14 Dec
2007
14 Dec
'07
1 a.m.
Internal UA's connecting to the sip proxy work fine, but as soon an External UA
connects to the sip proxy via a NAT'd public IP address - the call lasts for approx 20
seconds then disconnects. From what I can see openser is attempting to respond to
asterisk after the session is initiated via the public IP NAT'd address - and is
unable since asterisk and openser are behind the firewall on the internal network ( ELBOW
routing ). Any help getting this working would be most appreciated, I would also like to
document the issue to help others that encounter similar problems as mine.
External phones:
UA -> [(public IP) Firewall (NAT to localip)] -> openser -> asterisk
Internal phones:
UA -> openser -> asterisk
#### Layout ( asterisk and openser are on the same machine )
NATD public ip = xxx.206.xxx.136 -> 10.3.1.33
phone ip = xxx.206.xxx.137
openser = 10.3.1.33 p5060
asteriskSIP = 10.3.1.34 p5060
#### Asterisk SIP Config
[general]
canreinvite=no
externip=xxx.206.xxx.136
localnet=10.3.1.0/255.255.255.0
context=default
bindport=5060
bindaddr=10.3.1.34
sipdebug=yes
[openser]
type=friend
context=default
insecure=very
externalnotify=yes
allow=all
host=10.3.1.33
#### SER LOG
Dec 13 22:15:03 phonesys-slave openser[15976]: New request - M=INVITE
RURI=sip:500@xxx.206.xxx.136:5060;user=phone F=sip:pbaker@xxx.206.xxx.136
T=sip:500@xxx.206.xxx.136;user=phone IP=xxx.206.xxx.137
ID=f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137
Dec 13 22:15:03 phonesys-slave openser[15976]: Callee is not local - M=INVITE
RURI=sip:500@xxx.206.xxx.136:5060;user=phone F=sip:pbaker@xxx.206.xxx.136
T=sip:500@xxx.206.xxx.136;user=phone IP=xxx.206.xxx.137
ID=f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137
Dec 13 22:15:03 phonesys-slave openser[15976]: Request leaving server,
D-URI='<null>' - M=INVITE RURI=sip:500@10.3.1.34:5060;transport=udp
F=sip:pbaker@xxx.206.xxx.136 T=sip:500@xxx.206.xxx.136;user=phone IP=xxx.206.xxx.137
ID=f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137
Dec 13 22:15:03 phonesys-slave openser[15973]: Reply - S=100 D=Trying
F=sip:pbaker@xxx.206.xxx.136 T=sip:500@xxx.206.xxx.136;user=phone IP=10.3.1.34
ID=f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137
Dec 13 22:15:03 phonesys-slave openser[15978]: Reply - S=200 D=OK
F=sip:pbaker@xxx.206.xxx.136 T=sip:500@xxx.206.xxx.136;user=phone IP=10.3.1.34
ID=f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137
Dec 13 22:15:04 phonesys-slave openser[15989]: Reply - S=200 D=OK
F=sip:pbaker@xxx.206.xxx.136 T=sip:500@xxx.206.xxx.136;user=phone IP=10.3.1.34
ID=f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137
Dec 13 22:15:05 phonesys-slave openser[15986]: Reply - S=200 D=OK
F=sip:pbaker@xxx.206.xxx.136 T=sip:500@xxx.206.xxx.136;user=phone IP=10.3.1.34
ID=f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137
Dec 13 22:15:07 phonesys-slave openser[15976]: Reply - S=200 D=OK
F=sip:pbaker@xxx.206.xxx.136 T=sip:500@xxx.206.xxx.136;user=phone IP=10.3.1.34
ID=f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137
#### Asterisk LOG
*CLI> [Dec 13 22:15:03]
<--- SIP read from 10.3.1.33:5060 --->
INVITE sip:500@10.3.1.34:5060;transport=udp SIP/2.0
Record-Route: <sip:10.3.1.33;lr;ftag=CFE2AAA1-5B882804>
Via: SIP/2.0/UDP 10.3.1.33;branch=z9hG4bK2fc1.94c7bb96.0
Via: SIP/2.0/UDP xxx.206.xxx.137;rport=5060;branch=z9hG4bKcbd9a65f8D4E5FE0
From: "pbaker" <sip:pbaker@xxx.206.xxx.136>;tag=CFE2AAA1-5B882804
To: <sip:500@xxx.206.xxx.136;user=phone>
CSeq: 2 INVITE
Call-ID: f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137
Contact: <sip:pbaker@xxx.206.xxx.137>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE,
REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.1.0.2708
Supported: 100rel,replaces
Allow-Events: talk,hold,conference
Max-Forwards: 69
Content-Type: application/sdp
Content-Length: 254
v=0
o=- 978310717 978310717 IN IP4 xxx.206.xxx.137
s=Polycom IP Phone
c=IN IP4 xxx.206.xxx.137
t=0 0
m=audio 35022 RTP/AVP 0 8 18 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
<------------->
[Dec 13 22:15:03] --- (16 headers 11 lines) ---
[Dec 13 22:15:03] == Using TOS bits 0
[Dec 13 22:15:03] == Using CoS mark 5
[Dec 13 22:15:03] Sending to 10.3.1.33 : 5060 (no NAT)
[Dec 13 22:15:03] Using INVITE request as basis request -
f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137
[Dec 13 22:15:03] No user 'pbaker' in SIP users list
[Dec 13 22:15:03] Found peer 'openser' for 'pbaker' from 10.3.1.33:5060
[Dec 13 22:15:03] Found RTP audio format 0
[Dec 13 22:15:03] Found RTP audio format 8
[Dec 13 22:15:03] Found RTP audio format 18
[Dec 13 22:15:03] Found RTP audio format 101
[Dec 13 22:15:03] Peer audio RTP is at port xxx.206.xxx.137:35022
[Dec 13 22:15:03] Found audio description format PCMU for ID 0
[Dec 13 22:15:03] Found audio description format PCMA for ID 8
[Dec 13 22:15:03] Found audio description format G729 for ID 18
[Dec 13 22:15:03] Found audio description format telephone-event for ID 101
[Dec 13 22:15:03] Capabilities: us - 0x27f9fff
(g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|t140),
peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined -
0x10c (ulaw|alaw|g729)
[Dec 13 22:15:03] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
[Dec 13 22:15:03] Peer audio RTP is at port xxx.206.xxx.137:35022
[Dec 13 22:15:03] Looking for 500 in default (domain 10.3.1.34)
[Dec 13 22:15:03] list_route: hop: <sip:10.3.1.33;lr;ftag=CFE2AAA1-5B882804>
[Dec 13 22:15:03]
<--- Transmitting (no NAT) to 10.3.1.33:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.3.1.33;branch=z9hG4bK2fc1.94c7bb96.0;received=10.3.1.33
Via: SIP/2.0/UDP xxx.206.xxx.137;rport=5060;branch=z9hG4bKcbd9a65f8D4E5FE0
Record-Route: <sip:10.3.1.33;lr;ftag=CFE2AAA1-5B882804>
From: "pbaker" <sip:pbaker@xxx.206.xxx.136>;tag=CFE2AAA1-5B882804
To: <sip:500@xxx.206.xxx.136;user=phone>
Call-ID: f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137
CSeq: 2 INVITE
User-Agent: Asterisk PBX SVN-trunk-r92779
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:500@10.3.1.34>
Content-Length: 0
<------------>
[Dec 13 22:15:03] -- Executing [500@default:1]
Playback("SIP/xxx.206.xxx.136-090b5380", "demo-abouttotry") in new
stack
[Dec 13 22:15:03] Audio is at 10.3.1.34 port 47068
[Dec 13 22:15:03] Adding codec 0x4 (ulaw) to SDP
[Dec 13 22:15:03] Adding codec 0x8 (alaw) to SDP
[Dec 13 22:15:03] Adding codec 0x100 (g729) to SDP
[Dec 13 22:15:03] Adding non-codec 0x1 (telephone-event) to SDP
[Dec 13 22:15:03]
<--- Reliably Transmitting (no NAT) to 10.3.1.33:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.3.1.33;branch=z9hG4bK2fc1.94c7bb96.0;received=10.3.1.33
Via: SIP/2.0/UDP xxx.206.xxx.137;rport=5060;branch=z9hG4bKcbd9a65f8D4E5FE0
Record-Route: <sip:10.3.1.33;lr;ftag=CFE2AAA1-5B882804>
From: "pbaker" <sip:pbaker@xxx.206.xxx.136>;tag=CFE2AAA1-5B882804
To: <sip:500@xxx.206.xxx.136;user=phone>;tag=as26a99907
Call-ID: f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137
CSeq: 2 INVITE
User-Agent: Asterisk PBX SVN-trunk-r92779
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:500@10.3.1.34>
Content-Type: application/sdp
Content-Length: 333
v=0
o=root 45593872 45593872 IN IP4 10.3.1.34
s=Asterisk PBX SVN-trunk-r92779
c=IN IP4 10.3.1.34
t=0 0
m=audio 47068 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
[Dec 13 22:15:04] -- <SIP/xxx.206.xxx.136-090b5380> Playing
'demo-abouttotry.slin' (language 'en')
[Dec 13 22:15:04] Retransmitting #1 (no NAT) to 10.3.1.33:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.3.1.33;branch=z9hG4bK2fc1.94c7bb96.0;received=10.3.1.33
Via: SIP/2.0/UDP xxx.206.xxx.137;rport=5060;branch=z9hG4bKcbd9a65f8D4E5FE0
Record-Route: <sip:10.3.1.33;lr;ftag=CFE2AAA1-5B882804>
From: "pbaker" <sip:pbaker@xxx.206.xxx.136>;tag=CFE2AAA1-5B882804
To: <sip:500@xxx.206.xxx.136;user=phone>;tag=as26a99907
Call-ID: f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137
CSeq: 2 INVITE
User-Agent: Asterisk PBX SVN-trunk-r92779
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:500@10.3.1.34>
Content-Type: application/sdp
Content-Length: 333
v=0
o=root 45593872 45593872 IN IP4 10.3.1.34
s=Asterisk PBX SVN-trunk-r92779
c=IN IP4 10.3.1.34
t=0 0
m=audio 47068 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
[Dec 13 22:15:05] Retransmitting #2 (no NAT) to 10.3.1.33:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.3.1.33;branch=z9hG4bK2fc1.94c7bb96.0;received=10.3.1.33
Via: SIP/2.0/UDP xxx.206.xxx.137;rport=5060;branch=z9hG4bKcbd9a65f8D4E5FE0
Record-Route: <sip:10.3.1.33;lr;ftag=CFE2AAA1-5B882804>
From: "pbaker" <sip:pbaker@xxx.206.xxx.136>;tag=CFE2AAA1-5B882804
To: <sip:500@xxx.206.xxx.136;user=phone>;tag=as26a99907
Call-ID: f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137
CSeq: 2 INVITE
User-Agent: Asterisk PBX SVN-trunk-r92779
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:500@10.3.1.34>
Content-Type: application/sdp
Content-Length: 333
v=0
o=root 45593872 45593872 IN IP4 10.3.1.34
s=Asterisk PBX SVN-trunk-r92779
c=IN IP4 10.3.1.34
t=0 0
m=audio 47068 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
[Dec 13 22:15:07] Retransmitting #3 (no NAT) to 10.3.1.33:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.3.1.33;branch=z9hG4bK2fc1.94c7bb96.0;received=10.3.1.33
Via: SIP/2.0/UDP xxx.206.xxx.137;rport=5060;branch=z9hG4bKcbd9a65f8D4E5FE0
Record-Route: <sip:10.3.1.33;lr;ftag=CFE2AAA1-5B882804>
From: "pbaker" <sip:pbaker@xxx.206.xxx.136>;tag=CFE2AAA1-5B882804
To: <sip:500@xxx.206.xxx.136;user=phone>;tag=as26a99907
Call-ID: f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137
CSeq: 2 INVITE
User-Agent: Asterisk PBX SVN-trunk-r92779
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:500@10.3.1.34>
Content-Type: application/sdp
Content-Length: 333
v=0
o=root 45593872 45593872 IN IP4 10.3.1.34
s=Asterisk PBX SVN-trunk-r92779
c=IN IP4 10.3.1.34
t=0 0
m=audio 47068 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
[Dec 13 22:15:11] Retransmitting #4 (no NAT) to 10.3.1.33:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.3.1.33;branch=z9hG4bK2fc1.94c7bb96.0;received=10.3.1.33
Via: SIP/2.0/UDP xxx.206.xxx.137;rport=5060;branch=z9hG4bKcbd9a65f8D4E5FE0
Record-Route: <sip:10.3.1.33;lr;ftag=CFE2AAA1-5B882804>
From: "pbaker" <sip:pbaker@xxx.206.xxx.136>;tag=CFE2AAA1-5B882804
To: <sip:500@xxx.206.xxx.136;user=phone>;tag=as26a99907
Call-ID: f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137
CSeq: 2 INVITE
User-Agent: Asterisk PBX SVN-trunk-r92779
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:500@10.3.1.34>
Content-Type: application/sdp
Content-Length: 333
v=0
o=root 45593872 45593872 IN IP4 10.3.1.34
s=Asterisk PBX SVN-trunk-r92779
c=IN IP4 10.3.1.34
t=0 0
m=audio 47068 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
[Dec 13 22:15:15] Retransmitting #5 (no NAT) to 10.3.1.33:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.3.1.33;branch=z9hG4bK2fc1.94c7bb96.0;received=10.3.1.33
Via: SIP/2.0/UDP xxx.206.xxx.137;rport=5060;branch=z9hG4bKcbd9a65f8D4E5FE0
Record-Route: <sip:10.3.1.33;lr;ftag=CFE2AAA1-5B882804>
From: "pbaker" <sip:pbaker@xxx.206.xxx.136>;tag=CFE2AAA1-5B882804
To: <sip:500@xxx.206.xxx.136;user=phone>;tag=as26a99907
Call-ID: f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137
CSeq: 2 INVITE
User-Agent: Asterisk PBX SVN-trunk-r92779
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:500@10.3.1.34>
Content-Type: application/sdp
Content-Length: 333
v=0
o=root 45593872 45593872 IN IP4 10.3.1.34
s=Asterisk PBX SVN-trunk-r92779
c=IN IP4 10.3.1.34
t=0 0
m=audio 47068 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
[Dec 13 22:15:18] -- Executing [500@default:2]
Dial("SIP/xxx.206.xxx.136-090b5380",
"IAX2/guest@misery.digium.com/s@default") in new stack
[Dec 13 22:15:18] -- Called guest@misery.digium.com/s@default
[Dec 13 22:15:18] -- Call accepted by 216.207.245.8 (format gsm)
[Dec 13 22:15:18] -- Format for call is gsm
[Dec 13 22:15:18] -- IAX2/216.207.245.8:4569-1 is ringing
[Dec 13 22:15:19] -- IAX2/216.207.245.8:4569-1 answered SIP/xxx.206.xxx.136-090b5380
[Dec 13 22:15:19] Retransmitting #6 (no NAT) to 10.3.1.33:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.3.1.33;branch=z9hG4bK2fc1.94c7bb96.0;received=10.3.1.33
Via: SIP/2.0/UDP xxx.206.xxx.137;rport=5060;branch=z9hG4bKcbd9a65f8D4E5FE0
Record-Route: <sip:10.3.1.33;lr;ftag=CFE2AAA1-5B882804>
From: "pbaker" <sip:pbaker@xxx.206.xxx.136>;tag=CFE2AAA1-5B882804
To: <sip:500@xxx.206.xxx.136;user=phone>;tag=as26a99907
Call-ID: f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137
CSeq: 2 INVITE
User-Agent: Asterisk PBX SVN-trunk-r92779
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:500@10.3.1.34>
Content-Type: application/sdp
Content-Length: 333
v=0
o=root 45593872 45593872 IN IP4 10.3.1.34
s=Asterisk PBX SVN-trunk-r92779
c=IN IP4 10.3.1.34
t=0 0
m=audio 47068 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
[Dec 13 22:15:23] WARNING[16421]: chan_sip.c:2337 retrans_pkt: Maximum retries exceeded on
transmission f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 for seqno 2 (Critical Response)
[Dec 13 22:15:23] WARNING[16421]: chan_sip.c:2364 retrans_pkt: Hanging up call
f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 - no reply to our critical packet.
[Dec 13 22:15:23] -- Hungup 'IAX2/216.207.245.8:4569-1'
[Dec 13 22:15:23] == Spawn extension (default, 500, 2) exited non-zero on
'SIP/xxx.206.xxx.136-090b5380'
[Dec 13 22:15:24] Really destroying SIP dialog
'f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137' Method: INVITE
#### PCAP Diagram
|Time | xxx.206.xxx.137 | 10.3.1.33 |
|5.056 | Request: INVITE sip |SIP/SDP: Request: INVITE
sip:500@xxx.206.xxx.136:5060;user=phone, with session description
| |(5060) ------------------> (5060) |
|5.058 | Status: 100 Trying |SIP: Status: 100 Trying
| |(5060) <------------------ (5060) |
|5.058 | Status: 407 Proxy A |SIP: Status: 407 Proxy Authentication
Required
| |(5060) <------------------ (5060) |
|5.140 | Request: ACK sip:50 |SIP: Request: ACK
sip:500@xxx.206.xxx.136:5060
| |(5060) ------------------> (5060) |
|5.165 | Request: INVITE sip |SIP/SDP: Request: INVITE
sip:500@xxx.206.xxx.136:5060;user=phone, with session description
| |(5060) ------------------> (5060) |
|5.165 | Status: 100 Trying |SIP: Status: 100 Trying
| |(5060) <------------------ (5060) |
|5.168 | Status: 200 OK, wit |SIP/SDP: Status: 200 OK, with session
description
| |(5060) <------------------ (5060) |
|6.169 | Status: 200 OK, wit |SIP/SDP: Status: 200 OK, with session
description
| |(5060) <------------------ (5060) |
|7.168 | Status: 200 OK, wit |SIP/SDP: Status: 200 OK, with session
description
| |(5060) <------------------ (5060) |
|9.168 | Status: 200 OK, wit |SIP/SDP: Status: 200 OK, with session
description
| |(5060) <------------------ (5060) |
|13.169 | Status: 200 OK, wit |SIP/SDP: Status: 200 OK, with session
description
| |(5060) <------------------ (5060) |
|17.169 | Status: 200 OK, wit |SIP/SDP: Status: 200 OK, with session
description
| |(5060) <------------------ (5060) |
|21.169 | Status: 200 OK, wit |SIP/SDP: Status: 200 OK, with session
description
| |(5060) <------------------ (5060) |
#### OpenSER Config 1.2
########################################################################
# This configuration is autogenerated by sip:wizard
# (http://www.sipwise.com/wizard) on Thu Dec 13 22:43:43 +0100 2007
# for OpenSER 1.2
#
# Copyright (C) 2007 Sipwise (support(a)sipwise.com)
########################################################################
########################################################################
# By obtaining, using, and/or copying this configuration and/or its
# associated documentation, you agree that you have read, understood,
# and will comply with the Terms of Usage provided at
# http://www.sipwise.com/news/?page_id=6 as well as the following
# additions:
#
# Permission to use, copy, modify, and distribute this configuration and
# its associated documentation for any purpose and without fee is hereby
# granted, provided that the above copyright notice appears in all
# copies, and that both that copyright notice and this permission notice
# appear in supporting documentation, and that the name of Sipwise or
# the author will not be used in advertising or publicity pertaining to
# distribution of the configuration without specific, written prior
# permission.
########################################################################
########################################################################
# Before using this configuration, read the following prerequisites in
# order to gain the designated functionallity:
#
# base:
# You have to insert all locally served domains (i.e.
# "openserctl domain add your.domain.com").
#
# nat-rtpproxy:
# You have to install RTPProxy
# (http://www.openser.org/downloads/snapshots/rtpproxy/) for relaying
# RTP traffic.
#
# offnet-pstn:
# You have to add a routing entry for lcr (i.e. "openserctl lcr
# addroute '' '' 1 1"). Additionally, you have to add your
gateways
# (i.e. "openserctl lcr addgw my-test-gw 1.2.3.4 5060 sip udp 1").
#
########################################################################
########################################################################
# Configuration 'sip:wizard - Thu Dec 13 22:43:43 +0100 2007'
########################################################################
listen = udp:10.3.1.33:5060
mpath = "/usr/local/lib/openser/modules"
children = 8
debug = 3
fork = yes
group = "openser"
user = "openser"
disable_tcp = no
log_facility = LOG_DAEMON
log_stderror = no
tcp_children = 4
mhomed = no
server_signature = yes
sock_group = "openser"
sock_mode = 0600
sock_user = "openser"
unix_sock = "/tmp/openser.sock"
unix_sock_children = 1
reply_to_via = no
sip_warning = yes
check_via = no
dns = no
rev_dns = no
disable_core_dump = no
dns_try_ipv6 = yes
dns_use_search_list = yes
loadmodule "usrloc.so"
modparam("usrloc", "user_column", "username")
modparam("usrloc", "domain_column", "domain")
modparam("usrloc", "contact_column", "contact")
modparam("usrloc", "expires_column", "expires")
modparam("usrloc", "q_column", "q")
modparam("usrloc", "callid_column", "callid")
modparam("usrloc", "cseq_column", "cseq")
modparam("usrloc", "methods_column", "methods")
modparam("usrloc", "flags_column", "flags")
modparam("usrloc", "user_agent_column", "user_agent")
modparam("usrloc", "received_column", "received")
modparam("usrloc", "socket_column", "socket")
modparam("usrloc", "use_domain", 0)
modparam("usrloc", "desc_time_order", 0)
modparam("usrloc", "timer_interval", 60)
modparam("usrloc", "db_url",
"mysql://openser:openserrw@localhost/openser")
modparam("usrloc", "db_mode", 1)
modparam("usrloc", "matching_mode", 0)
modparam("usrloc", "cseq_delay", 20)
modparam("usrloc", "nat_bflag", 6)
loadmodule "textops.so"
loadmodule "rr.so"
modparam("rr", "enable_full_lr", 0)
modparam("rr", "append_fromtag", 1)
modparam("rr", "enable_double_rr", 1)
modparam("rr", "add_username", 0)
loadmodule "tm.so"
modparam("tm", "fr_timer", 30)
modparam("tm", "fr_inv_timer", 120)
modparam("tm", "wt_timer", 5)
modparam("tm", "delete_timer", 2)
modparam("tm", "noisy_ctimer", 0)
modparam("tm", "ruri_matching", 1)
modparam("tm", "via1_matching", 1)
modparam("tm", "unix_tx_timeout", 2)
modparam("tm", "restart_fr_on_each_reply", 1)
modparam("tm", "pass_provisional_replies", 0)
loadmodule "xlog.so"
modparam("xlog", "buf_size", 4096)
modparam("xlog", "force_color", 0)
loadmodule "mi_fifo.so"
modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo")
modparam("mi_fifo", "fifo_mode", 0660)
modparam("mi_fifo", "fifo_group", "openser")
modparam("mi_fifo", "fifo_user", "openser")
modparam("mi_fifo", "reply_dir", "/tmp/")
modparam("mi_fifo", "reply_indent", "\t")
loadmodule "domain.so"
modparam("domain", "db_url",
"mysql://openser:openserrw@localhost/openser")
modparam("domain", "db_mode", 1)
modparam("domain", "domain_table", "domain")
modparam("domain", "domain_col", "domain")
loadmodule "nathelper.so"
modparam("nathelper", "natping_interval", 60)
modparam("nathelper", "ping_nated_only", 1)
modparam("nathelper", "rtpproxy_sock",
"unix:/var/run/rtpproxy.sock")
modparam("nathelper", "rtpproxy_disable", 0)
modparam("nathelper", "rtpproxy_disable_tout", 60)
modparam("nathelper", "rtpproxy_tout", 1)
modparam("nathelper", "rtpproxy_retr", 5)
modparam("nathelper", "sipping_method", "OPTIONS")
modparam("nathelper", "received_avp", "$avp(i:801)")
loadmodule "sl.so"
modparam("sl", "enable_stats", 1)
loadmodule "uri.so"
loadmodule "registrar.so"
modparam("registrar", "default_expires", 3600)
modparam("registrar", "min_expires", 60)
modparam("registrar", "max_expires", 0)
modparam("registrar", "default_q", 0)
modparam("registrar", "append_branches", 1)
modparam("registrar", "case_sensitive", 0)
modparam("registrar", "received_param", "received")
modparam("registrar", "max_contacts", 0)
modparam("registrar", "retry_after", 0)
modparam("registrar", "method_filtering", 0)
modparam("registrar", "path_mode", 2)
modparam("registrar", "path_use_received", 0)
modparam("registrar", "received_avp", "$avp(i:801)")
loadmodule "maxfwd.so"
modparam("maxfwd", "max_limit", 256)
loadmodule "mysql.so"
modparam("mysql", "ping_interval", 300)
modparam("mysql", "auto_reconnect", 1)
loadmodule "auth.so"
modparam("auth", "nonce_expire", 300)
modparam("auth", "rpid_suffix",
";party=calling;id-type=subscriber;screen=yes")
modparam("auth", "rpid_avp", "$avp(s:rpid)")
loadmodule "auth_db.so"
modparam("auth_db", "db_url",
"mysql://openser:openserrw@localhost/openser")
modparam("auth_db", "user_column", "username")
modparam("auth_db", "domain_column", "domain")
modparam("auth_db", "password_column", "password")
modparam("auth_db", "password_column_2", "ha1b")
modparam("auth_db", "calculate_ha1", 1)
modparam("auth_db", "use_domain", 0)
modparam("auth_db", "load_credentials", "rpid")
loadmodule "uri_db.so"
modparam("uri_db", "db_url",
"mysql://openser:openserrw@localhost/openser")
modparam("uri_db", "uri_table", "uri")
modparam("uri_db", "uri_user_column", "username")
modparam("uri_db", "uri_domain_column", "domain")
modparam("uri_db", "uri_uriuser_column", "uri_user")
modparam("uri_db", "subscriber_table", "subscriber")
modparam("uri_db", "subscriber_user_column", "username")
modparam("uri_db", "subscriber_domain_column", "domain")
modparam("uri_db", "use_uri_table", 0)
modparam("uri_db", "use_domain", 0)
loadmodule "lcr.so"
modparam("lcr", "db_url",
"mysql://openser:openserrw@localhost/openser")
modparam("lcr", "gw_table", "gw")
modparam("lcr", "gw_name_column", "gw_name")
modparam("lcr", "ip_addr_column", "ip_addr")
modparam("lcr", "port_column", "port")
modparam("lcr", "uri_scheme_column", "uri_scheme")
modparam("lcr", "transport_column", "transport")
modparam("lcr", "grp_id_column", "grp_id")
modparam("lcr", "lcr_table", "lcr")
modparam("lcr", "strip_column", "strip")
modparam("lcr", "prefix_column", "prefix")
modparam("lcr", "from_uri_column", "from_uri")
modparam("lcr", "priority_column", "priority")
modparam("lcr", "gw_uri_avp", "1400")
modparam("lcr", "ruri_user_avp", "1402")
modparam("lcr", "contact_avp", "1401")
modparam("lcr", "fr_inv_timer_avp", "s:fr_inv_timer_avp")
modparam("lcr", "fr_inv_timer", 90)
modparam("lcr", "fr_inv_timer_next", 30)
modparam("lcr", "rpid_avp", "s:rpid")
########################################################################
# Request route 'main'
########################################################################
route[0]
{
xlog("L_INFO", "New request - M=$rm RURI=$ru F=$fu T=$tu IP=$si
ID=$ci\n");
force_rport();
if(msg:len > max_len)
{
xlog("L_INFO", "Message too big - M=$rm RURI=$ru F=$fu
T=$tu IP=$si ID=$ci\n");
sl_send_reply("513", "Message Too Big");
exit;
}
if (!mf_process_maxfwd_header("10"))
{
xlog("L_INFO", "Too many hops - M=$rm RURI=$ru F=$fu T=$tu
IP=$si ID=$ci\n");
sl_send_reply("483", "Too Many Hops");
exit;
}
if(!is_method("REGISTER"))
{
if(nat_uac_test("19"))
{
record_route(";nat=yes");
}
else
{
record_route();
}
}
if(is_method("CANCEL") || is_method("BYE"))
{
unforce_rtp_proxy();
}
if(loose_route())
{
if(!has_totag())
{
xlog("L_INFO", "Initial loose-routing rejected -
M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
sl_send_reply("403", "Initial Loose-Routing
Rejected");
exit;
}
if(nat_uac_test("19") || search("^Route:.*;nat=yes"))
{
fix_nated_contact();
setbflag(6);
}
route(3);
}
if(is_method("REGISTER"))
{
route(2);
}
if(is_method("INVITE"))
{
route(4);
}
if(is_method("CANCEL") || is_method("ACK"))
{
route(8);
}
route(9);
}
########################################################################
# Request route 'stop-rtp-proxy'
########################################################################
route[1]
{
if(isflagset(22))
{
unforce_rtp_proxy();
}
}
########################################################################
# Request route 'base-route-register'
########################################################################
route[2]
{
sl_send_reply("100", "Trying");
if(!www_authorize("", "subscriber"))
{
xlog("L_INFO", "Register authentication failed - M=$rm
RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
www_challenge("", "0");
exit;
}
if(!check_to())
{
xlog("L_INFO", "Spoofed To-URI detected - M=$rm RURI=$ru
F=$fu T=$tu IP=$si ID=$ci\n");
sl_send_reply("403", "Spoofed To-URI Detected");
exit;
}
consume_credentials();
if(!search("^Contact:[ ]*\*") && nat_uac_test("19"))
{
fix_nated_register();
setbflag(6);
}
if(!save("location"))
{
xlog("L_ERR", "Saving contact failed - M=$rm RURI=$ru F=$fu
T=$tu IP=$si ID=$ci\n");
sl_reply_error();
exit;
}
xlog("L_INFO", "Registration successful - M=$rm RURI=$ru F=$fu
T=$tu IP=$si ID=$ci\n");
exit;
}
########################################################################
# Request route 'base-outbound'
########################################################################
route[3]
{
if(isbflagset(6))
{
if(!isflagset(22) && !search("^Content-Length:[ ]*0"))
{
setflag(22);
force_rtp_proxy();
}
t_on_reply("2");
}
else
{
t_on_reply("1");
}
if(!isflagset(21))
{
t_on_failure("2");
}
if(isflagset(29))
{
append_branch();
}
if(is_present_hf("Proxy-Authorization"))
{
consume_credentials();
}
xlog("L_INFO", "Request leaving server, D-URI='$du' - M=$rm
RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
# no 100 (we already sent it) and no DNS blacklisting
if(!t_relay("0x05"))
{
sl_reply_error();
if(is_method("INVITE") && isbflagset(6))
{
unforce_rtp_proxy();
}
}
exit;
}
########################################################################
# Request route 'base-route-invite'
########################################################################
route[4]
{
sl_send_reply("100", "Trying");
if(from_gw())
{
xlog("L_INFO", "Call from PSTN' - M=$rm RURI=$ru F=$fu
T=$tu IP=$si ID=$ci\n");
setflag(23);
}
else
{
if(!proxy_authorize("", "subscriber"))
{
xlog("L_INFO", "Proxy authentication failed - M=$rm
RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
proxy_challenge("", "0");
exit;
}
if(!check_from())
{
xlog("L_INFO", "Spoofed From-URI detected - M=$rm
RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
sl_send_reply("403", "Spoofed From-URI
Detected");
exit;
}
}
if(nat_uac_test("19"))
{
fix_nated_contact();
setbflag(6);
}
route(5);
}
########################################################################
# Request route 'invite-find-callee'
########################################################################
route[5]
{
if(!is_domain_local("$rd"))
{
setflag(20);
route(7);
}
if(does_uri_exist())
{
xlog("L_INFO", "Callee is local - M=$rm RURI=$ru F=$fu
T=$tu IP=$si ID=$ci\n");
route(6);
}
else
{
xlog("L_INFO", "Callee is not local - M=$rm RURI=$ru F=$fu
T=$tu IP=$si ID=$ci\n");
route(7);
}
exit;
}
########################################################################
# Request route 'invite-to-internal'
########################################################################
route[6]
{
if(!lookup("location"))
{
xlog("L_INFO", "Local user offline - M=$rm RURI=$ru F=$fu
T=$tu IP=$si ID=$ci\n");
sl_send_reply("404", "User Offline");
}
else
{
xlog("L_INFO", "Local user online - M=$rm RURI=$ru F=$fu
T=$tu IP=$si ID=$ci\n");
route(3);
}
exit;
}
########################################################################
# Request route 'invite-to-external'
########################################################################
route[7]
{
if(isflagset(20))
{
xlog("L_INFO", "Call to foreign domain - M=$rm RURI=$ru
F=$fu T=$tu IP=$si ID=$ci\n");
route(3);
exit;
}
if(!isflagset(23))
{
# don't allow calls relaying from PSTN to PSTN, if not explicitely
forwarded
if(uri =~ "^sip:[0-9]+@")
{
# only route numeric users to PSTN
if(!load_gws())
{
xlog("L_ERR", "Error loading PSTN gateways
- M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
sl_send_reply("503", "PSTN Termination
Currently Unavailable");
exit;
}
if(!next_gw())
{
xlog("L_ERR", "No PSTN gateways available -
M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
sl_send_reply("503", "PSTN Termination
Currently Unavailable");
exit;
}
setflag(21);
t_on_failure("1");
route(3);
}
}
xlog("L_INFO", "Call to unknown user - M=$rm RURI=$ru F=$fu T=$tu
IP=$si ID=$ci\n");
sl_send_reply("404", "User Not Found");
exit;
}
########################################################################
# Request route 'base-route-local'
########################################################################
route[8]
{
t_on_reply("1");
if(t_check_trans())
{
xlog("L_INFO", "Request leaving server - M=$rm RURI=$ru
F=$fu T=$tu IP=$si ID=$ci\n");
if(!t_relay())
{
sl_reply_error();
}
}
else
{
xlog("L_INFO", "Dropping mis-routed request - M=$rm
RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
}
exit;
}
########################################################################
# Request route 'base-route-generic'
########################################################################
route[9]
{
xlog("L_INFO", "Method not supported - M=$rm RURI=$ru F=$fu T=$tu
IP=$si ID=$ci\n");
sl_send_reply("501", "Method Not Supported Here");
exit;
}
########################################################################
# Request route 'base-filter-failover'
########################################################################
route[10]
{
if(!t_check_status("408|500|503"))
{
xlog("L_INFO", "No failover routing needed for this
response code - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
route(1);
exit;
}
}
########################################################################
# Reply route 'base-standard-reply'
########################################################################
onreply_route[1]
{
xlog("L_INFO", "Reply - S=$rs D=$rr F=$fu T=$tu IP=$si
ID=$ci\n");
exit;
}
########################################################################
# Reply route 'base-nat-reply'
########################################################################
onreply_route[2]
{
xlog("L_INFO", "NAT-Reply - S=$rs D=$rr F=$fu T=$tu IP=$si
ID=$ci\n");
if(nat_uac_test("1"))
{
fix_nated_contact();
}
if(isbflagset(6) && status=~"(180)|(183)|2[0-9][0-9]")
{
if(!search("^Content-Length:[ ]*0"))
{
force_rtp_proxy();
}
}
exit;
}
########################################################################
# Failure route 'pstn-failover'
########################################################################
failure_route[1]
{
xlog("L_INFO", "Failure route for PSTN entered - M=$rm RURI=$ru
F=$fu T=$tu IP=$si ID=$ci\n");
route(10);
if(!next_gw())
{
xlog("L_ERR", "Failed to select next PSTN gateway - M=$rm
RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
route(1);
exit;
}
t_on_failure("1");
route(3);
}
########################################################################
# Failure route 'base-standard-failure'
########################################################################
failure_route[2]
{
route(10);
route(1);
}