Ringing() leads into sound problem, Answer() works ...!?

Asterisk Asterisk Support
1

Dear Forum ,

I have my Asterisk 1.6 in a LAN ( nat ) enviroment, registered to a external sip provider. Within the LAN I have a softphone for testing …

If I use the following statement in the extensions.conf , the external caller , calling a number of the sip provider, who is beeing connected to the internal softphone - cant hear anything :

exten =>061XXXXXX,n,Ringing()
exten =>061XXXXXX,n,Dial(SIP/90,100)

It doesnt happen all the time but more or less in 50% of the tries. The called party( softphone) can hear allways the caller.

If I change the the Rining() statement to an Answer() , it works allways !!! Whats wrong with ringing ? How can I avoid this ?

thanks for help !

regards,
Filip

2

You will need to provide more debug information.

Please copy paste a “sip set debug on” debug of a call.

3

ok here we go:

( the asterisk has the local ip 192.168.1.87 and it listen on 5061 for the local sip phone b/y the 5060 is blocked from another service but this should not have somthing to do with this problem , b/c with answer everything works fine…)

[Aug 31 09:40:16] WARNING[2598] chan_capi.c: config reload is not supported yet.
[Aug 31 09:40:16] NOTICE[2598] chan_iax2.c: Ignoring bindport on reload
[Aug 31 09:40:16] NOTICE[2598] chan_iax2.c: Ignoring bindaddr on reload
[Aug 31 09:40:38] VERBOSE[3135] logger.c:
<— SIP read from UDP://192.168.1.48:2473 —>

<------------->
[Aug 31 09:40:49] VERBOSE[3135] logger.c: Really destroying SIP dialog ‘6bfe5bf13ddbc29e3ecf13c57de7e688@192.168.1.187’ Method: REGISTER
[Aug 31 09:41:08] VERBOSE[3135] logger.c:
<— SIP read from UDP://192.168.1.48:2473 —>

<------------->
[Aug 31 09:41:12] VERBOSE[3135] logger.c:
<— SIP read from UDP://83.125.8.71:5060 —>
INVITE sip:0611XXXXXXX@79.254.41.219:5061 SIP/2.0
Record-Route: sip:83.125.8.71;lr=on;ftag=as79d79fce;nat=yes;dus-rr=fbd.b44
Via: SIP/2.0/UDP 83.125.8.71;branch=z9hG4bKf8ca.56b5e3c.0
Via: SIP/2.0/UDP 83.125.8.42:5060;rport=5060;branch=z9hG4bK55669e4c
Max-Forwards: 70
From: “0177XXXXXXX” sip:0177XXXXXXX@dus.net;tag=as79d79fce
To: sip:0611XXXXXXX@83.125.8.71
Contact: sip:0177XXXXXXX@83.125.8.42:5060
Call-ID: 47d2b2f653ff1cfc1dwd08b61fc96ac2@dus.net
CSeq: 102 INVITE
User-Agent: DUS.NET INGRESS-APP-ENGINE 10.3.1
Date: Fri, 31 Aug 2012 09:41:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Diversion: sip:+49611XXXXXXXXX@dus.net;user=phone;reason=unconditional
Content-Type: application/sdp
Content-Length: 481
DUS-Rgid: 4

v=0
o=root 660973538 660973538 IN IP4 83.125.8.42
s=DUS.NET INGRESS-APP-ENGINE 10.3.1
c=IN IP4 83.125.8.156
t=0 0
m=audio 24438 RTP/AVP 8 0 18 111 112 3 110 97 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:111 G726-32/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------->
[Aug 31 09:41:12] VERBOSE[3135] logger.c: — (18 headers 20 lines) —
[Aug 31 09:41:12] VERBOSE[3135] logger.c: Sending to 83.125.8.71 : 5060 (NAT)
[Aug 31 09:41:12] VERBOSE[3135] logger.c: Using INVITE request as basis request - 47d2b2f653ff1cfc1dwd08b61fc96ac2@dus.net
[Aug 31 09:41:12] VERBOSE[3135] logger.c: No user ‘0177XXXXXXX’ in SIP users list
[Aug 31 09:41:12] VERBOSE[3135] logger.c: Found peer ‘dusnet-duro’ for ‘0177XXXXXXX’ from 83.125.8.71:5060
[Aug 31 09:41:12] VERBOSE[3135] logger.c: Found RTP audio format 8
[Aug 31 09:41:12] VERBOSE[3135] logger.c: Found RTP audio format 0
[Aug 31 09:41:12] VERBOSE[3135] logger.c: Found RTP audio format 18
[Aug 31 09:41:12] VERBOSE[3135] logger.c: Found RTP audio format 111
[Aug 31 09:41:12] VERBOSE[3135] logger.c: Found RTP audio format 112
[Aug 31 09:41:12] VERBOSE[3135] logger.c: Found RTP audio format 3
[Aug 31 09:41:12] VERBOSE[3135] logger.c: Found RTP audio format 110
[Aug 31 09:41:12] VERBOSE[3135] logger.c: Found RTP audio format 97
[Aug 31 09:41:12] VERBOSE[3135] logger.c: Found RTP audio format 101
[Aug 31 09:41:12] VERBOSE[3135] logger.c: Peer audio RTP is at port 83.125.8.156:24438
[Aug 31 09:41:12] VERBOSE[3135] logger.c: Found audio description format PCMA for ID 8
[Aug 31 09:41:12] VERBOSE[3135] logger.c: Found audio description format PCMU for ID 0
[Aug 31 09:41:12] VERBOSE[3135] logger.c: Found audio description format G729 for ID 18
[Aug 31 09:41:12] VERBOSE[3135] logger.c: Found audio description format G726-32 for ID 111
[Aug 31 09:41:12] VERBOSE[3135] logger.c: Found audio description format AAL2-G726-32 for ID 112
[Aug 31 09:41:12] VERBOSE[3135] logger.c: Found audio description format GSM for ID 3
[Aug 31 09:41:12] VERBOSE[3135] logger.c: Found audio description format speex for ID 110
[Aug 31 09:41:12] VERBOSE[3135] logger.c: Found audio description format iLBC for ID 97
[Aug 31 09:41:12] VERBOSE[3135] logger.c: Found audio description format telephone-event for ID 101
[Aug 31 09:41:12] VERBOSE[3135] logger.c: Capabilities: us - 0x4 (ulaw), peer - audio=0xf1e (gsm|ulaw|alaw|g726|g729|speex|ilbc|g726aal2)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
[Aug 31 09:41:12] VERBOSE[3135] logger.c: Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event), combined - 0x0 (nothing)
[Aug 31 09:41:12] VERBOSE[3135] logger.c: Peer audio RTP is at port 83.125.8.156:24438
[Aug 31 09:41:12] VERBOSE[3135] logger.c: Looking for 0611XXXXXXX in DUSNET-IN (domain 79.254.41.219)
[Aug 31 09:41:12] VERBOSE[3135] logger.c: RDNIS for this call is is +496111XXXXXXXX (reason unconditional)
[Aug 31 09:41:12] VERBOSE[3135] logger.c: list_route: hop: sip:83.125.8.71;lr=on;ftag=as79d79fce;nat=yes;dus-rr=fbd.b44
[Aug 31 09:41:12] VERBOSE[3135] logger.c:
<— Transmitting (NAT) to 83.125.8.71:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 83.125.8.71;branch=z9hG4bKf8ca.56b5e3c.0;received=83.125.8.71
Via: SIP/2.0/UDP 83.125.8.42:5060;rport=5060;branch=z9hG4bK55669e4c
Record-Route: sip:83.125.8.71;lr=on;ftag=as79d79fce;nat=yes;dus-rr=fbd.b44
From: “0177XXXXXXX” sip:0177XXXXXXX@dus.net;tag=as79d79fce
To: sip:0611XXXXXXX@83.125.8.71
Call-ID: 47d2b2f653ff1cfc1dwd08b61fc96ac2@dus.net
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: sip:0611XXXXXXX@192.168.1.187:5061
Content-Length: 0

<------------>
[Aug 31 09:41:12] VERBOSE[3323] logger.c:
<— Transmitting (NAT) to 83.125.8.71:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 83.125.8.71;branch=z9hG4bKf8ca.56b5e3c.0;received=83.125.8.71
Via: SIP/2.0/UDP 83.125.8.42:5060;rport=5060;branch=z9hG4bK55669e4c
Record-Route: sip:83.125.8.71;lr=on;ftag=as79d79fce;nat=yes;dus-rr=fbd.b44
From: “0177XXXXXXX” sip:0177XXXXXXX@dus.net;tag=as79d79fce
To: sip:0611XXXXXXX@83.125.8.71;tag=as775e96b0
Call-ID: 47d2b2f653ff1cfc1dwd08b61fc96ac2@dus.net
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: sip:0611XXXXXXX@192.168.1.187:5061
Content-Length: 0

<------------>
[Aug 31 09:41:12] VERBOSE[3323] logger.c: Audio is at 192.168.1.187 port 16168
[Aug 31 09:41:12] VERBOSE[3323] logger.c: Adding codec 0x4 (ulaw) to SDP
[Aug 31 09:41:12] VERBOSE[3323] logger.c: Adding codec 0x2 (gsm) to SDP
[Aug 31 09:41:12] VERBOSE[3323] logger.c: Reliably Transmitting (no NAT) to 192.168.1.48:2473:
INVITE sip:90@192.168.1.48:2473;rinstance=13b5fa9b849e3e27 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.187:5061;branch=z9hG4bK2a84f7d0;rport
Max-Forwards: 70
From: “0177XXXXXXX” sip:0177XXXXXXX@192.168.1.187:5061;tag=as1546f58c
To: sip:90@192.168.1.48:2473;rinstance=13b5fa9b849e3e27
Contact: sip:0177XXXXXXX@192.168.1.187:5061
Call-ID: 388f2d041410952d063d080d26558b7b@192.168.1.187
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Fri, 31 Aug 2012 09:41:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 230

v=0
o=root 900312370 900312370 IN IP4 192.168.1.187
s=Asterisk PBX 1.6.0.1
c=IN IP4 192.168.1.187
t=0 0
m=audio 16168 RTP/AVP 0 3
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

[Aug 31 09:41:12] VERBOSE[3135] logger.c:
<— SIP read from UDP://192.168.1.48:2473 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.187:5061;branch=z9hG4bK2a84f7d0;rport=5061
Contact: sip:90@192.168.1.48:2473;rinstance=13b5fa9b849e3e27
To: sip:90@192.168.1.48:2473;rinstance=13b5fa9b849e3e27;tag=f330d935
From: "0177XXXXXXX"sip:0177XXXXXXX@192.168.1.187:5061;tag=as1546f58c
Call-ID: 388f2d041410952d063d080d26558b7b@192.168.1.187
CSeq: 102 INVITE
User-Agent: 3CXPhone 6.0.20943.0
Content-Length: 0

<------------->
[Aug 31 09:41:12] VERBOSE[3135] logger.c: — (9 headers 0 lines) —
[Aug 31 09:41:12] VERBOSE[3323] logger.c:
<— Transmitting (NAT) to 83.125.8.71:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 83.125.8.71;branch=z9hG4bKf8ca.56b5e3c.0;received=83.125.8.71
Via: SIP/2.0/UDP 83.125.8.42:5060;rport=5060;branch=z9hG4bK55669e4c
Record-Route: sip:83.125.8.71;lr=on;ftag=as79d79fce;nat=yes;dus-rr=fbd.b44
From: “0177XXXXXXX” sip:0177XXXXXXX@dus.net;tag=as79d79fce
To: sip:0611XXXXXXX@83.125.8.71;tag=as775e96b0
Call-ID: 47d2b2f653ff1cfc1dwd08b61fc96ac2@dus.net
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: sip:0611XXXXXXX@192.168.1.187:5061
Content-Length: 0

<------------>
[Aug 31 09:41:14] VERBOSE[3135] logger.c:
<— SIP read from UDP://192.168.1.48:2473 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.187:5061;branch=z9hG4bK2a84f7d0;rport=5061
Contact: sip:90@192.168.1.48:2473;rinstance=13b5fa9b849e3e27
To: sip:90@192.168.1.48:2473;rinstance=13b5fa9b849e3e27;tag=f330d935
From: "0177XXXXXXX"sip:0177XXXXXXX@192.168.1.187:5061;tag=as1546f58c
Call-ID: 388f2d041410952d063d080d26558b7b@192.168.1.187
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhone 6.0.20943.0
Content-Length: 176

v=0
o=3cxVCE 104957190 315772755 IN IP4 192.168.1.48
s=3cxVCE Audio Call
c=IN IP4 192.168.1.48
t=0 0
m=audio 40018 RTP/AVP 0 3
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000

<------------->
[Aug 31 09:41:14] VERBOSE[3135] logger.c: — (12 headers 8 lines) —
[Aug 31 09:41:14] VERBOSE[3135] logger.c: Found RTP audio format 0
[Aug 31 09:41:14] VERBOSE[3135] logger.c: Found RTP audio format 3
[Aug 31 09:41:14] VERBOSE[3135] logger.c: Peer audio RTP is at port 192.168.1.48:40018
[Aug 31 09:41:14] VERBOSE[3135] logger.c: Found audio description format PCMU for ID 0
[Aug 31 09:41:14] VERBOSE[3135] logger.c: Found audio description format GSM for ID 3
[Aug 31 09:41:14] VERBOSE[3135] logger.c: Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x6 (gsm|ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x6 (gsm|ulaw)
[Aug 31 09:41:14] VERBOSE[3135] logger.c: Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
[Aug 31 09:41:14] VERBOSE[3135] logger.c: Peer audio RTP is at port 192.168.1.48:40018
[Aug 31 09:41:14] VERBOSE[3135] logger.c: list_route: hop: sip:90@192.168.1.48:2473;rinstance=13b5fa9b849e3e27
[Aug 31 09:41:14] VERBOSE[3135] logger.c: set_destination: Parsing sip:90@192.168.1.48:2473;rinstance=13b5fa9b849e3e27 for address/port to send to
[Aug 31 09:41:14] VERBOSE[3135] logger.c: set_destination: set destination to 192.168.1.48, port 2473
[Aug 31 09:41:14] VERBOSE[3135] logger.c: Transmitting (no NAT) to 192.168.1.48:2473:
ACK sip:90@192.168.1.48:2473;rinstance=13b5fa9b849e3e27 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.187:5061;branch=z9hG4bK43a87344;rport
Max-Forwards: 70
From: “0177XXXXXXX” sip:0177XXXXXXX@192.168.1.187:5061;tag=as1546f58c
To: sip:90@192.168.1.48:2473;rinstance=13b5fa9b849e3e27;tag=f330d935
Contact: sip:0177XXXXXXX@192.168.1.187:5061
Call-ID: 388f2d041410952d063d080d26558b7b@192.168.1.187
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

[Aug 31 09:41:14] VERBOSE[3323] logger.c: Audio is at 192.168.1.187 port 16690
[Aug 31 09:41:14] VERBOSE[3323] logger.c: Adding codec 0x4 (ulaw) to SDP
[Aug 31 09:41:14] VERBOSE[3323] logger.c:
<— Reliably Transmitting (NAT) to 83.125.8.71:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 83.125.8.71;branch=z9hG4bKf8ca.56b5e3c.0;received=83.125.8.71
Via: SIP/2.0/UDP 83.125.8.42:5060;rport=5060;branch=z9hG4bK55669e4c
Record-Route: sip:83.125.8.71;lr=on;ftag=as79d79fce;nat=yes;dus-rr=fbd.b44
From: “0177XXXXXXX” sip:0177XXXXXXX@dus.net;tag=as79d79fce
To: sip:0611XXXXXXX@83.125.8.71;tag=as775e96b0
Call-ID: 47d2b2f653ff1cfc1dwd08b61fc96ac2@dus.net
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: sip:0611XXXXXXX@192.168.1.187:5061
Content-Type: application/sdp
Content-Length: 207

v=0
o=root 657240805 657240805 IN IP4 192.168.1.187
s=Asterisk PBX 1.6.0.1
c=IN IP4 192.168.1.187
t=0 0
m=audio 16690 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
[Aug 31 09:41:14] VERBOSE[3135] logger.c:
<— SIP read from UDP://83.125.8.71:5060 —>
ACK sip:0611XXXXXXX@79.254.41.219:5061 SIP/2.0
Via: SIP/2.0/UDP 83.125.8.71;branch=z9hG4bKcydzigwkX
Via: SIP/2.0/UDP 83.125.8.42:5060;rport=5060;branch=z9hG4bK6fb8a3ec
Max-Forwards: 70
From: “0177XXXXXXX” sip:0177XXXXXXX@dus.net;tag=as79d79fce
To: sip:0611XXXXXXX@83.125.8.71;tag=as775e96b0
Contact: sip:0177XXXXXXX@83.125.8.42:5060
Call-ID: 47d2b2f653ff1cfc1dwd08b61fc96ac2@dus.net
CSeq: 102 ACK
User-Agent: DUS.NET INGRESS-APP-ENGINE 10.3.1
Content-Length: 0

<------------->
[Aug 31 09:41:14] VERBOSE[3135] logger.c: — (11 headers 0 lines) —
[Aug 31 09:41:17] VERBOSE[3135] logger.c: Reliably Transmitting (NAT) to 83.125.8.83:5060:
OPTIONS sip:taris.dus.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.187:5061;branch=z9hG4bK1f81ba9b;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@192.168.1.187;tag=as4a8d721a
To: sip:taris.dus.net
Contact: sip:asterisk@192.168.1.187
Call-ID: 6b9f7e6b7ca5e4214a69758b55a33055@192.168.1.187
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Fri, 31 Aug 2012 09:41:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0

[Aug 31 09:41:17] VERBOSE[3135] logger.c:
<— SIP read from UDP://83.125.8.83:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.187:5061;branch=z9hG4bK1f81ba9b;rport=5061;received=79.254.41.219
From: “asterisk” sip:asterisk@192.168.1.187;tag=as4a8d721a
To: sip:taris.dus.net;tag=75ccf2b892ef0d374e533287500329e6.1600
Call-ID: 6b9f7e6b7ca5e4214a69758b55a33055@192.168.1.187
CSeq: 102 OPTIONS
Server: DUS.NET TERM-SIP-ENGINE 2.1
Content-Length: 0

<------------->
[Aug 31 09:41:17] VERBOSE[3135] logger.c: — (8 headers 0 lines) —
[Aug 31 09:41:17] VERBOSE[3135] logger.c: Really destroying SIP dialog ‘6b9f7e6b7ca5e4214a69758b55a33055@192.168.1.187’ Method: OPTIONS
[Aug 31 09:41:17] VERBOSE[3135] logger.c: Reliably Transmitting (NAT) to 83.125.8.71:5060:
OPTIONS sip:duro.dus.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.187:5061;branch=z9hG4bK452f9dd4;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@192.168.1.187;tag=as307f4363
To: sip:duro.dus.net
Contact: sip:asterisk@192.168.1.187
Call-ID: 38a0f1207a1dbed112e86b48405ec03a@192.168.1.187
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Fri, 31 Aug 2012 09:41:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0

[Aug 31 09:41:17] VERBOSE[3135] logger.c:
<— SIP read from UDP://83.125.8.71:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.187:5061;branch=z9hG4bK452f9dd4;rport=5061;received=79.254.41.219
From: “asterisk” sip:asterisk@192.168.1.187;tag=as307f4363
To: sip:duro.dus.net;tag=75ccf2b892ef0d374e533287500329e6.3995
Call-ID: 38a0f1207a1dbed112e86b48405ec03a@192.168.1.187
CSeq: 102 OPTIONS
Server: DUS.NET TERM-SIP-ENGINE 2.1
Content-Length: 0

<------------->
[Aug 31 09:41:17] VERBOSE[3135] logger.c: — (8 headers 0 lines) —
[Aug 31 09:41:17] VERBOSE[3135] logger.c: Really destroying SIP dialog ‘38a0f1207a1dbed112e86b48405ec03a@192.168.1.187’ Method: OPTIONS
[Aug 31 09:41:17] VERBOSE[3135] logger.c: Reliably Transmitting (NAT) to 83.125.8.71:5060:
OPTIONS sip:helios.dus.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.187:5061;branch=z9hG4bK08a617a2;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@192.168.1.187;tag=as02ca6551
To: sip:helios.dus.net
Contact: sip:asterisk@192.168.1.187
Call-ID: 606d658417163ae75ec2e5530e49cb9d@192.168.1.187
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Fri, 31 Aug 2012 09:41:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0

[Aug 31 09:41:17] VERBOSE[3135] logger.c:
<— SIP read from UDP://83.125.8.71:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.187:5061;branch=z9hG4bK08a617a2;rport=5061;received=79.254.41.219
From: “asterisk” sip:asterisk@192.168.1.187;tag=as02ca6551
To: sip:helios.dus.net;tag=75ccf2b892ef0d374e533287500329e6.daac
Call-ID: 606d658417163ae75ec2e5530e49cb9d@192.168.1.187
CSeq: 102 OPTIONS
Server: DUS.NET TERM-SIP-ENGINE 2.1
Content-Length: 0

<------------->
[Aug 31 09:41:17] VERBOSE[3135] logger.c: — (8 headers 0 lines) —
[Aug 31 09:41:17] VERBOSE[3135] logger.c: Really destroying SIP dialog ‘606d658417163ae75ec2e5530e49cb9d@192.168.1.187’ Method: OPTIONS
[Aug 31 09:41:17] VERBOSE[3135] logger.c: Reliably Transmitting (NAT) to 83.125.8.71:5060:
OPTIONS sip:zelos.dus.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.187:5061;branch=z9hG4bK05a24733;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@192.168.1.187;tag=as03643e0a
To: sip:zelos.dus.net
Contact: sip:asterisk@192.168.1.187
Call-ID: 6ddb150c50302d99657723254b4dc6e3@192.168.1.187
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Fri, 31 Aug 2012 09:41:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0

[Aug 31 09:41:17] VERBOSE[3135] logger.c:
<— SIP read from UDP://83.125.8.71:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.187:5061;branch=z9hG4bK05a24733;rport=5061;received=79.254.41.219
From: “asterisk” sip:asterisk@192.168.1.187;tag=as03643e0a
To: sip:zelos.dus.net;tag=75ccf2b892ef0d374e533287500329e6.b665
Call-ID: 6ddb150c50302d99657723254b4dc6e3@192.168.1.187
CSeq: 102 OPTIONS
Server: DUS.NET TERM-SIP-ENGINE 2.1
Content-Length: 0

<------------->
[Aug 31 09:41:17] VERBOSE[3135] logger.c: — (8 headers 0 lines) —
[Aug 31 09:41:17] VERBOSE[3135] logger.c: Really destroying SIP dialog ‘6ddb150c50302d99657723254b4dc6e3@192.168.1.187’ Method: OPTIONS
[Aug 31 09:41:17] VERBOSE[3135] logger.c: Reliably Transmitting (NAT) to 83.125.8.71:5060:
OPTIONS sip:talos.dus.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.187:5061;branch=z9hG4bK18ed759b;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@192.168.1.187;tag=as0d83e438
To: sip:talos.dus.net
Contact: sip:asterisk@192.168.1.187
Call-ID: 51f7671258346f1f7e89fa9a475be9f6@192.168.1.187
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Fri, 31 Aug 2012 09:41:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0

[Aug 31 09:41:17] VERBOSE[3135] logger.c:
<— SIP read from UDP://83.125.8.71:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.187:5061;branch=z9hG4bK18ed759b;rport=5061;received=79.254.41.219
From: “asterisk” sip:asterisk@192.168.1.187;tag=as0d83e438
To: sip:talos.dus.net;tag=75ccf2b892ef0d374e533287500329e6.4c98
Call-ID: 51f7671258346f1f7e89fa9a475be9f6@192.168.1.187
CSeq: 102 OPTIONS
Server: DUS.NET TERM-SIP-ENGINE 2.1
Content-Length: 0

<------------->
[Aug 31 09:41:17] VERBOSE[3135] logger.c: — (8 headers 0 lines) —
[Aug 31 09:41:17] VERBOSE[3135] logger.c: Really destroying SIP dialog ‘51f7671258346f1f7e89fa9a475be9f6@192.168.1.187’ Method: OPTIONS
[Aug 31 09:41:17] VERBOSE[3135] logger.c: Reliably Transmitting (NAT) to 83.125.8.71:5060:
OPTIONS sip:voip.dus.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.187:5061;branch=z9hG4bK7389b13c;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@192.168.1.187;tag=as77f1a5bf
To: sip:voip.dus.net
Contact: sip:asterisk@192.168.1.187
Call-ID: 1930c1cb405d74b0537e59e11e0bcbfb@192.168.1.187
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Fri, 31 Aug 2012 09:41:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0

[Aug 31 09:41:17] VERBOSE[3135] logger.c:
<— SIP read from UDP://83.125.8.71:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.187:5061;branch=z9hG4bK7389b13c;rport=5061;received=79.254.41.219
From: “asterisk” sip:asterisk@192.168.1.187;tag=as77f1a5bf
To: sip:voip.dus.net;tag=75ccf2b892ef0d374e533287500329e6.68aa
Call-ID: 1930c1cb405d74b0537e59e11e0bcbfb@192.168.1.187
CSeq: 102 OPTIONS
Server: DUS.NET TERM-SIP-ENGINE 2.1
Content-Length: 0

<------------->
[Aug 31 09:41:17] VERBOSE[3135] logger.c: — (8 headers 0 lines) —
[Aug 31 09:41:17] VERBOSE[3135] logger.c: Really destroying SIP dialog ‘1930c1cb405d74b0537e59e11e0bcbfb@192.168.1.187’ Method: OPTIONS
[Aug 31 09:41:27] VERBOSE[3135] logger.c:
<— SIP read from UDP://192.168.1.48:2473 —>
REGISTER sip:192.168.1.187:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.48:2473;branch=z9hG4bK-d8754z-d758c869514c931e-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:90@192.168.1.48:2473;rinstance=13b5fa9b849e3e27
To: "90"sip:90@192.168.1.187:5061
From: "90"sip:90@192.168.1.187:5061;tag=d267760b
Call-ID: N2VkNjNhNzczNTY0ODE2ZTUxMTk2YzlhNjJkYTczZTE.
CSeq: 100 REGISTER
Expires: 120
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Supported: replaces
User-Agent: 3CXPhone 6.0.20943.0
Authorization: Digest username=“90”,realm=“asterisk”,nonce=“1d4482dd”,uri=“sip:192.168.1.187:5061”,response=“77b5f607b601718a95d286b3b4891181”,algorithm=MD5
Content-Length: 0

<------------->
[Aug 31 09:41:27] VERBOSE[3135] logger.c: — (14 headers 0 lines) —
[Aug 31 09:41:27] VERBOSE[3135] logger.c: Sending to 192.168.1.48 : 2473 (NAT)
[Aug 31 09:41:27] VERBOSE[3135] logger.c:
<— Transmitting (no NAT) to 192.168.1.48:2473 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.48:2473;branch=z9hG4bK-d8754z-d758c869514c931e-1—d8754z-;received=192.168.1.48;rport=2473
From: "90"sip:90@192.168.1.187:5061;tag=d267760b
To: "90"sip:90@192.168.1.187:5061;tag=as2fcf6d61
Call-ID: N2VkNjNhNzczNTY0ODE2ZTUxMTk2YzlhNjJkYTczZTE.
CSeq: 100 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="63b136fa"
Content-Length: 0

<------------>
[Aug 31 09:41:27] VERBOSE[3135] logger.c: Scheduling destruction of SIP dialog ‘N2VkNjNhNzczNTY0ODE2ZTUxMTk2YzlhNjJkYTczZTE.’ in 32000 ms (Method: REGISTER)
[Aug 31 09:41:27] VERBOSE[3135] logger.c:
<— SIP read from UDP://192.168.1.48:2473 —>
REGISTER sip:192.168.1.187:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.48:2473;branch=z9hG4bK-d8754z-6e47325f4d62920c-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:90@192.168.1.48:2473;rinstance=13b5fa9b849e3e27
To: "90"sip:90@192.168.1.187:5061
From: "90"sip:90@192.168.1.187:5061;tag=d267760b
Call-ID: N2VkNjNhNzczNTY0ODE2ZTUxMTk2YzlhNjJkYTczZTE.
CSeq: 101 REGISTER
Expires: 120
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Supported: replaces
User-Agent: 3CXPhone 6.0.20943.0
Authorization: Digest username=“90”,realm=“asterisk”,nonce=“63b136fa”,uri=“sip:192.168.1.187:5061”,response=“544319977ede19e5a00c45d63a48f6fd”,algorithm=MD5
Content-Length: 0

<------------->
[Aug 31 09:41:27] VERBOSE[3135] logger.c: — (14 headers 0 lines) —
[Aug 31 09:41:27] VERBOSE[3135] logger.c: Sending to 192.168.1.48 : 2473 (NAT)
[Aug 31 09:41:27] VERBOSE[3135] logger.c:
<— Transmitting (no NAT) to 192.168.1.48:2473 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.48:2473;branch=z9hG4bK-d8754z-6e47325f4d62920c-1—d8754z-;received=192.168.1.48;rport=2473
From: "90"sip:90@192.168.1.187:5061;tag=d267760b
To: "90"sip:90@192.168.1.187:5061;tag=as2fcf6d61
Call-ID: N2VkNjNhNzczNTY0ODE2ZTUxMTk2YzlhNjJkYTczZTE.
CSeq: 101 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Expires: 180
Contact: sip:90@192.168.1.48:2473;rinstance=13b5fa9b849e3e27;expires=180
Date: Fri, 31 Aug 2012 09:41:27 GMT
Content-Length: 0

<------------>
[Aug 31 09:41:27] VERBOSE[3135] logger.c: Scheduling destruction of SIP dialog ‘N2VkNjNhNzczNTY0ODE2ZTUxMTk2YzlhNjJkYTczZTE.’ in 32000 ms (Method: REGISTER)

4

Your Asterisk is behind NAT and you should set “externip” and “localnet” parameters in your sip.conf. At the moment, the Contact header in your replys to the provider are wrong (local IP instead of the public IP).

5

thanks für the hint… externip is set ,but localnet not, so ill check that right now…