[general]
port=5060
bindaddr=0.0.0.0
externip=62.99.193.33
;externhost=maschi.homedns.org
localnet=192.168.0.0/255.255.255.0
context=default
srvlookup=yes
disallow=all
allow=gsm
allow=ulaw
allow=alaw
insecure=very
register => x:[email protected]/9627349
[sipgate]
type=friend
username=9627349
host=sipgate.at
secret=x
fromuser=9627349
fromdomain=sipgate.at
context=default
nat=yes
qualify=yes
insecure=very
[10]
type=friend
username=10
secret=t1
host=dynamic
qualify=1200
[11]
type=friend
username=11
secret=t2
host=dynamic
qualify=1200
[default]
include => 10
include => 11
include => fromsip
include => tosip
[10]
exten => 10,1,Dial(SIP/10,45)
exten => 10,2,Hangup
[11]
exten => 11,1,Dial(SIP/11,10)
exten => 11,2,VoiceMail(10@default)
exten => 11,4,Hangup
[tosip]
exten => _0.,1,SetCallerId("9627349")
exten => _0.,2,SetCIDName("9627349")
exten => _0.,3,Dial(SIP/${EXTEN:1}@sipgate,30,tr)
exten => _0.,4,Wait
exten => _0.,5,Hangup
[fromsip]
exten => 9627349,1,Dial(SIP/10,20,tr)
extem => 9627349,2,Hangup
CLI> -- Executing SetCallerID("SIP/10-8598", ""9627349"") in new stack
-- Executing SetCIDName("SIP/10-8598", ""9627349"") in new stack
-- Executing Dial("SIP/10-8598", "SIP/10000@sipgate|30|tr") in new stack
-- Called 10000@sipgate
-- Got SIP response 400 "Bad Request" back from 212.236.252.3
-- SIP/sipgate-3e0e is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Wait("SIP/10-8598", "") in new stack
-- Executing Hangup("SIP/10-8598", "") in new stack
== Spawn extension (default, 010000, 5) exited non-zero on 'SIP/10-8598'
*CLI> sip debug peer sipgate
SIP Debugging Enabled for IP: 212.236.252.3:5060
-- Executing SetCallerID("SIP/10-1214", ""9627349"") in new stack
-- Executing SetCIDName("SIP/10-1214", ""9627349"") in new stack
-- Executing Dial("SIP/10-1214", "SIP/10000@sipgate|30|tr") in new stack
We're at 62.99.193.33 port 16258
Answering/Requesting with root capability 0x4 (ulaw)
Answering with preferred capability 0x2 (gsm)
Answering with preferred capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
12 headers, 12 lines
Reliably Transmitting:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 62.99.193.33:5060;branch=z9hG4bK69c9d4bf;rport
From: ""9627349"" <sip:[email protected]>;tag=as5fe536fb
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email][email protected][/email]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sun, 06 Mar 2005 13:33:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 261
v=0
o=root 4255 4255 IN IP4 62.99.193.33
s=session
c=IN IP4 62.99.193.33
t=0 0
m=audio 16258 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
(NAT) to 212.236.252.3:5060
-- Called 10000@sipgate
Sip read:
SIP/2.0 400 Bad Request
Via: SIP/2.0/UDP 62.99.193.33:5060;branch=z9hG4bK69c9d4bf;rport=5060
From: ""9627349"" <sip:[email protected]>;tag=as5fe536fb
To: <sip:[email protected]>;tag=b1157e02d4a58892308b0557a5fe5724.ae1b
Call-ID: [email][email protected][/email]
CSeq: 102 INVITE
Server: sipgate ser
Content-Length: 0
Warning: 392 212.236.252.3:5060 "Noisy feedback tells: pid=10527 req_src_ip=62.99.193.33 req_src_port=5060 in_uri=sip:[email protected] out_uri=sip:[email protected] via_cnt==1"
9 headers, 0 lines
-- Got SIP response 400 "Bad Request" back from 212.236.252.3
Transmitting:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 62.99.193.33:5060;branch=z9hG4bK69c9d4bf;rport
From: ""9627349"" <sip:[email protected]>;tag=as5fe536fb
To: <sip:[email protected]>;tag=b1157e02d4a58892308b0557a5fe5724.ae1b
Contact: <sip:[email protected]>
Call-ID: [email][email protected][/email]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0
(NAT) to 212.236.252.3:5060
-- SIP/sipgate-476d is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Wait("SIP/10-1214", "") in new stack
-- Executing Hangup("SIP/10-1214", "") in new stack
== Spawn extension (default, 010000, 5) exited non-zero on 'SIP/10-1214'
Destroying call '[email protected]'
11 headers, 0 lines
Reliably Transmitting:
OPTIONS sip:sipgate.at SIP/2.0
Via: SIP/2.0/UDP 62.99.193.33:5060;branch=z9hG4bK24aad1f0
From: "asterisk" <sip:[email protected]>;tag=as6126586f
To: <sip:sipgate.at>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Sun, 06 Mar 2005 13:33:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Length: 0
(no NAT) to 212.236.252.3:5060
Sip read:
SIP/2.0 482 Loop Detected
Via: SIP/2.0/UDP 62.99.193.33:5060;branch=z9hG4bK24aad1f0
From: "asterisk" <sip:[email protected]>;tag=as6126586f
To: <sip:sipgate.at>;tag=b1157e02d4a58892308b0557a5fe5724.107c
Call-ID: [email protected]
CSeq: 102 OPTIONS
Server: sipgate ser
Content-Length: 0
Warning: 392 212.236.252.3:5060 "Noisy feedback tells: pid=10530 req_src_ip=62.99.193.33 req_src_port=5060 in_uri=sip:sipgate.at out_uri=sip:sipgate.at via_cnt==1"
9 headers, 0 lines
Destroying call '[email protected]'
*CLI>
Sip read:
0 headers, 0 lines
*CLI> Mar 6 14:34:16 NOTICE[4255]: chan_sip.c:4260 sip_reregister: -- Re-registration for [email][email protected][/email]
12 headers, 0 lines
Reliably Transmitting:
REGISTER sip:sipgate.at SIP/2.0
Via: SIP/2.0/UDP 62.99.193.33:5060;branch=z9hG4bK06b0c82f
From: <sip:[email protected]>;tag=as5774f7cd
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 104 REGISTER
User-Agent: Asterisk PBX
Authorization: Digest username="9627349", realm="sipgate.at", algorithm=MD5, uri="sip:sipgate.at", nonce="422b00fc993b4ad91b1ae97e5cbd78139dfc9bf7", response="81ff5c007d1ac3d7b4324325506f3645", opaque=""
Expires: 120
Contact: <sip:[email protected]>
Event: registration
Content-Length: 0
(no NAT) to 212.236.252.3:5060
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 62.99.193.33:5060;branch=z9hG4bK06b0c82f
From: <sip:[email protected]>;tag=as5774f7cd
To: <sip:[email protected]>;tag=b1157e02d4a58892308b0557a5fe5724.591c
Call-ID: [email protected]
CSeq: 104 REGISTER
Contact: <sip:[email protected]>;q=0.00;expires=120
Server: sipgate ser
Content-Length: 0
Warning: 392 212.236.252.3:5060 "Noisy feedback tells: pid=10533 req_src_ip=62.99.193.33 req_src_port=5060 in_uri=sip:sipgate.at out_uri=sip:sipgate.at via_cnt==1"
10 headers, 0 lines
Mar 6 14:34:16 NOTICE[4255]: chan_sip.c:7521 handle_response: Outbound Registration: Expiry for sipgate.at is 120 sec (Scheduling reregistration in 105000 ms)
Destroying call '[email protected]'
-- Executing SetCallerID("SIP/10-1d1b", ""9627349"") in new stack
-- Executing SetCIDName("SIP/10-1d1b", ""9627349"") in new stack
-- Executing Dial("SIP/10-1d1b", "SIP/0800664664@sipgate|30|tr") in new stack
-- Called 0800664664@sipgate
-- Got SIP response 500 "I'm terribly sorry, server error occured (1/SL)" back from 212.236.252.3
-- SIP/sipgate-1a3d is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Wait("SIP/10-1d1b", "") in new stack
-- Executing Hangup("SIP/10-1d1b", "") in new stack
== Spawn extension (default, 00800664664, 5) exited non-zero on 'SIP/10-1d1b'
Sip read:
0 headers, 0 lines
-- Executing SetCallerID("SIP/10-1358", ""9627349"") in new stack
-- Executing SetCIDName("SIP/10-1358", ""9627349"") in new stack
-- Executing Dial("SIP/10-1358", "SIP/0800664664@sipgate|30|tr") in new stack
We're at 62.99.193.33 port 14530
Answering/Requesting with root capability 0x4 (ulaw)
Answering with preferred capability 0x2 (gsm)
Answering with preferred capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
12 headers, 12 lines
Reliably Transmitting:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 62.99.193.33:5060;branch=z9hG4bK00484894;rport
From: ""9627349"" <sip:[email protected]>;tag=as4e4ece7f
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email][email protected][/email]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sun, 06 Mar 2005 13:35:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 261
v=0
o=root 4255 4255 IN IP4 62.99.193.33
s=session
c=IN IP4 62.99.193.33
t=0 0
m=audio 14530 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
(NAT) to 212.236.252.3:5060
-- Called 0800664664@sipgate
Sip read:
SIP/2.0 500 I'm terribly sorry, server error occured (1/SL)
Via: SIP/2.0/UDP 62.99.193.33:5060;branch=z9hG4bK00484894;rport=5060
From: ""9627349"" <sip:[email protected]>;tag=as4e4ece7f
To: <sip:[email protected]>;tag=b1157e02d4a58892308b0557a5fe5724.bec8
Call-ID: [email][email protected][/email]
CSeq: 102 INVITE
Server: sipgate ser
Content-Length: 0
Warning: 392 212.236.252.3:5060 "Noisy feedback tells: pid=10528 req_src_ip=62.99.193.33 req_src_port=5060 in_uri=sip:[email protected] out_uri=sip:[email protected] via_cnt==1"
9 headers, 0 lines
-- Got SIP response 500 "I'm terribly sorry, server error occured (1/SL)" back from 212.236.252.3
Transmitting:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 62.99.193.33:5060;branch=z9hG4bK00484894;rport
From: ""9627349"" <sip:[email protected]>;tag=as4e4ece7f
To: <sip:[email protected]>;tag=b1157e02d4a58892308b0557a5fe5724.bec8
Contact: <sip:[email protected]>
Call-ID: [email][email protected][/email]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0
(NAT) to 212.236.252.3:5060
-- SIP/sipgate-b50d is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Wait("SIP/10-1358", "") in new stack
-- Executing Hangup("SIP/10-1358", "") in new stack
== Spawn extension (default, 00800664664, 5) exited non-zero on 'SIP/10-1358'
Destroying call '[email protected]'
11 headers, 0 lines
Reliably Transmitting:
OPTIONS sip:sipgate.at SIP/2.0
Via: SIP/2.0/UDP 62.99.193.33:5060;branch=z9hG4bK62b0d3ec
From: "asterisk" <sip:[email protected]>;tag=as29370723
To: <sip:sipgate.at>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Sun, 06 Mar 2005 13:35:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Length: 0
(no NAT) to 212.236.252.3:5060
Sip read:
SIP/2.0 482 Loop Detected
Via: SIP/2.0/UDP 62.99.193.33:5060;branch=z9hG4bK62b0d3ec
From: "asterisk" <sip:[email protected]>;tag=as29370723
To: <sip:sipgate.at>;tag=b1157e02d4a58892308b0557a5fe5724.7840
Call-ID: [email protected]
CSeq: 102 OPTIONS
Server: sipgate ser
Content-Length: 0
Warning: 392 212.236.252.3:5060 "Noisy feedback tells: pid=10530 req_src_ip=62.99.193.33 req_src_port=5060 in_uri=sip:sipgate.at out_uri=sip:sipgate.at via_cnt==1"
9 headers, 0 lines
Destroying call '[email protected]'
exten => 101,1,Dial(SIP/613@fwd,30,tr)
exten => 102,1,Dial(SIP/10000@sipgate,30,tr)
Executing Dial....
Called 613@fwd
Sip/fwd-818d is ringing
Sip/fwd-818d answered SIP/10-3efb
Attempting Native Bridge of SIP/10-3efb and SIP/fwd-818d
;general dialout to fwd
exten => _393.,1,SetCallerID(${CALLERID})
exten => _393.,2,SetCIDName(${CIDNAME})
exten => _393.,3,Dial(SIP/${EXTEN:3}@fwd.pulver.com,60,r)
Maik schrieb:Asterisk hat leider (noch) keinen STUN Support. Das wuerde das ganze naemlich viel einfacher machen.