.
Nach einer weile klingelt die angerufene nummer. Ich nehme ab.
In keine der Richtungen kann man was hören/sprechen
Sip read:
SIP/2.0 200 OK
CSeq: 103 INVITE
Call-ID: [email protected]
Contact: sip:217.115.134.46:5060
Content-Length: 234
Content-Type: application/sdp
From: "asterisk" <sip:[email protected]>;tag=as0e1b7cbf
Record-Route: <sip:[email protected];ftag=as0e1b7cbf;lr=on>
To: <sip:[email protected]>;tag=2594856171214291648677
Via: SIP/2.0/UDP 192.168.1.192:5060;branch=z9hG4bK7e8282d9;received=212.37.49.98;rport=25157
v=0
o=root 20664 20665 IN IP4 217.175.252.226
s=session
c=IN IP4 217.115.141.46
t=0 0
m=audio 13158 RTP/AVP 0 18
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=silenceSuppff - - - -
10 headers, 11 lines
Found RTP audio format 0
Found RTP audio format 18
Peer audio RTP is at port 217.115.141.46:13158
Found description format GSM
Found description format PCMA
Found description format PCMU
Found description format G729
Capabilities: us - 0x2(GSM), peer - audio=0x10e(GSM|ULAW|ALAW|G729A)/video=0x0(EMPTY), combined - 0x2(GSM)
Non-codec capabilities: us - 0x1(G723), peer - 0x0(EMPTY), combined - 0x0(EMPTY)
list_route: hop: <sip:[email protected];ftag=as0e1b7cbf;lr=on>
list_route: hop: <sip:217.115.11.46:5060>
set_destination: Parsing <sip:[email protected];ftag=as0e1b7cbf;lr=on> for address/port to send to
-- SIP/sipsnipout-65f5 answered SIP/10:-d2b1
set_destination: set destination to 212.227.232.145, port 5060
Transmitting:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.192:5060;branch=z9hG4bK6ba91049
Route: <sip:217.115.141.xx:5060>
From: "asterisk" <sip:[email protected]>;tag=as0e1b7cbf
To: <sip:[email protected]>;tag=2594856171214291648677
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 ACK
User-Agent: Asterisk PBX
Content-Length: 0
(no NAT) to 212.227.22.145:5060
-- Attempting native bridge of SIP/10:-d2b1 and SIP/sipsnipout-65f5
-- Attempting native bridge of SIP/10:-d2b1 and SIP/sipsnipout-65f5
______________
Soweit sieht das ja alles ganz gut aus, aber wie gesagt, man hört nichts
manager.conf:
Code:
;
; Asterisk Call Management support
;
[general]
enabled = yes
port = 5038
bindaddr = 0.0.0.0
[ab]
secret = geheim
permit = 0.0.0.0/0.0.0.0
read = system,call,log,verbose,command,agent,user
write = system,call,log,verbose,command,agent,user
[cd]
secret = geheim
permit = 192.168.1.0/255.255.255.0
read = system,call,log,verbose,command,agent,user
write = system,call,log,verbose,command,agent,user
;[mark]
;secret = mysecret
;deny=0.0.0.0/0.0.0.0
;permit=209.16.236.73/255.255.255.0
;read = system,call,log,verbose,command,agent,user
;write = system,call,log,verbose,command,agent,user
sip.conf:
Code:
;
[general]
port = 5060
bindaddr = 0.0.0.0
;externip = 212.37.xxx.xxx
Localnet = 255.255.0.0
srvlookup = yes
context = default
disallow=all
language=de
;Uebertragunsart
allow=alaw
allow=ulaw
allow=g726
allow=gsm
;//
canreinvite=no
insecure=very
nat=yes
dtmfmode=info
tos = reliability
maxexpirey = 3600
defaultexpirey = 1200
register => 123456:[email protected]/123456
register => 1234567:[email protected]/1234567
[sipsnipout]
type=friend
username=123456
secret=passwort
host=sipsnip.com
fromuser=123456
fromdomain=sipsnip.com
context=default
canreinvite=no
;qualify=yes
disallow=all
allow=gsm
insecure=very
nat=yes
dtmfmode=info
tos=0x18
[sipgateout]
type=friend
username=1234567
secret=geheim
host=sipgate.de
fromuser=1234567
fromdomain=sipgate.de
context=default
canreinvite=no
;qualify=yes
disallow=all
allow=gsm
insecure=very
nat=yes
dtmfmode=info
tos=0x18
[10]
type=friend
username=10
callerid=10
auth=md5
secret=T1
host=dynamic
reinvite=no
canreinvite=no
qualify=1200
dtmfmode=rfc2833
context=default
disallow=all
allow=ulaw
allow=alaw
allow=g726
allow=gsm
[11]
type=friend
username=11
secret=T2
host=dynamic
qualify=1200
dtmfmode=rfc2833
context= default
disallow=all
allow=ulaw
allow=alaw
teil der extensions.conf
Code:
(....)
[TAPIwahl]
exten => _x.,1,Dial(SIP/${EXTEN}@sipsnipout)
(...)
Ich habe unter Configure Asterisk TAPI driver alles wie folgt ausgefüllt:
Host: 192.168.1.192
Port: 5038
User: ab
Password: geheim
User Channel: SIP/10
Inbound Chan: leer
Dial by context: enabled
Context: TAPIwahl
Caller ID: 560
Attempt to set outgoing ID: disabled
Dial out by using the "Dial"applicatior: disabled
Outgouing Chan: sip/sipsnipout
Hat wer ideen, damit der Anruf klappt?
Falls noch Infos (Confs etc.) benötigt werden, kurz bescheid sagen