Hi,
[Oct 31 14:31:26] NOTICE[2926] chan_sip.c: Outbound Registration: Expiry for sip.sipnl.net is 120 sec (Scheduling reregistration in 105 s)
Above is happening to me without any reason.
I can not receive calls at that time, so I am missing calls, because the line is dead when calling into the asterisk.
The asterisk is behind nat Linksys WRT54GL with DD-WRT v23 SP2 (09/15/06) std router software.
Thanks,
Robert
That message is not an error message and indicates normal operation.
But as soon as I see that message, I can not receive calls anymore.
I then have to wait 105 seconds…
How do you explain that?
All I can tell you is that that message is only output in the branch of handle_register_response that details with status 200 (i.e. success) responses.
The 200, in the message, is the value that the service provider has specified, or a default, 105, is calculated as configurable? percentage of that, subject to configurable? maximum and minimum values.
You really need to enable SIP tracing to see exactly what is happening.
“enable SIP tracing” I use “sip set debug” from: asteriskguru.com/tutorials/cli_cmd_14.html
and that this log line only came visible using that cli command
Note that I use FreePBX.
Below some more logs. Hope you can make something out of it.
<------------->
[Oct 31 16:15:14] VERBOSE[2926] logger.c: — (10 headers 0 lines) —
[Oct 31 16:15:14] VERBOSE[2926] logger.c: Scheduling destruction of SIP dialog ‘6baf49c46d9415a30b9c08642a555a7c@127.0.0.1’ in 32000 ms (Method: REGISTER)
[Oct 31 16:15:14] NOTICE[2926] chan_sip.c: Outbound Registration: Expiry for sip.sipnl.net is 120 sec (Scheduling reregistration in 105 s)
[Oct 31 16:15:44] VERBOSE[2926] logger.c:
<— SIP read from 10.0.0.153:6488 —>
<------------->
[Oct 31 16:15:46] VERBOSE[2926] logger.c: Really destroying SIP dialog ‘6baf49c46d9415a30b9c08642a555a7c@127.0.0.1’ Method: REGISTER
[Oct 31 16:15:48] VERBOSE[2926] logger.c: Reliably Transmitting (NAT) to 10.0.0.153:6488:
OPTIONS sip:1000@10.0.0.153:6488;rinstance=7aef9e2907c3359b SIP/2.0
Via: SIP/2.0/UDP 10.0.0.142:5060;branch=z9hG4bK79e2d855;rport
From: “Unknown” sip:Unknown@10.0.0.142;tag=as528ad3d1
To: sip:1000@10.0.0.153:6488;rinstance=7aef9e2907c3359b
Contact: sip:Unknown@10.0.0.142
Call-ID: 02593b4d66f47eed5f0464d72b14b690@10.0.0.142
CSeq: 102 OPTIONS
User-Agent: Meesterlijk PBX
Max-Forwards: 70
Date: Fri, 31 Oct 2008 15:15:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
[Oct 31 16:15:48] VERBOSE[2926] logger.c:
<— SIP read from 10.0.0.153:6488 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.142:5060;branch=z9hG4bK79e2d855;rport=5060
Contact: sip:10.0.0.153:6488
To: sip:1000@10.0.0.153:6488;rinstance=7aef9e2907c3359b;tag=ef433579
From: "Unknown"sip:Unknown@10.0.0.142;tag=as528ad3d1
Call-ID: 02593b4d66f47eed5f0464d72b14b690@10.0.0.142
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite release 1100l stamp 47546
Content-Length: 0
<------------->
[Oct 31 16:15:48] VERBOSE[2926] logger.c: — (12 headers 0 lines) —
[Oct 31 16:15:48] VERBOSE[2926] logger.c: Really destroying SIP dialog ‘02593b4d66f47eed5f0464d72b14b690@10.0.0.142’ Method: OPTIONS
[Oct 31 16:15:56] VERBOSE[2926] logger.c:
<— SIP read from 10.0.0.153:6488 —>
PUBLISH sip:1000@10.0.0.142 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.153:6488;branch=z9hG4bK-d8754z-7e689e7028379364-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:1000@10.0.0.153:6488
To: "Meesterlijk"sip:1000@10.0.0.142
From: "Meesterlijk"sip:1000@10.0.0.142;tag=215f4e38
Call-ID: MjJjNDNkODA2NWJjZmFhZjM3NzE1NGUzNWJhOTNkZjc.
CSeq: 1 PUBLISH
Expires: 3600
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/pidf+xml
User-Agent: X-Lite release 1100l stamp 47546
Event: presence
Content-Length: 418
<------------->
[Oct 31 16:15:56] VERBOSE[2926] logger.c: — (14 headers 1 lines) —
[Oct 31 16:15:56] VERBOSE[2926] logger.c: Sending to 10.0.0.153 : 6488 (NAT)
[Oct 31 16:15:56] VERBOSE[2926] logger.c:
<— Transmitting (NAT) to 10.0.0.153:6488 —>
SIP/2.0 501 Method Not Implemented
Via: SIP/2.0/UDP 10.0.0.153:6488;branch=z9hG4bK-d8754z-7e689e7028379364-1—d8754z-;received=10.0.0.153;rport=6488
From: "Meesterlijk"sip:1000@10.0.0.142;tag=215f4e38
To: "Meesterlijk"sip:1000@10.0.0.142;tag=as5fe9b051
Call-ID: MjJjNDNkODA2NWJjZmFhZjM3NzE1NGUzNWJhOTNkZjc.
CSeq: 1 PUBLISH
User-Agent: Meesterlijk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
[Oct 31 16:16:14] VERBOSE[2926] logger.c:
<— SIP read from 10.0.0.153:6488 —>
<------------->
[Oct 31 16:16:44] VERBOSE[2926] logger.c:
<— SIP read from 10.0.0.153:6488 —>
<------------->
[Oct 31 16:16:48] VERBOSE[2926] logger.c: Reliably Transmitting (NAT) to 10.0.0.153:6488:
OPTIONS sip:1000@10.0.0.153:6488;rinstance=7aef9e2907c3359b SIP/2.0
Via: SIP/2.0/UDP 10.0.0.142:5060;branch=z9hG4bK02d9633d;rport
From: “Unknown” sip:Unknown@10.0.0.142;tag=as2bf44845
To: sip:1000@10.0.0.153:6488;rinstance=7aef9e2907c3359b
Contact: sip:Unknown@10.0.0.142
Call-ID: 636903d61a1e88f54d55cad532b6dbc0@10.0.0.142
CSeq: 102 OPTIONS
User-Agent: Meesterlijk PBX
Max-Forwards: 70
Date: Fri, 31 Oct 2008 15:16:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
[Oct 31 16:16:48] VERBOSE[2926] logger.c:
<— SIP read from 10.0.0.153:6488 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.142:5060;branch=z9hG4bK02d9633d;rport=5060
Contact: sip:10.0.0.153:6488
To: sip:1000@10.0.0.153:6488;rinstance=7aef9e2907c3359b;tag=64136107
From: "Unknown"sip:Unknown@10.0.0.142;tag=as2bf44845
Call-ID: 636903d61a1e88f54d55cad532b6dbc0@10.0.0.142
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite release 1100l stamp 47546
Content-Length: 0
<------------->
[Oct 31 16:16:48] VERBOSE[2926] logger.c: — (12 headers 0 lines) —
[Oct 31 16:16:48] VERBOSE[2926] logger.c: Really destroying SIP dialog ‘636903d61a1e88f54d55cad532b6dbc0@10.0.0.142’ Method: OPTIONS
[Oct 31 16:16:59] NOTICE[2926] chan_sip.c: – Re-registration for 31##########@sip.sipnl.net
[Oct 31 16:16:59] DEBUG[2926] chan_sip.c: >>> Re-using Auth data for 31##########@sip.sipnl.net
[Oct 31 16:16:59] VERBOSE[2926] logger.c: REGISTER 13 headers, 0 lines
[Oct 31 16:16:59] VERBOSE[2926] logger.c: Reliably Transmitting (no NAT) to 83.98.222.4:5060:
REGISTER sip:sip.sipnl.net SIP/2.0
Via: SIP/2.0/UDP ##asterisk-server-ip##:5060;branch=z9hG4bK4dce5ee8;rport
From: sip:31##########@sip.sipnl.net;tag=as6a33a95e
To: sip:31##########@sip.sipnl.net
Call-ID: 6baf49c46d9415a30b9c08642a555a7c@127.0.0.1
CSeq: 154 REGISTER
User-Agent: Meesterlijk PBX
Max-Forwards: 70
Authorization: Digest username=“31##########”, realm=“sip.sipnl.net”, algorithm=MD5, uri=“sip:sip.sipnl.net”, nonce=“490b0da71fdb58f8bbe0f0c14ac91e27c659865b”, response="d4b4e8a6dce76cac1e8f649dd9e98a20"
Expires: 120
Contact: sip:0##########@##asterisk-server-ip##
Event: registration
Content-Length: 0
[Oct 31 16:17:00] VERBOSE[2926] logger.c:
<— SIP read from 83.98.222.4:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.142:5060;branch=z9hG4bK4dce5ee8;rport=5060
From: sip:31##########@sip.sipnl.net;tag=as6a33a95e
To: sip:31##########@sip.sipnl.net;tag=5e634920bcd56c1f88fdd8c382cc7b63-8fc0
Call-ID: 6baf49c46d9415a30b9c08642a555a7c@127.0.0.1
CSeq: 154 REGISTER
PortaBilling: currency:EUR
Contact: sip:0##########@10.0.0.142:5060;expires=295
Server: Sip EXpress router (0.9.6 (i386/freebsd))
Content-Length: 0
<------------->
[Oct 31 16:17:00] VERBOSE[2926] logger.c: — (10 headers 0 lines) —
[Oct 31 16:17:00] VERBOSE[2926] logger.c: Scheduling destruction of SIP dialog ‘6baf49c46d9415a30b9c08642a555a7c@127.0.0.1’ in 32000 ms (Method: REGISTER)
[Oct 31 16:17:00] NOTICE[2926] chan_sip.c: Outbound Registration: Expiry for sip.sipnl.net is 120 sec (Scheduling reregistration in 105 s)
[Oct 31 16:17:15] VERBOSE[2926] logger.c:
<— SIP read from 10.0.0.153:6488 —>
In isolation, that looks perfectly OK. Please note that the message you mention should appear after every successful registration, so it is not possible to tell from the trace whether anything went wrong with the registration. On the surface, nothing did.
The only time that the exchange that you show would not produce a successful registration, is if the other side is conforming, is if the number in the CSEQ was a repeat. One would need preceeding REGISTERS to establish that, preferably relative to a register that you know didn’t actually work.
The call id should be the same in all of them.
Note, you may find that you can get the key information using SIP history.
So you mean that:
Call-ID: 6baf49c46d9415a30b9c08642a555a7c@127.0.0.1 (should be the same)
CSeq: 154 REGISTER (should be different on any registration?)
It should increment for each re-REGISTER. If it doesn’t, it indicates a retransmission. However, the service provider should not be checking that number for a simple re-registration, I think you need to show an accurately timestamped trace to the service provider, plus evidence that a call failed within 295 seconds of the OK being sent.
The other possibility is that you are losing a lot of packets and the re-REGISTER is getting sent and lost so many times that the 295 seconds does expire, but, in that case, you sould be OK for the best part of 295 seconds after getting your message.
Call-ID: 6baf49c46d9415a30b9c08642a555a7c@127.0.0.1
CSeq: 285 REGISTER
Clearly different.
Since I have a 1:1 50Mbit dsl I do not think that I will loose connection that easely.
Can you see if the problem will resolve after you add “qualify=yes” to your sip.conf file?
Qualify yes makes no difference.
However I did find something else.
sip.sipnl.net is using portaone (portaone.com), and sip.sipnl.net is using 2 sip servers connected (probably load balancing). So sip.sipnl.net is using 2 ip’s for registering where my asterisk is connecting to.
Below the log fragment, look for the comments in bold.
So, the problem is in the registration, as long as the asterisk is connected to .5 I have no connection, when it switches to .4 I do get connected.
Last night it was reverse. So it has to do with how the registrations are handled at my asterisk, or at sipnl.net ??? I do not have enough voip/asterisk knowledge about this. Hope you guys can make something out of this.
<------------->
[Nov 1 09:09:55] VERBOSE[2926] logger.c: — (10 headers 0 lines) —
[Nov 1 09:09:55] VERBOSE[2926] logger.c: Scheduling destruction of SIP dialog ‘4f1abec40a09d467302d456564a6113d@127.0.0.1’ in 32000 ms (Method: REGISTER)
[Nov 1 09:09:55] NOTICE[2926] chan_sip.c: Outbound Registration: Expiry for sip.sipnl.net is 120 sec (Scheduling reregistration in 105 s)
[Nov 1 09:10:00] VERBOSE[2926] logger.c: Really destroying SIP dialog ‘4C7A39F3-A72311DD-9B32A8EA-14E1577B@83.98.222.252~1o’ Method: ACK
[Nov 1 09:10:10] VERBOSE[2926] logger.c: Reliably Transmitting (no NAT) to 83.98.222.5:5060:
OPTIONS sip:sip.sipnl.net SIP/2.0
Via: SIP/2.0/UDP ###My-asterisk-ip###:5060;branch=z9hG4bK697df5c8;rport
From: “Unknown” sip:Unknown@###My-asterisk-ip###;tag=as498c3b8b
To: sip:sip.sipnl.net
Contact: sip:Unknown@###My-asterisk-ip###
Call-ID: 25a986bc7f3c2b52502c64df0bd639f7@###My-asterisk-ip###
CSeq: 102 OPTIONS
User-Agent: Meesterlijk PBX
Max-Forwards: 70
Date: Sat, 01 Nov 2008 08:10:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
[Nov 1 09:10:10] VERBOSE[2926] logger.c:
<— SIP read from 83.98.222.5:5060 —>
SIP/2.0 501 Not Implemented
Via: SIP/2.0/UDP ###My-asterisk-ip###:5060;branch=z9hG4bK697df5c8;rport=5060
From: Unknown sip:Unknown@###My-asterisk-ip###;tag=as498c3b8b
To: sip:sip.sipnl.net
Call-ID: 25a986bc7f3c2b52502c64df0bd639f7@###My-asterisk-ip###
CSeq: 102 OPTIONS
Server: Sippy
<------------->
<<<< HERE I START WITH ‘07MYTESTCALL’, AND COMES IN ON 83.98.222.5 AND GET NO CONNECTION >>>>>
[Nov 1 09:10:10] VERBOSE[2926] logger.c: — (7 headers 0 lines) —
[Nov 1 09:10:10] VERBOSE[2926] logger.c: Really destroying SIP dialog ‘25a986bc7f3c2b52502c64df0bd639f7@###My-asterisk-ip###’ Method: OPTIONS
[Nov 1 09:10:27] VERBOSE[2926] logger.c: Really destroying SIP dialog ‘4f1abec40a09d467302d456564a6113d@127.0.0.1’ Method: REGISTER
[Nov 1 09:10:28] VERBOSE[2926] logger.c:
<— SIP read from 83.98.222.5:5060 —>
INVITE sip:0###My-phonenumber###@###My-asterisk-ip###:5060 SIP/2.0
Record-Route: sip:83.98.222.5;ftag=917035bc7f3cf0a28a8c341fddb11b9c;lr
Via: SIP/2.0/UDP 83.98.222.5;branch=z9hG4bK1f5e.5715bbf8dc94004d7bddbbc90407d3ea.0
Via: SIP/2.0/UDP 83.98.222.5:5061;branch=z9hG4bK6ca9248f049b99a46882369e46c9cfa7;rport=5061
Max-Forwards: 16
From: sip:07MYTESTCALL@83.98.222.5;tag=917035bc7f3cf0a28a8c341fddb11b9c
To: sip:31###My-phonenumber###@83.98.222.5
Call-ID: 60EFB7AA-A72311DD-9B48A8EA-14E1577B@83.98.222.252~1o
CSeq: 200 INVITE
Contact: Anonymous sip:83.98.222.5:5061
Expires: 300
User-Agent: Sippy
cisco-GUID: 1626002898-2804093405-3095658503-240686630
h323-conf-id: 1626002898-2804093405-3095658503-240686630
Content-disposition: session
Content-Length: 312
Content-Type: application/sdp
v=0
o=Sippy 147938252 0 IN IP4 83.98.222.5
s=SIP Call
t=0 0
m=audio 23272 RTP/AVP 8 0 18 100 101
c=IN IP4 83.98.222.252
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
<------------->
[Nov 1 09:10:28] VERBOSE[2926] logger.c: — (17 headers 14 lines) —
[Nov 1 09:10:28] WARNING[2926] rtp.c: Unable to set TOS to 184
[Nov 1 09:10:28] VERBOSE[2926] logger.c: Sending to 83.98.222.5 : 5060 (no NAT)
[Nov 1 09:10:28] VERBOSE[2926] logger.c: Using INVITE request as basis request - 60EFB7AA-A72311DD-9B48A8EA-14E1577B@83.98.222.252~1o
[Nov 1 09:10:28] VERBOSE[2926] logger.c: Found peer ‘breezz’
[Nov 1 09:10:28] VERBOSE[2926] logger.c:
<— Reliably Transmitting (no NAT) to 83.98.222.5:5060 —>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 83.98.222.5;branch=z9hG4bK1f5e.5715bbf8dc94004d7bddbbc90407d3ea.0;received=83.98.222.5
Via: SIP/2.0/UDP 83.98.222.5:5061;branch=z9hG4bK6ca9248f049b99a46882369e46c9cfa7;rport=5061
From: sip:07MYTESTCALL@83.98.222.5;tag=917035bc7f3cf0a28a8c341fddb11b9c
To: sip:31###My-phonenumber###@83.98.222.5;tag=as62b0ea83
Call-ID: 60EFB7AA-A72311DD-9B48A8EA-14E1577B@83.98.222.252~1o
CSeq: 200 INVITE
User-Agent: Meesterlijk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="2c2d5312"
Content-Length: 0
<------------>
[Nov 1 09:10:28] VERBOSE[2926] logger.c: Scheduling destruction of SIP dialog ‘60EFB7AA-A72311DD-9B48A8EA-14E1577B@83.98.222.252~1o’ in 6400 ms (Method: INVITE)
[Nov 1 09:10:28] VERBOSE[2926] logger.c:
<— SIP read from 83.98.222.5:5060 —>
ACK sip:0###My-phonenumber###@###My-asterisk-ip###:5060 SIP/2.0
Via: SIP/2.0/UDP 83.98.222.5;branch=z9hG4bK1f5e.5715bbf8dc94004d7bddbbc90407d3ea.0
From: sip:07MYTESTCALL@83.98.222.5;tag=917035bc7f3cf0a28a8c341fddb11b9c
Call-ID: 60EFB7AA-A72311DD-9B48A8EA-14E1577B@83.98.222.252~1o
To: sip:31###My-phonenumber###@83.98.222.5;tag=as62b0ea83
CSeq: 200 ACK
User-Agent: Sip EXpress router (0.9.6 (i386/freebsd))
Content-Length: 0
<------------->
[Nov 1 09:10:28] VERBOSE[2926] logger.c: — (8 headers 0 lines) —
[Nov 1 09:10:34] VERBOSE[2926] logger.c: Really destroying SIP dialog ‘60EFB7AA-A72311DD-9B48A8EA-14E1577B@83.98.222.252~1o’ Method: ACK
[Nov 1 09:11:03] VERBOSE[2926] logger.c:
<— SIP read from 83.98.222.5:5060 —>
INVITE sip:0###My-phonenumber###@###My-asterisk-ip###:5060 SIP/2.0
Record-Route: sip:83.98.222.5;ftag=ad542b1e3597cbfa40b961e0fea1d65d;lr
Via: SIP/2.0/UDP 83.98.222.5;branch=z9hG4bKed3a.8ecbd46ee2781adbe3cc687d0e34d0e9.0
Via: SIP/2.0/UDP 83.98.222.5:5061;branch=z9hG4bK60942342970b243bc20a3150baa119e2;rport=5061
Max-Forwards: 16
From: sip:07MYTESTCALL@83.98.222.5;tag=ad542b1e3597cbfa40b961e0fea1d65d
To: sip:31###My-phonenumber###@83.98.222.5
Call-ID: 75B0E377-A72311DD-9B61A8EA-14E1577B@83.98.222.252~1o
CSeq: 200 INVITE
Contact: Anonymous sip:83.98.222.5:5061
Expires: 300
User-Agent: Sippy
cisco-GUID: 1974246855-2804093405-3095920647-240686630
h323-conf-id: 1974246855-2804093405-3095920647-240686630
Content-disposition: session
Content-Length: 312
Content-Type: application/sdp
v=0
o=Sippy 152698380 0 IN IP4 83.98.222.5
s=SIP Call
t=0 0
m=audio 22960 RTP/AVP 8 0 18 100 101
c=IN IP4 83.98.222.252
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
<------------->
[Nov 1 09:11:03] VERBOSE[2926] logger.c: — (17 headers 14 lines) —
[Nov 1 09:11:03] WARNING[2926] rtp.c: Unable to set TOS to 184
[Nov 1 09:11:03] VERBOSE[2926] logger.c: Sending to 83.98.222.5 : 5060 (no NAT)
[Nov 1 09:11:03] VERBOSE[2926] logger.c: Using INVITE request as basis request - 75B0E377-A72311DD-9B61A8EA-14E1577B@83.98.222.252~1o
[Nov 1 09:11:03] VERBOSE[2926] logger.c: Found peer ‘breezz’
[Nov 1 09:11:03] VERBOSE[2926] logger.c:
<— Reliably Transmitting (no NAT) to 83.98.222.5:5060 —>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 83.98.222.5;branch=z9hG4bKed3a.8ecbd46ee2781adbe3cc687d0e34d0e9.0;received=83.98.222.5
Via: SIP/2.0/UDP 83.98.222.5:5061;branch=z9hG4bK60942342970b243bc20a3150baa119e2;rport=5061
From: sip:07MYTESTCALL@83.98.222.5;tag=ad542b1e3597cbfa40b961e0fea1d65d
To: sip:31###My-phonenumber###@83.98.222.5;tag=as3f1273f2
Call-ID: 75B0E377-A72311DD-9B61A8EA-14E1577B@83.98.222.252~1o
CSeq: 200 INVITE
User-Agent: Meesterlijk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="418c792b"
Content-Length: 0
<------------>
[Nov 1 09:11:03] VERBOSE[2926] logger.c: Scheduling destruction of SIP dialog ‘75B0E377-A72311DD-9B61A8EA-14E1577B@83.98.222.252~1o’ in 6400 ms (Method: INVITE)
[Nov 1 09:11:03] VERBOSE[2926] logger.c:
<— SIP read from 83.98.222.5:5060 —>
ACK sip:0###My-phonenumber###@###My-asterisk-ip###:5060 SIP/2.0
Via: SIP/2.0/UDP 83.98.222.5;branch=z9hG4bKed3a.8ecbd46ee2781adbe3cc687d0e34d0e9.0
From: sip:07MYTESTCALL@83.98.222.5;tag=ad542b1e3597cbfa40b961e0fea1d65d
Call-ID: 75B0E377-A72311DD-9B61A8EA-14E1577B@83.98.222.252~1o
To: sip:31###My-phonenumber###@83.98.222.5;tag=as3f1273f2
CSeq: 200 ACK
User-Agent: Sip EXpress router (0.9.6 (i386/freebsd))
Content-Length: 0
<------------->
[Nov 1 09:11:03] VERBOSE[2926] logger.c: — (8 headers 0 lines) —
[Nov 1 09:11:09] VERBOSE[2926] logger.c: Really destroying SIP dialog ‘75B0E377-A72311DD-9B61A8EA-14E1577B@83.98.222.252~1o’ Method: ACK
[Nov 1 09:11:10] VERBOSE[2926] logger.c: Reliably Transmitting (no NAT) to 83.98.222.5:5060:
OPTIONS sip:sip.sipnl.net SIP/2.0
Via: SIP/2.0/UDP ###My-asterisk-ip###:5060;branch=z9hG4bK2687e18e;rport
From: “Unknown” sip:Unknown@###My-asterisk-ip###;tag=as4a94cedf
To: sip:sip.sipnl.net
Contact: sip:Unknown@###My-asterisk-ip###
Call-ID: 3dc6b3131eae99d508c135b924cd6091@###My-asterisk-ip###
CSeq: 102 OPTIONS
User-Agent: Meesterlijk PBX
Max-Forwards: 70
Date: Sat, 01 Nov 2008 08:11:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
[Nov 1 09:11:10] VERBOSE[2926] logger.c:
<— SIP read from 83.98.222.5:5060 —>
SIP/2.0 501 Not Implemented
Via: SIP/2.0/UDP ###My-asterisk-ip###:5060;branch=z9hG4bK2687e18e;rport=5060
From: Unknown sip:Unknown@###My-asterisk-ip###;tag=as4a94cedf
To: sip:sip.sipnl.net
Call-ID: 3dc6b3131eae99d508c135b924cd6091@###My-asterisk-ip###
CSeq: 102 OPTIONS
Server: Sippy
<------------->
[Nov 1 09:11:10] VERBOSE[2926] logger.c: — (7 headers 0 lines) —
[Nov 1 09:11:10] VERBOSE[2926] logger.c: Really destroying SIP dialog ‘3dc6b3131eae99d508c135b924cd6091@###My-asterisk-ip###’ Method: OPTIONS
[Nov 1 09:11:15] VERBOSE[10451] logger.c: == Parsing ‘/etc/asterisk/manager.conf’:
[Nov 1 09:11:15] VERBOSE[10451] logger.c: Found
[Nov 1 09:11:15] VERBOSE[10451] logger.c: == Parsing ‘/etc/asterisk/manager_additional.conf’:
[Nov 1 09:11:15] VERBOSE[10451] logger.c: Found
[Nov 1 09:11:15] VERBOSE[10451] logger.c: == Parsing ‘/etc/asterisk/manager_custom.conf’:
[Nov 1 09:11:15] VERBOSE[10451] logger.c: Found
[Nov 1 09:11:15] VERBOSE[10451] logger.c: == Manager ‘admin’ logged on from 127.0.0.1
[Nov 1 09:11:15] VERBOSE[10451] logger.c: == Manager ‘admin’ logged off from 127.0.0.1
[Nov 1 09:11:20] VERBOSE[2926] logger.c:
<— SIP read from 83.98.222.5:5060 —>
INVITE sip:0###My-phonenumber###@###My-asterisk-ip###:5060 SIP/2.0
Record-Route: sip:83.98.222.5;ftag=0f829436d357bdc85d5d76418d86147f;lr
Via: SIP/2.0/UDP 83.98.222.5;branch=z9hG4bKfcf7.945870fdff5c9fd1b8f55354bd16bc7c.0
Via: SIP/2.0/UDP 83.98.222.5:5061;branch=z9hG4bK5baa39750c00755b3fa1e8e4cfcc64d2;rport=5061
Max-Forwards: 16
From: sip:07MYTESTCALL@83.98.222.5;tag=0f829436d357bdc85d5d76418d86147f
To: sip:31###My-phonenumber###@83.98.222.5
Call-ID: 801919CC-A72311DD-9B68A8EA-14E1577B@83.98.222.252~1o
CSeq: 200 INVITE
Contact: Anonymous sip:83.98.222.5:5061
Expires: 300
User-Agent: Sippy
cisco-GUID: 2148848668-2804093405-3095986183-240686630
h323-conf-id: 2148848668-2804093405-3095986183-240686630
Content-disposition: session
Content-Length: 312
Content-Type: application/sdp
v=0
o=Sippy 176356460 0 IN IP4 83.98.222.5
s=SIP Call
t=0 0
m=audio 23686 RTP/AVP 8 0 18 100 101
c=IN IP4 83.98.222.252
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
<------------->
[Nov 1 09:11:20] VERBOSE[2926] logger.c: — (17 headers 14 lines) —
[Nov 1 09:11:20] WARNING[2926] rtp.c: Unable to set TOS to 184
[Nov 1 09:11:20] VERBOSE[2926] logger.c: Sending to 83.98.222.5 : 5060 (no NAT)
[Nov 1 09:11:20] VERBOSE[2926] logger.c: Using INVITE request as basis request - 801919CC-A72311DD-9B68A8EA-14E1577B@83.98.222.252~1o
[Nov 1 09:11:20] VERBOSE[2926] logger.c: Found peer ‘breezz’
[Nov 1 09:11:20] VERBOSE[2926] logger.c:
<— Reliably Transmitting (no NAT) to 83.98.222.5:5060 —>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 83.98.222.5;branch=z9hG4bKfcf7.945870fdff5c9fd1b8f55354bd16bc7c.0;received=83.98.222.5
Via: SIP/2.0/UDP 83.98.222.5:5061;branch=z9hG4bK5baa39750c00755b3fa1e8e4cfcc64d2;rport=5061
From: sip:07MYTESTCALL@83.98.222.5;tag=0f829436d357bdc85d5d76418d86147f
To: sip:31###My-phonenumber###@83.98.222.5;tag=as26318f70
Call-ID: 801919CC-A72311DD-9B68A8EA-14E1577B@83.98.222.252~1o
CSeq: 200 INVITE
User-Agent: Meesterlijk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="5887d3c3"
Content-Length: 0
<------------>
[Nov 1 09:11:20] VERBOSE[2926] logger.c: Scheduling destruction of SIP dialog ‘801919CC-A72311DD-9B68A8EA-14E1577B@83.98.222.252~1o’ in 6400 ms (Method: INVITE)
[Nov 1 09:11:20] VERBOSE[2926] logger.c:
<— SIP read from 83.98.222.5:5060 —>
ACK sip:0###My-phonenumber###@###My-asterisk-ip###:5060 SIP/2.0
Via: SIP/2.0/UDP 83.98.222.5;branch=z9hG4bKfcf7.945870fdff5c9fd1b8f55354bd16bc7c.0
From: sip:07MYTESTCALL@83.98.222.5;tag=0f829436d357bdc85d5d76418d86147f
Call-ID: 801919CC-A72311DD-9B68A8EA-14E1577B@83.98.222.252~1o
To: sip:31###My-phonenumber###@83.98.222.5;tag=as26318f70
CSeq: 200 ACK
User-Agent: Sip EXpress router (0.9.6 (i386/freebsd))
Content-Length: 0
<------------->
<<<< FROM HERE I DO GET CONNECTED AND YOU SEE THE IP CHANGES >>>>
[Nov 1 09:11:20] VERBOSE[2926] logger.c: — (8 headers 0 lines) —
[Nov 1 09:11:27] VERBOSE[2926] logger.c: Really destroying SIP dialog ‘801919CC-A72311DD-9B68A8EA-14E1577B@83.98.222.252~1o’ Method: ACK
[Nov 1 09:11:40] NOTICE[2926] chan_sip.c: – Re-registration for 31###My-phonenumber###@sip.sipnl.net
[Nov 1 09:11:40] DEBUG[2926] chan_sip.c: >>> Re-using Auth data for 31###My-phonenumber###@sip.sipnl.net
[Nov 1 09:11:40] VERBOSE[2926] logger.c: REGISTER 13 headers, 0 lines
[Nov 1 09:11:40] VERBOSE[2926] logger.c: Reliably Transmitting (no NAT) to 83.98.222.4:5060:
REGISTER sip:sip.sipnl.net SIP/2.0
Via: SIP/2.0/UDP ###My-asterisk-ip###:5060;branch=z9hG4bK62c772c7;rport
From: sip:31###My-phonenumber###@sip.sipnl.net;tag=as74d8df38
To: sip:31###My-phonenumber###@sip.sipnl.net
Call-ID: 4f1abec40a09d467302d456564a6113d@127.0.0.1
CSeq: 105 REGISTER
User-Agent: Meesterlijk PBX
Max-Forwards: 70
Authorization: Digest username=“31###My-phonenumber###”, realm=“sip.sipnl.net”, algorithm=MD5, uri=“sip:sip.sipnl.net”, nonce=“490c0f967b9577112d673d2538a7199b58965b4d”, response="4f9d3fd3c078f970a420f8ebc2b5cdef"
Expires: 120
Contact: sip:0###My-phonenumber###@###My-asterisk-ip###
Event: registration
Content-Length: 0
[Nov 1 09:11:40] VERBOSE[2926] logger.c:
<— SIP read from 83.98.222.4:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.142:5060;branch=z9hG4bK62c772c7;rport=5060
From: sip:31###My-phonenumber###@sip.sipnl.net;tag=as74d8df38
To: sip:31###My-phonenumber###@sip.sipnl.net;tag=5e634920bcd56c1f88fdd8c382cc7b63-8c4d
Call-ID: 4f1abec40a09d467302d456564a6113d@127.0.0.1
CSeq: 105 REGISTER
PortaBilling: currency:EUR
Contact: sip:0###My-phonenumber###@10.0.0.142:5060;expires=295
Server: Sip EXpress router (0.9.6 (i386/freebsd))
Content-Length: 0
<------------->
[Nov 1 09:11:40] VERBOSE[2926] logger.c: — (10 headers 0 lines) —
[Nov 1 09:11:40] VERBOSE[2926] logger.c: Scheduling destruction of SIP dialog ‘4f1abec40a09d467302d456564a6113d@127.0.0.1’ in 32000 ms (Method: REGISTER)
[Nov 1 09:11:40] NOTICE[2926] chan_sip.c: Outbound Registration: Expiry for sip.sipnl.net is 120 sec (Scheduling reregistration in 105 s)
[Nov 1 09:12:01] VERBOSE[2926] logger.c:
<— SIP read from 83.98.222.4:5060 —>
INVITE sip:0###My-phonenumber###@###My-asterisk-ip###:5060 SIP/2.0
Record-Route: sip:83.98.222.4;ftag=db34bdb5119a10473943332ffaa7fc61;lr
Via: SIP/2.0/UDP 83.98.222.4;branch=z9hG4bKc0a7.38495d2447f30e043a0d9b6adaa7f8c8.0
Via: SIP/2.0/UDP 83.98.222.4:5061;branch=z9hG4bKeaf1456f7c33395eabab3bb12c38bd03;rport=5061
Max-Forwards: 16
From: sip:07MYTESTCALL@83.98.222.4;tag=db34bdb5119a10473943332ffaa7fc61
To: sip:31###My-phonenumber###@83.98.222.4
Call-ID: 983B2778-A72311DD-9B78A8EA-14E1577B@83.98.222.252~1o~1o
CSeq: 200 INVITE
Contact: Anonymous sip:83.98.222.4:5061
Expires: 300
User-Agent: Sippy
cisco-GUID: 2553653624-2804093405-3096117255-240686630
h323-conf-id: 2553653624-2804093405-3096117255-240686630
Content-disposition: session
Content-Length: 312
Content-Type: application/sdp
v=0
o=Sippy 158075564 0 IN IP4 83.98.222.4
s=SIP Call
t=0 0
m=audio 24212 RTP/AVP 8 0 18 100 101
c=IN IP4 83.98.222.252
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
<------------->
[Nov 1 09:12:01] VERBOSE[2926] logger.c: — (17 headers 14 lines) —
[Nov 1 09:12:01] WARNING[2926] rtp.c: Unable to set TOS to 184
[Nov 1 09:12:01] VERBOSE[2926] logger.c: Sending to 83.98.222.4 : 5060 (no NAT)
[Nov 1 09:12:01] VERBOSE[2926] logger.c: Using INVITE request as basis request - 983B2778-A72311DD-9B78A8EA-14E1577B@83.98.222.252~1o~1o
[Nov 1 09:12:01] VERBOSE[2926] logger.c: Found no matching peer or user for ‘83.98.222.4:5060’
[Nov 1 09:12:01] VERBOSE[2926] logger.c: Found RTP audio format 8
[Nov 1 09:12:01] VERBOSE[2926] logger.c: Found RTP audio format 0
[Nov 1 09:12:01] VERBOSE[2926] logger.c: Found RTP audio format 18
[Nov 1 09:12:01] VERBOSE[2926] logger.c: Found RTP audio format 100
[Nov 1 09:12:01] VERBOSE[2926] logger.c: Found RTP audio format 101
[Nov 1 09:12:01] VERBOSE[2926] logger.c: Peer audio RTP is at port 83.98.222.252:24212
[Nov 1 09:12:01] VERBOSE[2926] logger.c: Found audio description format PCMA for ID 8
[Nov 1 09:12:01] VERBOSE[2926] logger.c: Found audio description format PCMU for ID 0
[Nov 1 09:12:01] VERBOSE[2926] logger.c: Found audio description format G729 for ID 18
[Nov 1 09:12:01] VERBOSE[2926] logger.c: Found unknown media description format X-NSE for ID 100
[Nov 1 09:12:01] VERBOSE[2926] logger.c: Found audio description format telephone-event for ID 101
[Nov 1 09:12:01] VERBOSE[2926] logger.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
[Nov 1 09:12:01] VERBOSE[2926] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Nov 1 09:12:01] VERBOSE[2926] logger.c: Peer audio RTP is at port 83.98.222.252:24212
[Nov 1 09:12:01] VERBOSE[2926] logger.c: Looking for 0###My-phonenumber### in from-sip-external (domain ###My-asterisk-ip###)
[Nov 1 09:12:01] VERBOSE[2926] logger.c: list_route: hop: sip:83.98.222.4;ftag=db34bdb5119a10473943332ffaa7fc61;lr
[Nov 1 09:12:01] VERBOSE[2926] logger.c:
<— Transmitting (no NAT) to 83.98.222.4:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 83.98.222.4;branch=z9hG4bKc0a7.38495d2447f30e043a0d9b6adaa7f8c8.0;received=83.98.222.4
Via: SIP/2.0/UDP 83.98.222.4:5061;branch=z9hG4bKeaf1456f7c33395eabab3bb12c38bd03;rport=5061
Record-Route: sip:83.98.222.4;ftag=db34bdb5119a10473943332ffaa7fc61;lr
From: sip:07MYTESTCALL@83.98.222.4;tag=db34bdb5119a10473943332ffaa7fc61
To: sip:31###My-phonenumber###@83.98.222.4
Call-ID: 983B2778-A72311DD-9B78A8EA-14E1577B@83.98.222.252~1o~1o
CSeq: 200 INVITE
User-Agent: Meesterlijk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:0###My-phonenumber###@###My-asterisk-ip###
Content-Length: 0
<------------>
[Nov 1 09:12:01] VERBOSE[10454] logger.c: – Executing [0###My-phonenumber###@from-sip-external:1] NoOp(“SIP/83.98.222.4-09a2ebf0”, “Received incoming SIP connection from unknown peer to 0###My-phonenumber###”) in new stack
[Nov 1 09:12:01] VERBOSE[10454] logger.c: – Executing [0###My-phonenumber###@from-sip-external:2] Set(“SIP/83.98.222.4-09a2ebf0”, “DID=0###My-phonenumber###”) in new stack
[Nov 1 09:12:01] VERBOSE[10454] logger.c: – Executing [0###My-phonenumber###@from-sip-external:3] Goto(“SIP/83.98.222.4-09a2ebf0”, “s|1”) in new stack
[Nov 1 09:12:01] VERBOSE[10454] logger.c: – Goto (from-sip-external,s,1)
[Nov 1 09:12:01] VERBOSE[10454] logger.c: – Executing [s@from-sip-external:1] GotoIf(“SIP/83.98.222.4-09a2ebf0”, “1?from-trunk|0###My-phonenumber###|1”) in new stack
[Nov 1 09:12:01] VERBOSE[10454] logger.c: – Goto (from-trunk,0###My-phonenumber###,1)
[Nov 1 09:12:01] VERBOSE[10454] logger.c: – Executing [0###My-phonenumber###@from-trunk:1] NoOp(“SIP/83.98.222.4-09a2ebf0”, “Catch-All DID Match - Found 0###My-phonenumber### - You probably want a DID for this.”) in new stack
[Nov 1 09:12:01] VERBOSE[10454] logger.c: – Executing [0###My-phonenumber###@from-trunk:2] Goto(“SIP/83.98.222.4-09a2ebf0”, “ext-did|s|1”) in new stack
[Nov 1 09:12:01] VERBOSE[10454] logger.c: – Goto (ext-did,s,1)
[Nov 1 09:12:01] VERBOSE[10454] logger.c: – Executing [s@ext-did:1] Set(“SIP/83.98.222.4-09a2ebf0”, “__FROM_DID=s”) in new stack
[Nov 1 09:12:01] VERBOSE[10454] logger.c: – Executing [s@ext-did:2] Gosub(“SIP/83.98.222.4-09a2ebf0”, “app-blacklist-check|s|1”) in new stack
[Nov 1 09:12:01] VERBOSE[10454] logger.c: – Executing [s@app-blacklist-check:1] LookupBlacklist(“SIP/83.98.222.4-09a2ebf0”, “”) in new stack
[Nov 1 09:12:01] VERBOSE[10454] logger.c: – Executing [s@app-blacklist-check:2] GotoIf(“SIP/83.98.222.4-09a2ebf0”, “0?blacklisted”) in new stack
[Nov 1 09:12:01] VERBOSE[10454] logger.c: – Executing [s@app-blacklist-check:3] Return(“SIP/83.98.222.4-09a2ebf0”, “”) in new stack
[Nov 1 09:12:01] VERBOSE[10454] logger.c: – Executing [s@ext-did:3] ExecIf(“SIP/83.98.222.4-09a2ebf0”, “1 |Set|CALLERID(name)=07MYTESTCALL”) in new stack
[Nov 1 09:12:01] VERBOSE[10454] logger.c: – Executing [s@ext-did:4] SetMusicOnHold(“SIP/83.98.222.4-09a2ebf0”, “none”) in new stack
[Nov 1 09:12:01] VERBOSE[10454] logger.c: – Executing [s@ext-did:5] Set(“SIP/83.98.222.4-09a2ebf0”, “__MOHCLASS=none”) in new stack
[Nov 1 09:12:01] VERBOSE[10454] logger.c: – Executing [s@ext-did:6] Set(“SIP/83.98.222.4-09a2ebf0”, “__CALLINGPRES_SV=allowed_not_screened”) in new stack
[Nov 1 09:12:01] VERBOSE[10454] logger.c: – Executing [s@ext-did:7] SetCallerPres(“SIP/83.98.222.4-09a2ebf0”, “allowed_not_screened”) in new stack
[Nov 1 09:12:01] VERBOSE[10454] logger.c: – Executing [s@ext-did:8] Goto(“SIP/83.98.222.4-09a2ebf0”, “timeconditions|9|1”) in new stack
[Nov 1 09:12:01] VERBOSE[10454] logger.c: – Goto (timeconditions,9,1)
[Nov 1 09:12:01] VERBOSE[10454] logger.c: – Executing [9@timeconditions:1] GotoIfTime(“SIP/83.98.222.4-09a2ebf0”, “09:29-11:59|mon-fri||?app-announcement-8|s|1”) in new stack
[Nov 1 09:12:01] VERBOSE[10454] logger.c: – Executing [9@timeconditions:2] Goto(“SIP/83.98.222.4-09a2ebf0”, “timeconditions|10|1”) in new stack
[Nov 1 09:12:01] VERBOSE[10454] logger.c: – Goto (timeconditions,10,1)
[Nov 1 09:12:01] VERBOSE[10454] logger.c: – Executing [10@timeconditions:1] GotoIfTime(“SIP/83.98.222.4-09a2ebf0”, “14:15-16:59|mon-fri||?app-announcement-9|s|1”) in new stack
[Nov 1 09:12:01] VERBOSE[10454] logger.c: – Executing [10@timeconditions:2] GotoIfTime(“SIP/83.98.222.4-09a2ebf0”, “12:00-13:14|mon-fri||?app-announcement-9|s|1”) in new stack
[Nov 1 09:12:01] VERBOSE[10454] logger.c: – Executing [10@timeconditions:3] Goto(“SIP/83.98.222.4-09a2ebf0”, “timeconditions|11|1”) in new stack
[Nov 1 09:12:01] VERBOSE[10454] logger.c: – Goto (timeconditions,11,1)
[Nov 1 09:12:01] VERBOSE[10454] logger.c: – Executing [11@timeconditions:1] GotoIfTime(“SIP/83.98.222.4-09a2ebf0”, “13:15-14:14|mon-fri||?app-announcement-10|s|1”) in new stack
[Nov 1 09:12:01] VERBOSE[10454] logger.c: – Executing [11@timeconditions:2] Goto(“SIP/83.98.222.4-09a2ebf0”, “timeconditions|12|1”) in new stack
[Nov 1 09:12:01] VERBOSE[10454] logger.c: – Goto (timeconditions,12,1)
[Nov 1 09:12:01] VERBOSE[10454] logger.c: – Executing [12@timeconditions:1] GotoIfTime(“SIP/83.98.222.4-09a2ebf0”, “17:00-09:29|mon-fri||?app-announcement-11|s|1”) in new stack
[Nov 1 09:12:01] VERBOSE[10454] logger.c: – Executing [12@timeconditions:2] GotoIfTime(“SIP/83.98.222.4-09a2ebf0”, “17:00-09:29|fri-mon||?app-announcement-11|s|1”) in new stack
[Nov 1 09:12:01] VERBOSE[10454] logger.c: – Goto (app-announcement-11,s,1)
[Nov 1 09:12:01] VERBOSE[10454] logger.c: – Executing [s@app-announcement-11:1] GotoIf(“SIP/83.98.222.4-09a2ebf0”, “0?begin”) in new stack
[Nov 1 09:12:01] VERBOSE[10454] logger.c: – Executing [s@app-announcement-11:2] Answer(“SIP/83.98.222.4-09a2ebf0”, “”) in new stack
[Nov 1 09:12:01] VERBOSE[10454] logger.c: Audio is at ###My-asterisk-ip### port 14994
[Nov 1 09:12:01] VERBOSE[10454] logger.c: Adding codec 0x4 (ulaw) to SDP
[Nov 1 09:12:01] VERBOSE[10454] logger.c: Adding codec 0x8 (alaw) to SDP
[Nov 1 09:12:01] VERBOSE[10454] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
[Nov 1 09:12:01] VERBOSE[10454] logger.c:
<— Reliably Transmitting (no NAT) to 83.98.222.4:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 83.98.222.4;branch=z9hG4bKc0a7.38495d2447f30e043a0d9b6adaa7f8c8.0;received=83.98.222.4
Via: SIP/2.0/UDP 83.98.222.4:5061;branch=z9hG4bKeaf1456f7c33395eabab3bb12c38bd03;rport=5061
Record-Route: sip:83.98.222.4;ftag=db34bdb5119a10473943332ffaa7fc61;lr
From: sip:07MYTESTCALL@83.98.222.4;tag=db34bdb5119a10473943332ffaa7fc61
To: sip:31###My-phonenumber###@83.98.222.4;tag=as28062672
Call-ID: 983B2778-A72311DD-9B78A8EA-14E1577B@83.98.222.252~1o~1o
CSeq: 200 INVITE
User-Agent: Meesterlijk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:0###My-phonenumber###@###My-asterisk-ip###
Content-Type: application/sdp
Content-Length: 266
v=0
o=root 2910 2910 IN IP4 ###My-asterisk-ip###
s=session
c=IN IP4 ###My-asterisk-ip###
t=0 0
m=audio 14994 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
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