For some reason when trunking with Asterisk PBX's (yes, I know) FS wants to send the ACK to the internal ip found in the Contact field of the 200 OK. We have the force rport setting on but it's still not responding to that IP. Register's work. Most of the sip signalling works, just when the customer specifies the Contact filed with an internal ip. Below is a packet capture and our external.xml conf file. U 2008/11/04 09:17:08.259672 64.74.188.23:5060 -> 68.188.189.202:5060 INVITE sip:989607XXXX@68.188.189.202:5060 SIP/2.0. Via: SIP/2.0/UDP 64.74.188.23;rport;branch=z9hG4bKc6y6pr80HyeaN. Max-Forwards: 68. From: "user" <sip:303452XXXX@64.74.188.23>;tag=Dt6v81cDZXa3B. To: <sip:989607XXXX@68.188.189.202:5060>. Call-ID: de7c471c-252e-122c-3cba-5f1bac93b621. CSeq: 106789378 INVITE. Contact: <sip:mod_sofia@64.74.188.23:5060>. User-Agent: FreeSWITCH-mod_sofia/1.0.1-exported. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. Supported: 100rel, timer, precondition, path, replaces. Allow-Events: talk, presence, dialog, call-info, sla, include-session- description, presence.winfo, message-summary. Min-SE: 120. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 370. U 2008/11/04 09:21:32.861720 68.188.189.202:5060 -> 64.74.188.23:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 64.74.188.23 ;branch=z9hG4bKZFem0FmX4g19H;received=64.74.188.23;rport=5060. From: "user" <sip:303452XXXX@64.74.188.23>;tag=4UF788r8ct8aD. To: <sip:989607XXXX@68.188.189.202:5060>. Call-ID: 7c2d8578-252f-122c-3cba-5f1bac93b621. CSeq: 106789510 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Supported: replaces. Contact: <sip:989607XXXX@192.168.0.5>. Content-Length: 0. . U 2008/11/04 09:21:32.861845 68.188.189.202:5060 -> 64.74.188.23:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 64.74.188.23 ;branch=z9hG4bKZFem0FmX4g19H;received=64.74.188.23;rport=5060. From: "user" <sip:303452XXXX@64.74.188.23>;tag=4UF788r8ct8aD. To: <sip:989607XXXX@68.188.189.202:5060>;tag=as1da4b7aa. Call-ID: 7c2d8578-252f-122c-3cba-5f1bac93b621. CSeq: 106789510 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Supported: replaces. Contact: <sip:989607XXXX@192.168.0.5>. Content-Type: application/sdp. Content-Length: 285. . v=0. o=root 10970 10970 IN IP4 192.168.0.5. s=session. c=IN IP4 192.168.0.5. t=0 0. m=audio 15876 RTP/AVP 18 0 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. U 2008/11/04 09:21:32.862573 64.74.188.23:5060 -> 192.168.0.5:5060 ACK sip:989607XXXX@192.168.0.5 SIP/2.0. Via: SIP/2.0/UDP 64.74.188.23;rport;branch=z9hG4bK0r7c2a501SQvD. Max-Forwards: 70. From: "user" <sip:303452XXXX@64.74.188.23>;tag=4UF788r8ct8aD. To: <sip:989607XXXX@68.188.189.202:5060>;tag=as1da4b7aa. Call-ID: 7c2d8578-252f-122c-3cba-5f1bac93b621. CSeq: 106789510 ACK. Contact: <sip:mod_sofia@64.74.188.23:5060>. Content-Length: 0. . external.xml <settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="multiple-registrations" value="true"/> <param name="manage-presence" value="true"/> <param name="aggressive-nat-detection" value="true"/> <param name="NDLB-force-rport" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="true"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> </settings>
You need to set localnet and externip or externhost on Asterisk so it doesn't lie about its IP. /b On Nov 4, 2008, at 11:10 AM, David Aldworth wrote:
For some reason when trunking with Asterisk PBX's (yes, I know) FS wants to send the ACK to the internal ip found in the Contact field of the 200 OK. We have the force rport setting on but it's still not responding to that IP. Register's work. Most of the sip signalling works, just when the customer specifies the Contact filed with an internal ip. Below is a packet capture and our external.xml conf file.
U 2008/11/04 09:17:08.259672 64.74.188.23:5060 -> 68.188.189.202:5060 INVITE sip:989607XXXX@68.188.189.202:5060 SIP/2.0. Via: SIP/2.0/UDP 64.74.188.23;rport;branch=z9hG4bKc6y6pr80HyeaN. Max-Forwards: 68. From: "user" <sip:303452XXXX@64.74.188.23>;tag=Dt6v81cDZXa3B. To: <sip:989607XXXX@68.188.189.202:5060>. Call-ID: de7c471c-252e-122c-3cba-5f1bac93b621. CSeq: 106789378 INVITE. Contact: <sip:mod_sofia@64.74.188.23:5060>. User-Agent: FreeSWITCH-mod_sofia/1.0.1-exported. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. Supported: 100rel, timer, precondition, path, replaces. Allow-Events: talk, presence, dialog, call-info, sla, include-session- description, presence.winfo, message-summary. Min-SE: 120. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 370.
U 2008/11/04 09:21:32.861720 68.188.189.202:5060 -> 64.74.188.23:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 64.74.188.23 ;branch=z9hG4bKZFem0FmX4g19H;received=64.74.188.23;rport=5060. From: "user" <sip:303452XXXX@64.74.188.23>;tag=4UF788r8ct8aD. To: <sip:989607XXXX@68.188.189.202:5060>. Call-ID: 7c2d8578-252f-122c-3cba-5f1bac93b621. CSeq: 106789510 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Supported: replaces. Contact: <sip:989607XXXX@192.168.0.5>. Content-Length: 0. .
U 2008/11/04 09:21:32.861845 68.188.189.202:5060 -> 64.74.188.23:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 64.74.188.23 ;branch=z9hG4bKZFem0FmX4g19H;received=64.74.188.23;rport=5060. From: "user" <sip:303452XXXX@64.74.188.23>;tag=4UF788r8ct8aD. To: <sip:989607XXXX@68.188.189.202:5060>;tag=as1da4b7aa. Call-ID: 7c2d8578-252f-122c-3cba-5f1bac93b621. CSeq: 106789510 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Supported: replaces. Contact: <sip:989607XXXX@192.168.0.5>. Content-Type: application/sdp. Content-Length: 285. . v=0. o=root 10970 10970 IN IP4 192.168.0.5. s=session. c=IN IP4 192.168.0.5. t=0 0. m=audio 15876 RTP/AVP 18 0 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv.
U 2008/11/04 09:21:32.862573 64.74.188.23:5060 -> 192.168.0.5:5060 ACK sip:989607XXXX@192.168.0.5 SIP/2.0. Via: SIP/2.0/UDP 64.74.188.23;rport;branch=z9hG4bK0r7c2a501SQvD. Max-Forwards: 70. From: "user" <sip:303452XXXX@64.74.188.23>;tag=4UF788r8ct8aD. To: <sip:989607XXXX@68.188.189.202:5060>;tag=as1da4b7aa. Call-ID: 7c2d8578-252f-122c-3cba-5f1bac93b621. CSeq: 106789510 ACK. Contact: <sip:mod_sofia@64.74.188.23:5060>. Content-Length: 0. .
external.xml
<settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="multiple-registrations" value="true"/> <param name="manage-presence" value="true"/> <param name="aggressive-nat-detection" value="true"/> <param name="NDLB-force-rport" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="true"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> </settings>
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Hi bkw - We did that and it does indeed fix the issue. However, in the case that you have multiple SIP UA's behind a router there tend to be many dynamically generated ports in use. The obvious solution would be to statically map a port to an internal IP and then set the externip and localhost settings. I agree, this would work. Except if you are using a dsl or cable modem provider that also like to update your WAN ip on a regular basis. But what confuses me more is that all the sip messaging works fine right up until the ACK we send back to the 200 OK. Obviously FS is sending the ACK to the Contact field IP but is there a way in FS to tell it to just respond on the IP and port that the 200 OK came from? I thought that is what the force rport setting did but i guess it does not. David On Nov 4, 2008, at 10:32 AM, Brian West wrote:
You need to set localnet and externip or externhost on Asterisk so it doesn't lie about its IP.
/b
On Nov 4, 2008, at 11:10 AM, David Aldworth wrote:
For some reason when trunking with Asterisk PBX's (yes, I know) FS wants to send the ACK to the internal ip found in the Contact field of the 200 OK. We have the force rport setting on but it's still not responding to that IP. Register's work. Most of the sip signalling works, just when the customer specifies the Contact filed with an internal ip. Below is a packet capture and our external.xml conf file.
U 2008/11/04 09:17:08.259672 64.74.188.23:5060 -> 68.188.189.202:5060 INVITE sip:989607XXXX@68.188.189.202:5060 SIP/2.0. Via: SIP/2.0/UDP 64.74.188.23;rport;branch=z9hG4bKc6y6pr80HyeaN. Max-Forwards: 68. From: "user" <sip:303452XXXX@64.74.188.23>;tag=Dt6v81cDZXa3B. To: <sip:989607XXXX@68.188.189.202:5060>. Call-ID: de7c471c-252e-122c-3cba-5f1bac93b621. CSeq: 106789378 INVITE. Contact: <sip:mod_sofia@64.74.188.23:5060>. User-Agent: FreeSWITCH-mod_sofia/1.0.1-exported. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. Supported: 100rel, timer, precondition, path, replaces. Allow-Events: talk, presence, dialog, call-info, sla, include- session- description, presence.winfo, message-summary. Min-SE: 120. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 370.
U 2008/11/04 09:21:32.861720 68.188.189.202:5060 -> 64.74.188.23:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 64.74.188.23 ;branch=z9hG4bKZFem0FmX4g19H;received=64.74.188.23;rport=5060. From: "user" <sip:303452XXXX@64.74.188.23>;tag=4UF788r8ct8aD. To: <sip:989607XXXX@68.188.189.202:5060>. Call-ID: 7c2d8578-252f-122c-3cba-5f1bac93b621. CSeq: 106789510 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Supported: replaces. Contact: <sip:989607XXXX@192.168.0.5>. Content-Length: 0. .
U 2008/11/04 09:21:32.861845 68.188.189.202:5060 -> 64.74.188.23:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 64.74.188.23 ;branch=z9hG4bKZFem0FmX4g19H;received=64.74.188.23;rport=5060. From: "user" <sip:303452XXXX@64.74.188.23>;tag=4UF788r8ct8aD. To: <sip:989607XXXX@68.188.189.202:5060>;tag=as1da4b7aa. Call-ID: 7c2d8578-252f-122c-3cba-5f1bac93b621. CSeq: 106789510 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Supported: replaces. Contact: <sip:989607XXXX@192.168.0.5>. Content-Type: application/sdp. Content-Length: 285. . v=0. o=root 10970 10970 IN IP4 192.168.0.5. s=session. c=IN IP4 192.168.0.5. t=0 0. m=audio 15876 RTP/AVP 18 0 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv.
U 2008/11/04 09:21:32.862573 64.74.188.23:5060 -> 192.168.0.5:5060 ACK sip:989607XXXX@192.168.0.5 SIP/2.0. Via: SIP/2.0/UDP 64.74.188.23;rport;branch=z9hG4bK0r7c2a501SQvD. Max-Forwards: 70. From: "user" <sip:303452XXXX@64.74.188.23>;tag=4UF788r8ct8aD. To: <sip:989607XXXX@68.188.189.202:5060>;tag=as1da4b7aa. Call-ID: 7c2d8578-252f-122c-3cba-5f1bac93b621. CSeq: 106789510 ACK. Contact: <sip:mod_sofia@64.74.188.23:5060>. Content-Length: 0. .
external.xml
<settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="multiple-registrations" value="true"/> <param name="manage-presence" value="true"/> <param name="aggressive-nat-detection" value="true"/> <param name="NDLB-force-rport" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="true"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> </settings>
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You can use the param "NDLB-force-rport" to force it to use rport no matter what. /b On Nov 4, 2008, at 11:43 AM, David Aldworth wrote:
Hi bkw -
We did that and it does indeed fix the issue. However, in the case that you have multiple SIP UA's behind a router there tend to be many dynamically generated ports in use. The obvious solution would be to statically map a port to an internal IP and then set the externip and localhost settings. I agree, this would work. Except if you are using a dsl or cable modem provider that also like to update your WAN ip on a regular basis. But what confuses me more is that all the sip messaging works fine right up until the ACK we send back to the 200 OK. Obviously FS is sending the ACK to the Contact field IP but is there a way in FS to tell it to just respond on the IP and port that the 200 OK came from? I thought that is what the force rport setting did but i guess it does not.
David
That is actually already on. Any idea? On Nov 4, 2008, at 10:45 AM, Brian West wrote:
You can use the param "NDLB-force-rport" to force it to use rport no matter what.
/b
On Nov 4, 2008, at 11:43 AM, David Aldworth wrote:
Hi bkw -
We did that and it does indeed fix the issue. However, in the case that you have multiple SIP UA's behind a router there tend to be many dynamically generated ports in use. The obvious solution would be to statically map a port to an internal IP and then set the externip and localhost settings. I agree, this would work. Except if you are using a dsl or cable modem provider that also like to update your WAN ip on a regular basis. But what confuses me more is that all the sip messaging works fine right up until the ACK we send back to the 200 OK. Obviously FS is sending the ACK to the Contact field IP but is there a way in FS to tell it to just respond on the IP and port that the 200 OK came from? I thought that is what the force rport setting did but i guess it does not.
David
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It was in there when the server got brought up. Here is the full .xml settings: <settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="multiple-registrations" value="true"/> <param name="manage-presence" value="true"/> <param name="aggressive-nat-detection" value="true"/> <param name="NDLB-force-rport" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="true"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> </settings> On Nov 4, 2008, at 10:52 AM, Brian West wrote:
Make sure you restart the profile for it to take effect.
/b
On Nov 4, 2008, at 11:49 AM, David Aldworth wrote:
That is actually already on.
Any idea?
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keep in mind that the spec says when a contact addr has a new ip that you are supposed to change the ip for all the rest of the messages to that new location. So we are doing what we are supposed to. If you want to force the behavior you can use the nat hack to lock onto the first ip/port the dialog used. make an acl called ast in acl.conf.xml with the asterisk box ip in it add apply_nat_acl=ast to the sip profile params. Then when anything from that acl matches FS will use the nat hacks to lock on the address it sends packets to. On Tue, Nov 4, 2008 at 11:55 AM, David Aldworth <daldworth@teliax.com>wrote:
It was in there when the server got brought up.
Here is the full .xml settings:
<settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="multiple-registrations" value="true"/> <param name="manage-presence" value="true"/> <param name="aggressive-nat-detection" value="true"/> <param name="NDLB-force-rport" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="true"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> </settings>
On Nov 4, 2008, at 10:52 AM, Brian West wrote:
Make sure you restart the profile for it to take effect.
/b
On Nov 4, 2008, at 11:49 AM, David Aldworth wrote:
That is actually already on.
Any idea?
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Anthony, In hopes of matching all IP's we added a very simple: <list name="nat" default="allow"> </list> To the acl.conf.xml and we added: <param name="apply_nat_acl" value="nat"/> To the sip profile. Unfortunately there was no affect. What would be the correct acl to match all IP's? David On Nov 4, 2008, at 1:18 PM, Anthony Minessale wrote:
keep in mind that the spec says when a contact addr has a new ip that you are supposed to change the ip for all the rest of the messages to that new location. So we are doing what we are supposed to.
If you want to force the behavior you can use the nat hack to lock onto the first ip/port the dialog used.
make an acl called ast in acl.conf.xml with the asterisk box ip in it
add apply_nat_acl=ast to the sip profile params.
Then when anything from that acl matches FS will use the nat hacks to lock on the address it sends packets to.
On Tue, Nov 4, 2008 at 11:55 AM, David Aldworth <daldworth@teliax.com> wrote: It was in there when the server got brought up.
Here is the full .xml settings:
<settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="multiple-registrations" value="true"/> <param name="manage-presence" value="true"/> <param name="aggressive-nat-detection" value="true"/> <param name="NDLB-force-rport" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="true"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> </settings>
On Nov 4, 2008, at 10:52 AM, Brian West wrote:
Make sure you restart the profile for it to take effect.
/b
On Nov 4, 2008, at 11:49 AM, David Aldworth wrote:
That is actually already on.
Any idea?
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
0.0.0.0/0 should match all IP space. /b On Nov 5, 2008, at 5:16 PM, David Aldworth wrote:
Anthony, In hopes of matching all IP's we added a very simple:
<list name="nat" default="allow"> </list>
To the acl.conf.xml and we added:
<param name="apply_nat_acl" value="nat"/>
To the sip profile. Unfortunately there was no affect. What would be the correct acl to match all IP's?
David
Brian, we updated the acl to: <list name="nat" default="allow"> <node type="allow" cidr="0.0.0.0/0"/> </list> And the ACK is still going to the wrong (right but wrong) ip/port. Is there any way to get that ACK to go to the ip/port of the UDP header? David On Nov 5, 2008, at 4:21 PM, Brian West wrote:
0.0.0.0/0 should match all IP space.
/b
On Nov 5, 2008, at 5:16 PM, David Aldworth wrote:
Anthony, In hopes of matching all IP's we added a very simple:
<list name="nat" default="allow"> </list>
To the acl.conf.xml and we added:
<param name="apply_nat_acl" value="nat"/>
To the sip profile. Unfortunately there was no affect. What would be the correct acl to match all IP's?
David
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Please open a bug on http://jira.freeswitch.org . Please include a full debug output from the freeswitch console with TPORT_LOG enabled (info on the sofia page on the wiki). Mike On Nov 5, 2008, at 11:39 PM, David Aldworth wrote:
Brian, we updated the acl to:
<list name="nat" default="allow"> <node type="allow" cidr="0.0.0.0/0"/> </list>
And the ACK is still going to the wrong (right but wrong) ip/port.
Is there any way to get that ACK to go to the ip/port of the UDP header?
David
On Nov 5, 2008, at 4:21 PM, Brian West wrote:
0.0.0.0/0 should match all IP space.
/b
On Nov 5, 2008, at 5:16 PM, David Aldworth wrote:
Anthony, In hopes of matching all IP's we added a very simple:
<list name="nat" default="allow"> </list>
To the acl.conf.xml and we added:
<param name="apply_nat_acl" value="nat"/>
To the sip profile. Unfortunately there was no affect. What would be the correct acl to match all IP's?
David
did you remember to add <param name="apply_nat_acl" value="nat"/> to the profile in question and restart? On Wed, Nov 5, 2008 at 10:39 PM, David Aldworth <daldworth@teliax.com>wrote:
Brian, we updated the acl to:
<list name="nat" default="allow"> <node type="allow" cidr="0.0.0.0/0"/> </list>
And the ACK is still going to the wrong (right but wrong) ip/port.
Is there any way to get that ACK to go to the ip/port of the UDP header?
David
On Nov 5, 2008, at 4:21 PM, Brian West wrote:
0.0.0.0/0 should match all IP space.
/b
On Nov 5, 2008, at 5:16 PM, David Aldworth wrote:
Anthony, In hopes of matching all IP's we added a very simple:
<list name="nat" default="allow"> </list>
To the acl.conf.xml and we added:
<param name="apply_nat_acl" value="nat"/>
To the sip profile. Unfortunately there was no affect. What would be the correct acl to match all IP's?
David
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Yes. Below are settings that have been persistent through recent testing. Is there anything else we can try or should we open a jira? <settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="multiple-registrations" value="true"/> <param name="manage-presence" value="true"/> <param name="aggressive-nat-detection" value="true"/> <param name="NDLB-force-rport" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="true"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="apply_nat_acl" value="nat"/> </settings> On Nov 6, 2008, at 7:01 AM, Anthony Minessale wrote:
did you remember to add <param name="apply_nat_acl" value="nat"/> to the profile in question and restart?
On Wed, Nov 5, 2008 at 10:39 PM, David Aldworth <daldworth@teliax.com> wrote: Brian, we updated the acl to:
<list name="nat" default="allow"> <node type="allow" cidr="0.0.0.0/0"/> </list>
And the ACK is still going to the wrong (right but wrong) ip/port.
Is there any way to get that ACK to go to the ip/port of the UDP header?
David
On Nov 5, 2008, at 4:21 PM, Brian West wrote:
0.0.0.0/0 should match all IP space.
/b
On Nov 5, 2008, at 5:16 PM, David Aldworth wrote:
Anthony, In hopes of matching all IP's we added a very simple:
<list name="nat" default="allow"> </list>
To the acl.conf.xml and we added:
<param name="apply_nat_acl" value="nat"/>
To the sip profile. Unfortunately there was no affect. What would be the correct acl to match all IP's?
David
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
doh, I keep doing that sorry. apply-nat-acl not apply_nat_acl On Thu, Nov 6, 2008 at 8:22 AM, David Aldworth <daldworth@teliax.com> wrote:
Yes. Below are settings that have been persistent through recent testing. Is there anything else we can try or should we open a jira? <settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="multiple-registrations" value="true"/> <param name="manage-presence" value="true"/> <param name="aggressive-nat-detection" value="true"/> <param name="NDLB-force-rport" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="true"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="apply_nat_acl" value="nat"/> </settings>
On Nov 6, 2008, at 7:01 AM, Anthony Minessale wrote:
did you remember to add <param name="apply_nat_acl" value="nat"/> to the profile in question and restart?
On Wed, Nov 5, 2008 at 10:39 PM, David Aldworth <daldworth@teliax.com>wrote:
Brian, we updated the acl to:
<list name="nat" default="allow"> <node type="allow" cidr="0.0.0.0/0"/> </list>
And the ACK is still going to the wrong (right but wrong) ip/port.
Is there any way to get that ACK to go to the ip/port of the UDP header?
David
On Nov 5, 2008, at 4:21 PM, Brian West wrote:
0.0.0.0/0 should match all IP space.
/b
On Nov 5, 2008, at 5:16 PM, David Aldworth wrote:
Anthony, In hopes of matching all IP's we added a very simple:
<list name="nat" default="allow"> </list>
To the acl.conf.xml and we added:
<param name="apply_nat_acl" value="nat"/>
To the sip profile. Unfortunately there was no affect. What would be the correct acl to match all IP's?
David
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
No love. They set extern ip so the IP comes through correctly, but the acl did not seem to have any affect. We are still sending to the wrong port. Sip trace, acl.conf.xml and sip profile are below: U 2008/11/06 10:46:01.924795 70.88.65.1:50085 -> 70.42.223.23:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 70.42.223.23 ;branch=z9hG4bKU7360cS96r7Sg;received=70.42.223.23;rport=5060. From: "TELIAX FAX" <sip:303825XXXX@70.42.223.23>;tag=armgX7QeNQ94N. To: <sip:317376XXXX@70.88.65.1:50085>. Call-ID: 9e67419c-26cd-122c-0b81-e9d53e66cb70. CSeq: 106878444 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Contact: <sip:317376XXXX@70.88.65.1>. Content-Length: 0. . U 2008/11/06 10:46:01.931791 70.88.65.1:50085 -> 70.42.223.23:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 70.42.223.23 ;branch=z9hG4bKU7360cS96r7Sg;received=70.42.223.23;rport=5060. From: "TELIAX FAX" <sip:303825XXXX@70.42.223.23>;tag=armgX7QeNQ94N. To: <sip:317376XXXX@70.88.65.1:50085>;tag=as78a21a0c. Call-ID: 9e67419c-26cd-122c-0b81-e9d53e66cb70. CSeq: 106878444 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Contact: <sip:317376XXXX@70.88.65.1>. Content-Length: 0. . U 2008/11/06 10:46:01.932294 70.88.65.1:50085 -> 70.42.223.23:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 70.42.223.23 ;branch=z9hG4bKU7360cS96r7Sg;received=70.42.223.23;rport=5060. From: "TELIAX FAX" <sip:303825XXXX@70.42.223.23>;tag=armgX7QeNQ94N. To: <sip:317376XXXX@70.88.65.1:50085>;tag=as78a21a0c. Call-ID: 9e67419c-26cd-122c-0b81-e9d53e66cb70. CSeq: 106878444 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Contact: <sip:317376XXXX@70.88.65.1>. Content-Type: application/sdp. Content-Length: 257. . v=0. o=root 2901 2901 IN IP4 70.88.65.1. s=session. c=IN IP4 70.88.65.1. t=0 0. m=audio 19378 RTP/AVP 0 8 3 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:3 GSM/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. U 2008/11/06 10:46:01.932694 70.42.223.23:5060 -> 70.88.65.1:5060 ACK sip:317376XXXX@70.88.65.1 SIP/2.0. Via: SIP/2.0/UDP 70.42.223.23;rport;branch=z9hG4bKvgXZ279c41Xcc. Max-Forwards: 70. From: "TELIAX FAX" <sip:303825XXXX@70.42.223.23>;tag=armgX7QeNQ94N. To: <sip:317376XXXX@70.88.65.1:50085>;tag=as78a21a0c. Call-ID: 9e67419c-26cd-122c-0b81-e9d53e66cb70. CSeq: 106878444 ACK. Contact: <sip:mod_sofia@70.42.223.23:5060>. Content-Length: 0. Here is the acl: <configuration name="acl.conf" description="Network Lists"> <network-lists> <list name="dl-candidates" default="allow"> <node type="deny" cidr="10.0.0.0/8"/> <node type="deny" cidr="172.16.0.0/12"/> <node type="deny" cidr="192.168.0.0/16"/> </list> <list name="rfc1918" default="deny"> <node type="allow" cidr="10.0.0.0/8"/> <node type="allow" cidr="172.16.0.0/12"/> <node type="allow" cidr="192.168.0.0/16"/> </list> <list name="lan" default="allow"> <node type="deny" cidr="192.168.42.0/24"/> <node type="allow" cidr="192.168.42.42/32"/> </list> <list name="strict" default="deny"> <node type="allow" cidr="208.102.123.124/32"/> </list> <list name="domains" default="deny"> <node type="allow" domain="$${domain}"/> </list> <list name="nat" default="allow"> <node type="allow" cidr="0.0.0.0/0"/> </list> </network-lists> </configuration> And here is the sip profile: <profile name="external"> <gateways> <X-PRE-PROCESS cmd="include" data="external/*.xml"/> </gateways> <domains> <domain name="$${domain}" parse="true"/> </domains> <settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="multiple-registrations" value="true"/> <param name="manage-presence" value="true"/> <param name="aggressive-nat-detection" value="true"/> <param name="NDLB-force-rport" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="true"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="apply-nat-acl" value="nat"/> </settings> </profile> On Nov 6, 2008, at 8:37 AM, Anthony Minessale wrote:
doh, I keep doing that sorry.
apply-nat-acl not apply_nat_acl
On Thu, Nov 6, 2008 at 8:22 AM, David Aldworth <daldworth@teliax.com> wrote: Yes. Below are settings that have been persistent through recent testing. Is there anything else we can try or should we open a jira?
<settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="multiple-registrations" value="true"/> <param name="manage-presence" value="true"/> <param name="aggressive-nat-detection" value="true"/> <param name="NDLB-force-rport" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="true"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="apply_nat_acl" value="nat"/> </settings>
On Nov 6, 2008, at 7:01 AM, Anthony Minessale wrote:
did you remember to add <param name="apply_nat_acl" value="nat"/> to the profile in question and restart?
On Wed, Nov 5, 2008 at 10:39 PM, David Aldworth <daldworth@teliax.com> wrote: Brian, we updated the acl to:
<list name="nat" default="allow"> <node type="allow" cidr="0.0.0.0/0"/> </list>
And the ACK is still going to the wrong (right but wrong) ip/port.
Is there any way to get that ACK to go to the ip/port of the UDP header?
David
On Nov 5, 2008, at 4:21 PM, Brian West wrote:
0.0.0.0/0 should match all IP space.
/b
On Nov 5, 2008, at 5:16 PM, David Aldworth wrote:
Anthony, In hopes of matching all IP's we added a very simple:
<list name="nat" default="allow"> </list>
To the acl.conf.xml and we added:
<param name="apply_nat_acl" value="nat"/>
To the sip profile. Unfortunately there was no affect. What would be the correct acl to match all IP's?
David
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-- Anthony Minessale II
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FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
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This is svn trunk? There is no reason this should not work. it happens all the time where this setting breaks it for people going the other way when they don't want it to happen. If you can't get it working we can probably configure it for you. On Thu, Nov 6, 2008 at 11:55 AM, David Aldworth <daldworth@teliax.com>wrote:
No love. They set extern ip so the IP comes through correctly, but the acl did not seem to have any affect. We are still sending to the wrong port. Sip trace, acl.conf.xml and sip profile are below: U 2008/11/06 10:46:01.924795 70.88.65.1:50085 -> 70.42.223.23:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 70.42.223.23;branch=z9hG4bKU7360cS96r7Sg;received= 70.42.223.23;rport=5060. From: "TELIAX FAX" <sip:303825XXXX@70.42.223.23<sip%3A303825XXXX@70.42.223.23>
;tag=armgX7QeNQ94N. To: <sip:317376XXXX@70.88.65.1:50085>. Call-ID: 9e67419c-26cd-122c-0b81-e9d53e66cb70. CSeq: 106878444 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Contact: <sip:317376XXXX@70.88.65.1 <sip%3A317376XXXX@70.88.65.1>>. Content-Length: 0. .
U 2008/11/06 10:46:01.931791 70.88.65.1:50085 -> 70.42.223.23:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 70.42.223.23;branch=z9hG4bKU7360cS96r7Sg;received= 70.42.223.23;rport=5060. From: "TELIAX FAX" <sip:303825XXXX@70.42.223.23<sip%3A303825XXXX@70.42.223.23>
;tag=armgX7QeNQ94N. To: <sip:317376XXXX@70.88.65.1:50085>;tag=as78a21a0c. Call-ID: 9e67419c-26cd-122c-0b81-e9d53e66cb70. CSeq: 106878444 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Contact: <sip:317376XXXX@70.88.65.1 <sip%3A317376XXXX@70.88.65.1>>. Content-Length: 0. .
U 2008/11/06 10:46:01.932294 70.88.65.1:50085 -> 70.42.223.23:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 70.42.223.23;branch=z9hG4bKU7360cS96r7Sg;received= 70.42.223.23;rport=5060. From: "TELIAX FAX" <sip:303825XXXX@70.42.223.23<sip%3A303825XXXX@70.42.223.23>
;tag=armgX7QeNQ94N. To: <sip:317376XXXX@70.88.65.1:50085>;tag=as78a21a0c. Call-ID: 9e67419c-26cd-122c-0b81-e9d53e66cb70. CSeq: 106878444 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Contact: <sip:317376XXXX@70.88.65.1 <sip%3A317376XXXX@70.88.65.1>>. Content-Type: application/sdp. Content-Length: 257. . v=0. o=root 2901 2901 IN IP4 70.88.65.1. s=session. c=IN IP4 70.88.65.1. t=0 0. m=audio 19378 RTP/AVP 0 8 3 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:3 GSM/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -.
U 2008/11/06 10:46:01.932694 70.42.223.23:5060 -> 70.88.65.1:5060 ACK sip:317376XXXX@70.88.65.1 <sip%3A317376XXXX@70.88.65.1> SIP/2.0. Via: SIP/2.0/UDP 70.42.223.23;rport;branch=z9hG4bKvgXZ279c41Xcc. Max-Forwards: 70. From: "TELIAX FAX" <sip:303825XXXX@70.42.223.23<sip%3A303825XXXX@70.42.223.23>
;tag=armgX7QeNQ94N. To: <sip:317376XXXX@70.88.65.1:50085>;tag=as78a21a0c. Call-ID: 9e67419c-26cd-122c-0b81-e9d53e66cb70. CSeq: 106878444 ACK. Contact: <sip:mod_sofia@70.42.223.23:5060>. Content-Length: 0.
Here is the acl:
<configuration name="acl.conf" description="Network Lists"> <network-lists> <list name="dl-candidates" default="allow"> <node type="deny" cidr="10.0.0.0/8"/> <node type="deny" cidr="172.16.0.0/12"/> <node type="deny" cidr="192.168.0.0/16"/> </list> <list name="rfc1918" default="deny"> <node type="allow" cidr="10.0.0.0/8"/> <node type="allow" cidr="172.16.0.0/12"/> <node type="allow" cidr="192.168.0.0/16"/> </list> <list name="lan" default="allow"> <node type="deny" cidr="192.168.42.0/24"/> <node type="allow" cidr="192.168.42.42/32"/> </list> <list name="strict" default="deny"> <node type="allow" cidr="208.102.123.124/32"/> </list> <list name="domains" default="deny"> <node type="allow" domain="$${domain}"/> </list> <list name="nat" default="allow"> <node type="allow" cidr="0.0.0.0/0"/> </list> </network-lists> </configuration>
And here is the sip profile:
<profile name="external">
<gateways> <X-PRE-PROCESS cmd="include" data="external/*.xml"/> </gateways>
<domains> <domain name="$${domain}" parse="true"/> </domains>
<settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="multiple-registrations" value="true"/> <param name="manage-presence" value="true"/> <param name="aggressive-nat-detection" value="true"/> <param name="NDLB-force-rport" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="true"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="apply-nat-acl" value="nat"/> </settings> </profile>
On Nov 6, 2008, at 8:37 AM, Anthony Minessale wrote:
doh, I keep doing that sorry.
apply-nat-acl not apply_nat_acl
On Thu, Nov 6, 2008 at 8:22 AM, David Aldworth <daldworth@teliax.com>wrote:
Yes. Below are settings that have been persistent through recent testing. Is there anything else we can try or should we open a jira? <settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="multiple-registrations" value="true"/> <param name="manage-presence" value="true"/> <param name="aggressive-nat-detection" value="true"/> <param name="NDLB-force-rport" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="true"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="apply_nat_acl" value="nat"/> </settings>
On Nov 6, 2008, at 7:01 AM, Anthony Minessale wrote:
did you remember to add <param name="apply_nat_acl" value="nat"/> to the profile in question and restart?
On Wed, Nov 5, 2008 at 10:39 PM, David Aldworth <daldworth@teliax.com>wrote:
Brian, we updated the acl to:
<list name="nat" default="allow"> <node type="allow" cidr="0.0.0.0/0"/> </list>
And the ACK is still going to the wrong (right but wrong) ip/port.
Is there any way to get that ACK to go to the ip/port of the UDP header?
David
On Nov 5, 2008, at 4:21 PM, Brian West wrote:
0.0.0.0/0 should match all IP space.
/b
On Nov 5, 2008, at 5:16 PM, David Aldworth wrote:
Anthony, In hopes of matching all IP's we added a very simple:
<list name="nat" default="allow"> </list>
To the acl.conf.xml and we added:
<param name="apply_nat_acl" value="nat"/>
To the sip profile. Unfortunately there was no affect. What would be the correct acl to match all IP's?
David
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Anthony. Did you want to log in and check it out? I can send you the files if you think it's something else. David On Nov 6, 2008, at 12:38 PM, Anthony Minessale wrote:
This is svn trunk? There is no reason this should not work. it happens all the time where this setting breaks it for people going the other way when they don't want it to happen.
If you can't get it working we can probably configure it for you.
On Thu, Nov 6, 2008 at 11:55 AM, David Aldworth <daldworth@teliax.com> wrote: No love. They set extern ip so the IP comes through correctly, but the acl did not seem to have any affect. We are still sending to the wrong port. Sip trace, acl.conf.xml and sip profile are below:
U 2008/11/06 10:46:01.924795 70.88.65.1:50085 -> 70.42.223.23:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 70.42.223.23 ;branch=z9hG4bKU7360cS96r7Sg;received=70.42.223.23;rport=5060. From: "TELIAX FAX" <sip:303825XXXX@70.42.223.23>;tag=armgX7QeNQ94N. To: <sip:317376XXXX@70.88.65.1:50085>. Call-ID: 9e67419c-26cd-122c-0b81-e9d53e66cb70. CSeq: 106878444 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Contact: <sip:317376XXXX@70.88.65.1>. Content-Length: 0. .
U 2008/11/06 10:46:01.931791 70.88.65.1:50085 -> 70.42.223.23:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 70.42.223.23 ;branch=z9hG4bKU7360cS96r7Sg;received=70.42.223.23;rport=5060. From: "TELIAX FAX" <sip:303825XXXX@70.42.223.23>;tag=armgX7QeNQ94N. To: <sip:317376XXXX@70.88.65.1:50085>;tag=as78a21a0c. Call-ID: 9e67419c-26cd-122c-0b81-e9d53e66cb70. CSeq: 106878444 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Contact: <sip:317376XXXX@70.88.65.1>. Content-Length: 0. .
U 2008/11/06 10:46:01.932294 70.88.65.1:50085 -> 70.42.223.23:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 70.42.223.23 ;branch=z9hG4bKU7360cS96r7Sg;received=70.42.223.23;rport=5060. From: "TELIAX FAX" <sip:303825XXXX@70.42.223.23>;tag=armgX7QeNQ94N. To: <sip:317376XXXX@70.88.65.1:50085>;tag=as78a21a0c. Call-ID: 9e67419c-26cd-122c-0b81-e9d53e66cb70. CSeq: 106878444 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Contact: <sip:317376XXXX@70.88.65.1>. Content-Type: application/sdp. Content-Length: 257. . v=0. o=root 2901 2901 IN IP4 70.88.65.1. s=session. c=IN IP4 70.88.65.1. t=0 0. m=audio 19378 RTP/AVP 0 8 3 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:3 GSM/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -.
U 2008/11/06 10:46:01.932694 70.42.223.23:5060 -> 70.88.65.1:5060 ACK sip:317376XXXX@70.88.65.1 SIP/2.0. Via: SIP/2.0/UDP 70.42.223.23;rport;branch=z9hG4bKvgXZ279c41Xcc. Max-Forwards: 70. From: "TELIAX FAX" <sip:303825XXXX@70.42.223.23>;tag=armgX7QeNQ94N. To: <sip:317376XXXX@70.88.65.1:50085>;tag=as78a21a0c. Call-ID: 9e67419c-26cd-122c-0b81-e9d53e66cb70. CSeq: 106878444 ACK. Contact: <sip:mod_sofia@70.42.223.23:5060>. Content-Length: 0.
Here is the acl:
<configuration name="acl.conf" description="Network Lists"> <network-lists> <list name="dl-candidates" default="allow"> <node type="deny" cidr="10.0.0.0/8"/> <node type="deny" cidr="172.16.0.0/12"/> <node type="deny" cidr="192.168.0.0/16"/> </list> <list name="rfc1918" default="deny"> <node type="allow" cidr="10.0.0.0/8"/> <node type="allow" cidr="172.16.0.0/12"/> <node type="allow" cidr="192.168.0.0/16"/> </list> <list name="lan" default="allow"> <node type="deny" cidr="192.168.42.0/24"/> <node type="allow" cidr="192.168.42.42/32"/> </list> <list name="strict" default="deny"> <node type="allow" cidr="208.102.123.124/32"/> </list> <list name="domains" default="deny"> <node type="allow" domain="$${domain}"/> </list> <list name="nat" default="allow"> <node type="allow" cidr="0.0.0.0/0"/> </list> </network-lists> </configuration>
And here is the sip profile:
<profile name="external">
<gateways> <X-PRE-PROCESS cmd="include" data="external/*.xml"/> </gateways>
<domains> <domain name="$${domain}" parse="true"/> </domains>
<settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="multiple-registrations" value="true"/> <param name="manage-presence" value="true"/> <param name="aggressive-nat-detection" value="true"/> <param name="NDLB-force-rport" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="true"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="apply-nat-acl" value="nat"/> </settings> </profile>
On Nov 6, 2008, at 8:37 AM, Anthony Minessale wrote:
doh, I keep doing that sorry.
apply-nat-acl not apply_nat_acl
On Thu, Nov 6, 2008 at 8:22 AM, David Aldworth <daldworth@teliax.com> wrote: Yes. Below are settings that have been persistent through recent testing. Is there anything else we can try or should we open a jira?
<settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="multiple-registrations" value="true"/> <param name="manage-presence" value="true"/> <param name="aggressive-nat-detection" value="true"/> <param name="NDLB-force-rport" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="true"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="apply_nat_acl" value="nat"/> </settings>
On Nov 6, 2008, at 7:01 AM, Anthony Minessale wrote:
did you remember to add <param name="apply_nat_acl" value="nat"/> to the profile in question and restart?
On Wed, Nov 5, 2008 at 10:39 PM, David Aldworth <daldworth@teliax.com
wrote: Brian, we updated the acl to:
<list name="nat" default="allow"> <node type="allow" cidr="0.0.0.0/0"/> </list>
And the ACK is still going to the wrong (right but wrong) ip/port.
Is there any way to get that ACK to go to the ip/port of the UDP header?
David
On Nov 5, 2008, at 4:21 PM, Brian West wrote:
0.0.0.0/0 should match all IP space.
/b
On Nov 5, 2008, at 5:16 PM, David Aldworth wrote:
Anthony, In hopes of matching all IP's we added a very simple:
<list name="nat" default="allow"> </list>
To the acl.conf.xml and we added:
<param name="apply_nat_acl" value="nat"/>
To the sip profile. Unfortunately there was no affect. What would be the correct acl to match all IP's?
David
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
We're still having a problem with this. As you can see from the below the ACK goes to the port in the Contact field of the 200 OK instead of that of the UDP header, which is where their router is expecting to get the call from. Help! David On Nov 6, 2008, at 10:55 AM, David Aldworth wrote:
No love. They set extern ip so the IP comes through correctly, but the acl did not seem to have any affect. We are still sending to the wrong port. Sip trace, acl.conf.xml and sip profile are below:
U 2008/11/06 10:46:01.924795 70.88.65.1:50085 -> 70.42.223.23:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 70.42.223.23 ;branch=z9hG4bKU7360cS96r7Sg;received=70.42.223.23;rport=5060. From: "TELIAX FAX" <sip:303825XXXX@70.42.223.23>;tag=armgX7QeNQ94N. To: <sip:317376XXXX@70.88.65.1:50085>. Call-ID: 9e67419c-26cd-122c-0b81-e9d53e66cb70. CSeq: 106878444 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Contact: <sip:317376XXXX@70.88.65.1>. Content-Length: 0. .
U 2008/11/06 10:46:01.931791 70.88.65.1:50085 -> 70.42.223.23:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 70.42.223.23 ;branch=z9hG4bKU7360cS96r7Sg;received=70.42.223.23;rport=5060. From: "TELIAX FAX" <sip:303825XXXX@70.42.223.23>;tag=armgX7QeNQ94N. To: <sip:317376XXXX@70.88.65.1:50085>;tag=as78a21a0c. Call-ID: 9e67419c-26cd-122c-0b81-e9d53e66cb70. CSeq: 106878444 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Contact: <sip:317376XXXX@70.88.65.1>. Content-Length: 0. .
U 2008/11/06 10:46:01.932294 70.88.65.1:50085 -> 70.42.223.23:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 70.42.223.23 ;branch=z9hG4bKU7360cS96r7Sg;received=70.42.223.23;rport=5060. From: "TELIAX FAX" <sip:303825XXXX@70.42.223.23>;tag=armgX7QeNQ94N. To: <sip:317376XXXX@70.88.65.1:50085>;tag=as78a21a0c. Call-ID: 9e67419c-26cd-122c-0b81-e9d53e66cb70. CSeq: 106878444 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Contact: <sip:317376XXXX@70.88.65.1>. Content-Type: application/sdp. Content-Length: 257. . v=0. o=root 2901 2901 IN IP4 70.88.65.1. s=session. c=IN IP4 70.88.65.1. t=0 0. m=audio 19378 RTP/AVP 0 8 3 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:3 GSM/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -.
U 2008/11/06 10:46:01.932694 70.42.223.23:5060 -> 70.88.65.1:5060 ACK sip:317376XXXX@70.88.65.1 SIP/2.0. Via: SIP/2.0/UDP 70.42.223.23;rport;branch=z9hG4bKvgXZ279c41Xcc. Max-Forwards: 70. From: "TELIAX FAX" <sip:303825XXXX@70.42.223.23>;tag=armgX7QeNQ94N. To: <sip:317376XXXX@70.88.65.1:50085>;tag=as78a21a0c. Call-ID: 9e67419c-26cd-122c-0b81-e9d53e66cb70. CSeq: 106878444 ACK. Contact: <sip:mod_sofia@70.42.223.23:5060>. Content-Length: 0.
Here is the acl:
<configuration name="acl.conf" description="Network Lists"> <network-lists> <list name="dl-candidates" default="allow"> <node type="deny" cidr="10.0.0.0/8"/> <node type="deny" cidr="172.16.0.0/12"/> <node type="deny" cidr="192.168.0.0/16"/> </list> <list name="rfc1918" default="deny"> <node type="allow" cidr="10.0.0.0/8"/> <node type="allow" cidr="172.16.0.0/12"/> <node type="allow" cidr="192.168.0.0/16"/> </list> <list name="lan" default="allow"> <node type="deny" cidr="192.168.42.0/24"/> <node type="allow" cidr="192.168.42.42/32"/> </list> <list name="strict" default="deny"> <node type="allow" cidr="208.102.123.124/32"/> </list> <list name="domains" default="deny"> <node type="allow" domain="$${domain}"/> </list> <list name="nat" default="allow"> <node type="allow" cidr="0.0.0.0/0"/> </list> </network-lists> </configuration>
And here is the sip profile:
<profile name="external">
<gateways> <X-PRE-PROCESS cmd="include" data="external/*.xml"/> </gateways>
<domains> <domain name="$${domain}" parse="true"/> </domains>
<settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="multiple-registrations" value="true"/> <param name="manage-presence" value="true"/> <param name="aggressive-nat-detection" value="true"/> <param name="NDLB-force-rport" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="true"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="apply-nat-acl" value="nat"/> </settings> </profile>
On Nov 6, 2008, at 8:37 AM, Anthony Minessale wrote:
doh, I keep doing that sorry.
apply-nat-acl not apply_nat_acl
On Thu, Nov 6, 2008 at 8:22 AM, David Aldworth <daldworth@teliax.com> wrote: Yes. Below are settings that have been persistent through recent testing. Is there anything else we can try or should we open a jira?
<settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="multiple-registrations" value="true"/> <param name="manage-presence" value="true"/> <param name="aggressive-nat-detection" value="true"/> <param name="NDLB-force-rport" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="true"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="apply_nat_acl" value="nat"/> </settings>
On Nov 6, 2008, at 7:01 AM, Anthony Minessale wrote:
did you remember to add <param name="apply_nat_acl" value="nat"/> to the profile in question and restart?
On Wed, Nov 5, 2008 at 10:39 PM, David Aldworth <daldworth@teliax.com
wrote: Brian, we updated the acl to:
<list name="nat" default="allow"> <node type="allow" cidr="0.0.0.0/0"/> </list>
And the ACK is still going to the wrong (right but wrong) ip/port.
Is there any way to get that ACK to go to the ip/port of the UDP header?
David
On Nov 5, 2008, at 4:21 PM, Brian West wrote:
0.0.0.0/0 should match all IP space.
/b
On Nov 5, 2008, at 5:16 PM, David Aldworth wrote:
Anthony, In hopes of matching all IP's we added a very simple:
<list name="nat" default="allow"> </list>
To the acl.conf.xml and we added:
<param name="apply_nat_acl" value="nat"/>
To the sip profile. Unfortunately there was no affect. What would be the correct acl to match all IP's?
David
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
brian is out of town today can you ping me on irc and send me to login credential and i can try to have a look. again, you understand that we are not doing anything wrong here and what I am trying to do is a hack for your sake right? On Wed, Nov 19, 2008 at 8:05 AM, David Aldworth <daldworth@teliax.com>wrote:
We're still having a problem with this. As you can see from the below the ACK goes to the port in the Contact field of the 200 OK instead of that of the UDP header, which is where their router is expecting to get the call from. Help!
David
On Nov 6, 2008, at 10:55 AM, David Aldworth wrote:
No love. They set extern ip so the IP comes through correctly, but the acl did not seem to have any affect. We are still sending to the wrong port. Sip trace, acl.conf.xml and sip profile are below: U 2008/11/06 10:46:01.924795 70.88.65.1:50085 -> 70.42.223.23:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 70.42.223.23;branch=z9hG4bKU7360cS96r7Sg;received= 70.42.223.23;rport=5060. From: "TELIAX FAX" <sip:303825XXXX@70.42.223.23>;tag=armgX7QeNQ94N. To: <sip:317376XXXX@70.88.65.1:50085>. Call-ID: 9e67419c-26cd-122c-0b81-e9d53e66cb70. CSeq: 106878444 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Contact: <sip:317376XXXX@70.88.65.1>. Content-Length: 0. .
U 2008/11/06 10:46:01.931791 70.88.65.1:50085 -> 70.42.223.23:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 70.42.223.23;branch=z9hG4bKU7360cS96r7Sg;received= 70.42.223.23;rport=5060. From: "TELIAX FAX" <sip:303825XXXX@70.42.223.23>;tag=armgX7QeNQ94N. To: <sip:317376XXXX@70.88.65.1:50085>;tag=as78a21a0c. Call-ID: 9e67419c-26cd-122c-0b81-e9d53e66cb70. CSeq: 106878444 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Contact: <sip:317376XXXX@70.88.65.1>. Content-Length: 0. .
U 2008/11/06 10:46:01.932294 70.88.65.1:50085 -> 70.42.223.23:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 70.42.223.23;branch=z9hG4bKU7360cS96r7Sg;received= 70.42.223.23;rport=5060. From: "TELIAX FAX" <sip:303825XXXX@70.42.223.23>;tag=armgX7QeNQ94N. To: <sip:317376XXXX@70.88.65.1:50085>;tag=as78a21a0c. Call-ID: 9e67419c-26cd-122c-0b81-e9d53e66cb70. CSeq: 106878444 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Contact: <sip:317376XXXX@70.88.65.1>. Content-Type: application/sdp. Content-Length: 257. . v=0. o=root 2901 2901 IN IP4 70.88.65.1. s=session. c=IN IP4 70.88.65.1. t=0 0. m=audio 19378 RTP/AVP 0 8 3 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:3 GSM/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -.
U 2008/11/06 10:46:01.932694 70.42.223.23:5060 -> 70.88.65.1:5060 ACK sip:317376XXXX@70.88.65.1 SIP/2.0. Via: SIP/2.0/UDP 70.42.223.23;rport;branch=z9hG4bKvgXZ279c41Xcc. Max-Forwards: 70. From: "TELIAX FAX" <sip:303825XXXX@70.42.223.23>;tag=armgX7QeNQ94N. To: <sip:317376XXXX@70.88.65.1:50085>;tag=as78a21a0c. Call-ID: 9e67419c-26cd-122c-0b81-e9d53e66cb70. CSeq: 106878444 ACK. Contact: <sip:mod_sofia@70.42.223.23:5060>. Content-Length: 0.
Here is the acl:
<configuration name="acl.conf" description="Network Lists"> <network-lists> <list name="dl-candidates" default="allow"> <node type="deny" cidr="10.0.0.0/8"/> <node type="deny" cidr="172.16.0.0/12"/> <node type="deny" cidr="192.168.0.0/16"/> </list> <list name="rfc1918" default="deny"> <node type="allow" cidr="10.0.0.0/8"/> <node type="allow" cidr="172.16.0.0/12"/> <node type="allow" cidr="192.168.0.0/16"/> </list> <list name="lan" default="allow"> <node type="deny" cidr="192.168.42.0/24"/> <node type="allow" cidr="192.168.42.42/32"/> </list> <list name="strict" default="deny"> <node type="allow" cidr="208.102.123.124/32"/> </list> <list name="domains" default="deny"> <node type="allow" domain="$${domain}"/> </list> <list name="nat" default="allow"> <node type="allow" cidr="0.0.0.0/0"/> </list> </network-lists> </configuration>
And here is the sip profile:
<profile name="external">
<gateways> <X-PRE-PROCESS cmd="include" data="external/*.xml"/> </gateways>
<domains> <domain name="$${domain}" parse="true"/> </domains>
<settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="multiple-registrations" value="true"/> <param name="manage-presence" value="true"/> <param name="aggressive-nat-detection" value="true"/> <param name="NDLB-force-rport" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="true"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="apply-nat-acl" value="nat"/> </settings> </profile>
On Nov 6, 2008, at 8:37 AM, Anthony Minessale wrote:
doh, I keep doing that sorry.
apply-nat-acl not apply_nat_acl
On Thu, Nov 6, 2008 at 8:22 AM, David Aldworth <daldworth@teliax.com>wrote:
Yes. Below are settings that have been persistent through recent testing. Is there anything else we can try or should we open a jira? <settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="multiple-registrations" value="true"/> <param name="manage-presence" value="true"/> <param name="aggressive-nat-detection" value="true"/> <param name="NDLB-force-rport" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="true"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="apply_nat_acl" value="nat"/> </settings>
On Nov 6, 2008, at 7:01 AM, Anthony Minessale wrote:
did you remember to add <param name="apply_nat_acl" value="nat"/> to the profile in question and restart?
On Wed, Nov 5, 2008 at 10:39 PM, David Aldworth <daldworth@teliax.com>wrote:
Brian, we updated the acl to:
<list name="nat" default="allow"> <node type="allow" cidr="0.0.0.0/0"/> </list>
And the ACK is still going to the wrong (right but wrong) ip/port.
Is there any way to get that ACK to go to the ip/port of the UDP header?
David
On Nov 5, 2008, at 4:21 PM, Brian West wrote:
0.0.0.0/0 should match all IP space.
/b
On Nov 5, 2008, at 5:16 PM, David Aldworth wrote:
Anthony, In hopes of matching all IP's we added a very simple:
<list name="nat" default="allow"> </list>
To the acl.conf.xml and we added:
<param name="apply_nat_acl" value="nat"/>
To the sip profile. Unfortunately there was no affect. What would be the correct acl to match all IP's?
David
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
I'm in there. bigdc1 On Nov 19, 2008, at 7:12 AM, Anthony Minessale wrote:
brian is out of town today can you ping me on irc and send me to login credential and i can try to have a look.
again, you understand that we are not doing anything wrong here and what I am trying to do is a hack for your sake right?
On Wed, Nov 19, 2008 at 8:05 AM, David Aldworth <daldworth@teliax.com> wrote: We're still having a problem with this. As you can see from the below the ACK goes to the port in the Contact field of the 200 OK instead of that of the UDP header, which is where their router is expecting to get the call from.
Help!
David
On Nov 6, 2008, at 10:55 AM, David Aldworth wrote:
No love. They set extern ip so the IP comes through correctly, but the acl did not seem to have any affect. We are still sending to the wrong port. Sip trace, acl.conf.xml and sip profile are below:
U 2008/11/06 10:46:01.924795 70.88.65.1:50085 -> 70.42.223.23:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 70.42.223.23 ;branch=z9hG4bKU7360cS96r7Sg;received=70.42.223.23;rport=5060. From: "TELIAX FAX" <sip:303825XXXX@70.42.223.23>;tag=armgX7QeNQ94N. To: <sip:317376XXXX@70.88.65.1:50085>. Call-ID: 9e67419c-26cd-122c-0b81-e9d53e66cb70. CSeq: 106878444 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Contact: <sip:317376XXXX@70.88.65.1>. Content-Length: 0. .
U 2008/11/06 10:46:01.931791 70.88.65.1:50085 -> 70.42.223.23:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 70.42.223.23 ;branch=z9hG4bKU7360cS96r7Sg;received=70.42.223.23;rport=5060. From: "TELIAX FAX" <sip:303825XXXX@70.42.223.23>;tag=armgX7QeNQ94N. To: <sip:317376XXXX@70.88.65.1:50085>;tag=as78a21a0c. Call-ID: 9e67419c-26cd-122c-0b81-e9d53e66cb70. CSeq: 106878444 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Contact: <sip:317376XXXX@70.88.65.1>. Content-Length: 0. .
U 2008/11/06 10:46:01.932294 70.88.65.1:50085 -> 70.42.223.23:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 70.42.223.23 ;branch=z9hG4bKU7360cS96r7Sg;received=70.42.223.23;rport=5060. From: "TELIAX FAX" <sip:303825XXXX@70.42.223.23>;tag=armgX7QeNQ94N. To: <sip:317376XXXX@70.88.65.1:50085>;tag=as78a21a0c. Call-ID: 9e67419c-26cd-122c-0b81-e9d53e66cb70. CSeq: 106878444 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Contact: <sip:317376XXXX@70.88.65.1>. Content-Type: application/sdp. Content-Length: 257. . v=0. o=root 2901 2901 IN IP4 70.88.65.1. s=session. c=IN IP4 70.88.65.1. t=0 0. m=audio 19378 RTP/AVP 0 8 3 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:3 GSM/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -.
U 2008/11/06 10:46:01.932694 70.42.223.23:5060 -> 70.88.65.1:5060 ACK sip:317376XXXX@70.88.65.1 SIP/2.0. Via: SIP/2.0/UDP 70.42.223.23;rport;branch=z9hG4bKvgXZ279c41Xcc. Max-Forwards: 70. From: "TELIAX FAX" <sip:303825XXXX@70.42.223.23>;tag=armgX7QeNQ94N. To: <sip:317376XXXX@70.88.65.1:50085>;tag=as78a21a0c. Call-ID: 9e67419c-26cd-122c-0b81-e9d53e66cb70. CSeq: 106878444 ACK. Contact: <sip:mod_sofia@70.42.223.23:5060>. Content-Length: 0.
Here is the acl:
<configuration name="acl.conf" description="Network Lists"> <network-lists> <list name="dl-candidates" default="allow"> <node type="deny" cidr="10.0.0.0/8"/> <node type="deny" cidr="172.16.0.0/12"/> <node type="deny" cidr="192.168.0.0/16"/> </list> <list name="rfc1918" default="deny"> <node type="allow" cidr="10.0.0.0/8"/> <node type="allow" cidr="172.16.0.0/12"/> <node type="allow" cidr="192.168.0.0/16"/> </list> <list name="lan" default="allow"> <node type="deny" cidr="192.168.42.0/24"/> <node type="allow" cidr="192.168.42.42/32"/> </list> <list name="strict" default="deny"> <node type="allow" cidr="208.102.123.124/32"/> </list> <list name="domains" default="deny"> <node type="allow" domain="$${domain}"/> </list> <list name="nat" default="allow"> <node type="allow" cidr="0.0.0.0/0"/> </list> </network-lists> </configuration>
And here is the sip profile:
<profile name="external">
<gateways> <X-PRE-PROCESS cmd="include" data="external/*.xml"/> </gateways>
<domains> <domain name="$${domain}" parse="true"/> </domains>
<settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="multiple-registrations" value="true"/> <param name="manage-presence" value="true"/> <param name="aggressive-nat-detection" value="true"/> <param name="NDLB-force-rport" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="true"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="apply-nat-acl" value="nat"/> </settings> </profile>
On Nov 6, 2008, at 8:37 AM, Anthony Minessale wrote:
doh, I keep doing that sorry.
apply-nat-acl not apply_nat_acl
On Thu, Nov 6, 2008 at 8:22 AM, David Aldworth <daldworth@teliax.com> wrote: Yes. Below are settings that have been persistent through recent testing. Is there anything else we can try or should we open a jira?
<settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="multiple-registrations" value="true"/> <param name="manage-presence" value="true"/> <param name="aggressive-nat-detection" value="true"/> <param name="NDLB-force-rport" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="true"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="apply_nat_acl" value="nat"/> </settings>
On Nov 6, 2008, at 7:01 AM, Anthony Minessale wrote:
did you remember to add <param name="apply_nat_acl" value="nat"/> to the profile in question and restart?
On Wed, Nov 5, 2008 at 10:39 PM, David Aldworth <daldworth@teliax.com
wrote: Brian, we updated the acl to:
<list name="nat" default="allow"> <node type="allow" cidr="0.0.0.0/0"/> </list>
And the ACK is still going to the wrong (right but wrong) ip/port.
Is there any way to get that ACK to go to the ip/port of the UDP header?
David
On Nov 5, 2008, at 4:21 PM, Brian West wrote:
0.0.0.0/0 should match all IP space.
/b
On Nov 5, 2008, at 5:16 PM, David Aldworth wrote:
Anthony, In hopes of matching all IP's we added a very simple:
<list name="nat" default="allow"> </list>
To the acl.conf.xml and we added:
<param name="apply_nat_acl" value="nat"/>
To the sip profile. Unfortunately there was no affect. What would be the correct acl to match all IP's?
David
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
2008/11/19 David Aldworth <daldworth@teliax.com>:
We're still having a problem with this. As you can see from the below the ACK goes to the port in the Contact field of the 200 OK instead of that of the UDP header, which is where their router is expecting to get the call from.
The ACK for a 200 OK (INVITE) is an in-dialog request, so it must be sent to the SIP URI in the top most "Route" header (if there is a proxy in the patch that added it during the INVITE) or to the SIP URI indicated in the "Contact" of the 200 OK. This is the right behaviour according to the SIP specs. In your case, the 200 OK sent by Asterisk is: ----------------------- U 2008/11/06 10:46:01.932294 70.88.65.1:50085 -> 70.42.223.23:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 70.42.223.23;branch=z9hG4bKU7360cS96r7Sg;received=70.42.223.23;rport=5060. User-Agent: Asterisk PBX. Contact: <sip:317376XXXX@70.88.65.1>. ------------------ Note that the Contact says no port, so it means default SIP port => 5060. But note that the response comes from port 50085, why? it makes no sense, but it's not a problem in FS but in your environment or your network. Could you show us the INVITE sent from FS to Asterisk (including also the real sourde/destination address). Is that IVITE sent to port 50085 or to port 5060 of Asterisk? Is there NAT somewhere between FS and Asterisk in this scenario? -- Iñaki Baz Castillo <ibc@aliax.net>
participants (6)
-
Anthony Minessale
-
Brian West
-
David Aldworth
-
David Aldworth
-
Iñaki Baz Castillo
-
Michael Jerris