Patrick:
Is your NAT SIP aware? What should be happening here is the 200 from
your asterisk box will be sent to openser, and relayed out to your
polycom. The polycom should send the ACK to the address in the top
Record-Route header. In this trace, that is a private IP. Your NAT
needs to re-write that to the public IP of your openser box as it hits
the internet. If the private IP makes it out onto the internet, you
will find your polycom is sending the ACK to 10.3.1.33. When the
asterisk box doesn't receive the ACK, the call will drop.
Thanks,
Phil
On Dec 13, 2007 6:00 PM, Patrick Baker <[EMAIL PROTECTED]> wrote:
> 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:[EMAIL PROTECTED]:5060;user=phone F=sip:[EMAIL PROTECTED]
> T=sip:[EMAIL PROTECTED];user=phone IP=xxx.206.xxx.137 [EMAIL PROTECTED]
> Dec 13 22:15:03 phonesys-slave openser[15976]: Callee is not local - M=INVITE
> RURI=sip:[EMAIL PROTECTED]:5060;user=phone F=sip:[EMAIL PROTECTED]
> T=sip:[EMAIL PROTECTED];user=phone IP=xxx.206.xxx.137 [EMAIL PROTECTED]
> Dec 13 22:15:03 phonesys-slave openser[15976]: Request leaving server,
> D-URI='<null>' - M=INVITE RURI=sip:[EMAIL PROTECTED]:5060;transport=udp
> F=sip:[EMAIL PROTECTED] T=sip:[EMAIL PROTECTED];user=phone IP=xxx.206.xxx.137
> [EMAIL PROTECTED]
> Dec 13 22:15:03 phonesys-slave openser[15973]: Reply - S=100 D=Trying
> F=sip:[EMAIL PROTECTED] T=sip:[EMAIL PROTECTED];user=phone IP=10.3.1.34
> [EMAIL PROTECTED]
> Dec 13 22:15:03 phonesys-slave openser[15978]: Reply - S=200 D=OK
> F=sip:[EMAIL PROTECTED] T=sip:[EMAIL PROTECTED];user=phone IP=10.3.1.34
> [EMAIL PROTECTED]
> Dec 13 22:15:04 phonesys-slave openser[15989]: Reply - S=200 D=OK
> F=sip:[EMAIL PROTECTED] T=sip:[EMAIL PROTECTED];user=phone IP=10.3.1.34
> [EMAIL PROTECTED]
> Dec 13 22:15:05 phonesys-slave openser[15986]: Reply - S=200 D=OK
> F=sip:[EMAIL PROTECTED] T=sip:[EMAIL PROTECTED];user=phone IP=10.3.1.34
> [EMAIL PROTECTED]
> Dec 13 22:15:07 phonesys-slave openser[15976]: Reply - S=200 D=OK
> F=sip:[EMAIL PROTECTED] T=sip:[EMAIL PROTECTED];user=phone IP=10.3.1.34
> [EMAIL PROTECTED]
>
> #### Asterisk LOG
>
> *CLI> [Dec 13 22:15:03]
> <--- SIP read from 10.3.1.33:5060 --->
> INVITE sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]>;tag=CFE2AAA1-5B882804
> To: <sip:[EMAIL PROTECTED];user=phone>
> CSeq: 2 INVITE
> Call-ID: [EMAIL PROTECTED]
> Contact: <sip:[EMAIL PROTECTED]>
> 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 - [EMAIL PROTECTED]
> [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:[EMAIL PROTECTED]>;tag=CFE2AAA1-5B882804
> To: <sip:[EMAIL PROTECTED];user=phone>
> Call-ID: [EMAIL PROTECTED]
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX SVN-trunk-r92779
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:[EMAIL PROTECTED]>
> Content-Length: 0
>
>
> <------------>
> [Dec 13 22:15:03] -- Executing [EMAIL PROTECTED]: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:[EMAIL PROTECTED]>;tag=CFE2AAA1-5B882804
> To: <sip:[EMAIL PROTECTED];user=phone>;tag=as26a99907
> Call-ID: [EMAIL PROTECTED]
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX SVN-trunk-r92779
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:[EMAIL PROTECTED]>
> 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:[EMAIL PROTECTED]>;tag=CFE2AAA1-5B882804
> To: <sip:[EMAIL PROTECTED];user=phone>;tag=as26a99907
> Call-ID: [EMAIL PROTECTED]
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX SVN-trunk-r92779
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:[EMAIL PROTECTED]>
> 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:[EMAIL PROTECTED]>;tag=CFE2AAA1-5B882804
> To: <sip:[EMAIL PROTECTED];user=phone>;tag=as26a99907
> Call-ID: [EMAIL PROTECTED]
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX SVN-trunk-r92779
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:[EMAIL PROTECTED]>
> 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:[EMAIL PROTECTED]>;tag=CFE2AAA1-5B882804
> To: <sip:[EMAIL PROTECTED];user=phone>;tag=as26a99907
> Call-ID: [EMAIL PROTECTED]
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX SVN-trunk-r92779
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:[EMAIL PROTECTED]>
> 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:[EMAIL PROTECTED]>;tag=CFE2AAA1-5B882804
> To: <sip:[EMAIL PROTECTED];user=phone>;tag=as26a99907
> Call-ID: [EMAIL PROTECTED]
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX SVN-trunk-r92779
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:[EMAIL PROTECTED]>
> 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:[EMAIL PROTECTED]>;tag=CFE2AAA1-5B882804
> To: <sip:[EMAIL PROTECTED];user=phone>;tag=as26a99907
> Call-ID: [EMAIL PROTECTED]
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX SVN-trunk-r92779
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:[EMAIL PROTECTED]>
> 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 [EMAIL PROTECTED]:2]
> Dial("SIP/xxx.206.xxx.136-090b5380", "IAX2/[EMAIL PROTECTED]/[EMAIL
> PROTECTED]") in new stack
> [Dec 13 22:15:18] -- Called [EMAIL PROTECTED]/[EMAIL PROTECTED]
> [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:[EMAIL PROTECTED]>;tag=CFE2AAA1-5B882804
> To: <sip:[EMAIL PROTECTED];user=phone>;tag=as26a99907
> Call-ID: [EMAIL PROTECTED]
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX SVN-trunk-r92779
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:[EMAIL PROTECTED]>
> 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 [EMAIL PROTECTED] for seqno 2 (Critical
> Response)
> [Dec 13 22:15:23] WARNING[16421]: chan_sip.c:2364 retrans_pkt: Hanging up
> call [EMAIL PROTECTED] - 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 '[EMAIL PROTECTED]' Method:
> INVITE
>
> #### PCAP Diagram
>
> |Time | xxx.206.xxx.137 | 10.3.1.33 |
> |5.056 | Request: INVITE sip |SIP/SDP: Request: INVITE
> sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]:5060
> | |(5060) ------------------> (5060) |
> |5.165 | Request: INVITE sip |SIP/SDP: Request: INVITE
> sip:[EMAIL PROTECTED]: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 ([EMAIL PROTECTED])
> ########################################################################
>
> ########################################################################
> # 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:[EMAIL PROTECTED]/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:[EMAIL PROTECTED]/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:[EMAIL PROTECTED]/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:[EMAIL PROTECTED]/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:[EMAIL PROTECTED]/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);
> }
>
> _______________________________________________
> Users mailing list
> [email protected]
> http://lists.openser.org/cgi-bin/mailman/listinfo/users
>
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