Hallo.
Ich versuche schon seit einer Woche Asterisk mit Tiscali zu verheiraten. Leider habe ich wohl noch eine kleine Hürde zu nehmen. Mit Xlite und Kphone funktioniert die Einwahl problemlos über tiscali. Nur bei asterisk bekomme ich folgende Fehlermeldung:
-- Executing Dial("SIP/chris-cc32", "SIP/{EXTEN:1}@tiscali||tr") in new stack
-- Called {EXTEN:1}@tiscali
May 23 16:12:45 WARNING[22046]: chan_sip.c:685 retrans_pkt: Maximum retries exceeded on call [email protected] for seqno 102 (Critical Request)
== No one is available to answer at this time
-- Executing Hangup("SIP/chris-cc32", "") in new stack
== Spawn extension (default, 2064xxxx, 2) exited non-zero on 'SIP/chris-cc32'
-- Executing Hangup("SIP/chris-cc32", "") in new stack
== Spawn extension (default, h, 1) exited non-zero on 'SIP/chris-cc32'
-- Saved useragent "kphone/4.1.0" for peer chris
Web.de und nikotel funktionieren auf meinem Asterisk problemlos.
bei sip debug peer tiscali kommt folgendes:
meine sip.conf Einstellungen:
vielen Dank.
Ich versuche schon seit einer Woche Asterisk mit Tiscali zu verheiraten. Leider habe ich wohl noch eine kleine Hürde zu nehmen. Mit Xlite und Kphone funktioniert die Einwahl problemlos über tiscali. Nur bei asterisk bekomme ich folgende Fehlermeldung:
-- Executing Dial("SIP/chris-cc32", "SIP/{EXTEN:1}@tiscali||tr") in new stack
-- Called {EXTEN:1}@tiscali
May 23 16:12:45 WARNING[22046]: chan_sip.c:685 retrans_pkt: Maximum retries exceeded on call [email protected] for seqno 102 (Critical Request)
== No one is available to answer at this time
-- Executing Hangup("SIP/chris-cc32", "") in new stack
== Spawn extension (default, 2064xxxx, 2) exited non-zero on 'SIP/chris-cc32'
-- Executing Hangup("SIP/chris-cc32", "") in new stack
== Spawn extension (default, h, 1) exited non-zero on 'SIP/chris-cc32'
-- Saved useragent "kphone/4.1.0" for peer chris
Web.de und nikotel funktionieren auf meinem Asterisk problemlos.
bei sip debug peer tiscali kommt folgendes:
Code:
SIP Debugging Enabled for IP: 62.26.64.24:5060
-- Executing Dial("SIP/chris-b183", "SIP/{EXTEN:1}@tiscali||tr") in new stack
We're at 83.129.xx.xxx port 16336
Answering/Requesting with root capability 4
12 headers, 8 lines
Reliably Transmitting:
INVITE sip:{EXTEN:1}@tiscali.de SIP/2.0
Via: SIP/2.0/UDP 83.129.xx.xxx:5060;branch=z9hG4bK14b762a9;rport
From: "username" <sip:[email protected]>;tag=as24449d2a
To: <sip:{EXTEN:1}@tiscali.de>
Contact: <sip:[email protected]>
Call-ID: [email][email protected][/email]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Mon, 23 May 2005 14:18:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 162
v=0
o=root 22046 22046 IN IP4 83.129.xx.xxx
s=session
c=IN IP4 83.129.xx.xxx
t=0 0
m=audio 16336 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
(NAT) to 62.26.64.24:5060
-- Called {EXTEN:1}@tiscali
Retransmitting #1 (NAT):
INVITE sip:{EXTEN:1}@tiscali.de SIP/2.0
Via: SIP/2.0/UDP 83.129.xx.xxx:5060;branch=z9hG4bK14b762a9;rport
From: "username" <sip:[email protected]>;tag=as24449d2a
To: <sip:{EXTEN:1}@tiscali.de>
Contact: <sip:[email protected]>
Call-ID: [email][email protected][/email]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Mon, 23 May 2005 14:18:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 162
v=0
o=root 22046 22046 IN IP4 83.129.xx.xxx
s=session
c=IN IP4 83.129.xx.xxx
t=0 0
m=audio 16336 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
to 62.26.64.24:5060
Retransmitting #2 (NAT):
INVITE sip:{EXTEN:1}@tiscali.de SIP/2.0
Via: SIP/2.0/UDP 83.129.xx.xxx:5060;branch=z9hG4bK14b762a9;rport
From: "username" <sip:[email protected]>;tag=as24449d2a
To: <sip:{EXTEN:1}@tiscali.de>
Contact: <sip:[email protected]>
Call-ID: [email][email protected][/email]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Mon, 23 May 2005 14:18:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 162
v=0
o=root 22046 22046 IN IP4 83.129.xx.xxx
s=session
c=IN IP4 83.129.xx.xxx
t=0 0
m=audio 16336 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
to 62.26.64.24:5060
Retransmitting #3 (NAT):
INVITE sip:{EXTEN:1}@tiscali.de SIP/2.0
Via: SIP/2.0/UDP 83.129.xx.xxx:5060;branch=z9hG4bK14b762a9;rport
From: "username" <sip:[email protected]>;tag=as24449d2a
To: <sip:{EXTEN:1}@tiscali.de>
Contact: <sip:[email protected]>
Call-ID: [email][email protected][/email]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Mon, 23 May 2005 14:18:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 162
v=0
o=root 22046 22046 IN IP4 83.129.xx.xxx
s=session
c=IN IP4 83.129.xx.xxx
t=0 0
m=audio 16336 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
to 62.26.64.24:5060
Retransmitting #4 (NAT):
INVITE sip:{EXTEN:1}@tiscali.de SIP/2.0
Via: SIP/2.0/UDP 83.129.xx.xxx:5060;branch=z9hG4bK14b762a9;rport
From: "username" <sip:[email protected]>;tag=as24449d2a
To: <sip:{EXTEN:1}@tiscali.de>
Contact: <sip:[email protected]>
Call-ID: [email][email protected][/email]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Mon, 23 May 2005 14:18:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 162
v=0
o=root 22046 22046 IN IP4 83.129.xx.xxx
s=session
c=IN IP4 83.129.xx.xxx
t=0 0
m=audio 16336 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
to 62.26.64.24:5060
Retransmitting #5 (NAT):
INVITE sip:{EXTEN:1}@tiscali.de SIP/2.0
Via: SIP/2.0/UDP 83.129.xx.xxx:5060;branch=z9hG4bK14b762a9;rport
From: "username" <sip:[email protected]>;tag=as24449d2a
To: <sip:{EXTEN:1}@tiscali.de>
Contact: <sip:[email protected]>
Call-ID: [email][email protected][/email]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Mon, 23 May 2005 14:18:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 162
v=0
o=root 22046 22046 IN IP4 83.129.xx.xxx
s=session
c=IN IP4 83.129.xx.xxx
t=0 0
m=audio 16336 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
to 62.26.64.24:5060
May 23 16:18:59 WARNING[22046]: chan_sip.c:685 retrans_pkt: Maximum retries exceeded on call [email][email protected][/email] for seqno 102 (Critical Request)
== No one is available to answer at this time
-- Executing Hangup("SIP/chris-b183", "") in new stack
== Spawn extension (default, 2064356437, 2) exited non-zero on 'SIP/chris-b183'
-- Executing Hangup("SIP/chris-b183", "") in new stack
== Spawn extension (default, h, 1) exited non-zero on 'SIP/chris-b183'
Destroying call '[email protected]'meine sip.conf Einstellungen:
Code:
[general]
bindaddr = 0.0.0.0
localnet = 192.168.1.0/24
externip = dyndns.org
port = 5060
context = default
maxexpirey = 3600
defaultexpirey = 600
srvlookup = yes
tos = 0x18
disallow = all
allow = gsm
allow = alaw
allow = ulaw
allow = g729
[tiscali]
type=friend
host=tiscali.de
secret=
username=
fromdomain=tiscali.de
fromuser=
context=incoming
canreinvite=no
qualify=no
disallow=all
allow=ulaw
;allow=g729
insecure=very
nat=yes
dtmfmode=info
tos=0x18vielen Dank.
