Hallo!
Viele Firmen haben ja eine Warteschleife oder Menüs mit Optionen, die zu einem Tastendruck auffordern. "z.B. wählen sie die 3 für ihren Kontostand.."
Momentan ist es so, das Verbindungen getrennt werden, sobald man in einem solchen Voicemenü eine beliebige Ziffer drückt
Woran mag das liegen? An Sipsnip? Oder an Asterisk? Wie kann ich das beheben?
v=0ux*CLI>
o=root 20664 20665 IN IP4 217.15.252.354
s=session
c=IN IP4 217.10.170.176
t=0 0
m=audio 16230 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 217.10.170.176:16230
Found description format GSM
Found description format PCMA
Found description format PCMU
Found description format G729
Capabilities: us - 0xa(GSM|ALAW), peer - audio=0x10e(GSM|ULAW|ALAW|G729A)/video=0x0(EMPTY), combined - 0xa(GSM|ALAW)
Non-codec capabilities: us - 0x1(G723), peer - 0x0(EMPTY), combined - 0x0(EMPTY)
list_route: hop: <sip:*33492xxxx0*[email protected];ftag=as78d9d50c;lr=on>
list_route: hop: <sip:217.160.170.76:5060>
-- SIP/sipsnipout-af7f answered CAPI[contr1/99]/14
set_destination: Parsing <sip:*3349xxx0*[email protected];ftag=as78d9d50c;lr=on> for address/port to send to
set_destination: set destination to 23.27.22.14, port 5060
Transmitting:
ACK sip:*334923xxx0*[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK0839ab52
Route: <sip:217.160.170.76:5060>
From: "14" <sip:[email protected]>;tag=as78d9d50c
To: <sip:*33492xxx0*[email protected]>;tag=3183094013873910713708
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 ACK
User-Agent: Asterisk PBX
Content-Length: 0
(no NAT) to 212.27.122.145:5060
set_destination: Parsing <sip:*334x0*[email protected];ftag=as78d9d50c;lr=on> for address/port to send to
set_destination: set destination to 212.227.22.145, port 5060
Reliably Transmitting:
INFO sip:217.160.170.176:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK664a5c82
Route: <sip:217.160.170.76:5060>
From: "14" <sip:[email protected]>;tag=as78d9d50c
To: <sip:*33492xxx40*[email protected]>;tag=31830940613873910713708
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 104 INFO
User-Agent: Asterisk PBX
Content-Type: application/dtmf-relay
Content-Length: 24
Signal=1
Duration=250
(no NAT) to 212.27.22.14:5060
linux*CLI>
Sip read:
SIP/2.0 400 Request cannot be handled at this time
Via: SIP/2.0/UDP 192.168.0.160:5060;rport=1112;received=22.37.49.198;branch=z9hG4bK664a5c82
From: "14" <sip:[email protected]>;tag=as78d9d50c
To: <sip:*3349xxx0*[email protected]>;tag=31830940613873910713708
Call-ID: [email protected]
CSeq: 104 INFO
6 headers, 0 lines
-- Got SIP response 400 "Request cannot be handled at this time" back from 212.227.22.145
== Spawn extension (DISAwahl, 023xxxx0, 10) exited non-zero on 'CAPI[contr1/99]/14'
-- CAPI Hangingup
-- removed pipe for PLCI = 0x101
Destroying call '[email protected]'
linux*CLI>
Viele Firmen haben ja eine Warteschleife oder Menüs mit Optionen, die zu einem Tastendruck auffordern. "z.B. wählen sie die 3 für ihren Kontostand.."
Momentan ist es so, das Verbindungen getrennt werden, sobald man in einem solchen Voicemenü eine beliebige Ziffer drückt
Woran mag das liegen? An Sipsnip? Oder an Asterisk? Wie kann ich das beheben?
v=0ux*CLI>
o=root 20664 20665 IN IP4 217.15.252.354
s=session
c=IN IP4 217.10.170.176
t=0 0
m=audio 16230 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 217.10.170.176:16230
Found description format GSM
Found description format PCMA
Found description format PCMU
Found description format G729
Capabilities: us - 0xa(GSM|ALAW), peer - audio=0x10e(GSM|ULAW|ALAW|G729A)/video=0x0(EMPTY), combined - 0xa(GSM|ALAW)
Non-codec capabilities: us - 0x1(G723), peer - 0x0(EMPTY), combined - 0x0(EMPTY)
list_route: hop: <sip:*33492xxxx0*[email protected];ftag=as78d9d50c;lr=on>
list_route: hop: <sip:217.160.170.76:5060>
-- SIP/sipsnipout-af7f answered CAPI[contr1/99]/14
set_destination: Parsing <sip:*3349xxx0*[email protected];ftag=as78d9d50c;lr=on> for address/port to send to
set_destination: set destination to 23.27.22.14, port 5060
Transmitting:
ACK sip:*334923xxx0*[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK0839ab52
Route: <sip:217.160.170.76:5060>
From: "14" <sip:[email protected]>;tag=as78d9d50c
To: <sip:*33492xxx0*[email protected]>;tag=3183094013873910713708
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 ACK
User-Agent: Asterisk PBX
Content-Length: 0
(no NAT) to 212.27.122.145:5060
set_destination: Parsing <sip:*334x0*[email protected];ftag=as78d9d50c;lr=on> for address/port to send to
set_destination: set destination to 212.227.22.145, port 5060
Reliably Transmitting:
INFO sip:217.160.170.176:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK664a5c82
Route: <sip:217.160.170.76:5060>
From: "14" <sip:[email protected]>;tag=as78d9d50c
To: <sip:*33492xxx40*[email protected]>;tag=31830940613873910713708
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 104 INFO
User-Agent: Asterisk PBX
Content-Type: application/dtmf-relay
Content-Length: 24
Signal=1
Duration=250
(no NAT) to 212.27.22.14:5060
linux*CLI>
Sip read:
SIP/2.0 400 Request cannot be handled at this time
Via: SIP/2.0/UDP 192.168.0.160:5060;rport=1112;received=22.37.49.198;branch=z9hG4bK664a5c82
From: "14" <sip:[email protected]>;tag=as78d9d50c
To: <sip:*3349xxx0*[email protected]>;tag=31830940613873910713708
Call-ID: [email protected]
CSeq: 104 INFO
6 headers, 0 lines
-- Got SIP response 400 "Request cannot be handled at this time" back from 212.227.22.145
== Spawn extension (DISAwahl, 023xxxx0, 10) exited non-zero on 'CAPI[contr1/99]/14'
-- CAPI Hangingup
-- removed pipe for PLCI = 0x101
Destroying call '[email protected]'
linux*CLI>