Guys. Im having NAT problems. Any good tips on how to debug remote SIPS, how to see which ports are been sent and received, etc?
sip debug sip debug peer username sip debug peer ip_address -----Original Message----- From: Anton Krall [mailto:akrall-lists@intruder.com.mx] Sent: Monday, May 02, 2005 1:44 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Debuging SIP Guys. Im having NAT problems. Any good tips on how to debug remote SIPS, how to see which ports are been sent and received, etc? _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
sip debug Hope this will help.. Regrads, Primoz -----Original Message----- From: Anton Krall [mailto:akrall-lists@intruder.com.mx] Sent: 2. maj 2005 20:44 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Debuging SIP Guys. Im having NAT problems. Any good tips on how to debug remote SIPS, how to see which ports are been sent and received, etc? _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Hi all, I'm newbie to VOIP/SIP/asterisk... and I having problems with SIP on local network. I have Freebsd server 5.3 running asterisk and two x-lite clients. I added following lines to sip.conf [tina] type=friend host=dynamic dtmfmode=inband context=sip [primozz] type=friend host=dynamic dtmfmode=inband context=sip And following to extensions.conf [sip] exten => 1000,1,Dial,SIP/tina exten => 2000,1,Dial,SIP/primozz *CLI> sip show users Username Secret Accountcode Def.Context ACL NAT primozz sip No RFC35 tina sip No RFC35 I have X-Lite clinet on Win XP and while trying to make call to "tina" I got 404 error - not found. Same for vice versa...Both users are local.>From debug below following line:To: <sip:tina@192.168.1.3>;tag=as1283188b is very strange to me. Instead od 192.168.1.3 there should be 192.168.1.1. Do I need to put some ware static IP for each client ? And following is debug from asterisk: Peer audio RTP is at port 192.168.1.3:8000 Found description format pcmu Found description format pcma Found description format gsm Found description format iLBC Found description format speex Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Looking for tina in sip list_route: hop: <sip:primozz@192.168.1.3:5060> Reliably Transmitting (no NAT): SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bKBD00F479A53B4C97B6FF319C9D6B06CA From: Primoz <sip:primozz@192.168.1.3>;tag=1716760483 To: <sip:tina@192.168.1.3>;tag=as1283188b Call-ID: B5CEF7FC-640F-42FD-B93E-FFDE1F9EE6F8@192.168.1.3 CSeq: 22324 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:tina@192.168.1.12> Content-Length: 0 to 192.168.1.3:5060 Sip read: ACK sip:tina@192.168.1.3 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.3:5060;rport;branch=z9hG4bKBD00F479A53B4C97B6FF319C9D6B06CA From: Primoz <sip:primozz@192.168.1.3>;tag=1716760483 To: <sip:tina@192.168.1.3>;tag=as1283188b Contact: <sip:primozz@192.168.1.3:5060> Call-ID: B5CEF7FC-640F-42FD-B93E-FFDE1F9EE6F8@192.168.1.3 CSeq: 22324 ACK Max-Forwards: 70 Content-Length: 0 Thanks for help ! Regards, Primoz
Are you using IPTables? Here's my IPTables config for SIP and RTP... you need to open UDP ports 5060 and 10000-20000 $IPTABLES -A FORWARD -i $EXTIF -o $INTIF -p udp -m udp --sport 5060 --dport 5060 -j ACCEPT $IPTABLES -A FORWARD -i $EXTIF -o $INTIF -p udp -m udp --sport 10000:20000 --dport 10000:20000 -j ACCEPT $IPTABLES -A FORWARD -i $INTIF -o $EXTIF -p udp -m udp --sport 5060 --dport 5060 -j ACCEPT $IPTABLES -A FORWARD -i $INTIF -o $EXTIF -p udp -m udp --sport 10000:20000 --dport 10000:20000 -j ACCEPT $IPTABLES -t nat -A PREROUTING -i $EXTIF -p udp -m udp -s EXTERNAL_SIP_SOURCE --dport 5060 -j DNAT --to-destination ASTERISK_SERVER $IPTABLES -t nat -A PREROUTING -i $EXTIF -p udp -m udp -s EXTERNAL_SIP_SOURCE --dport 10000:20000 -j DNAT --to-destination ASTERISK_SERVER Hope this helps, Ian Ian Pattison, Senior Analyst Technology Associates Inc. Tel: 905-459-2100 ext. 204 Mobile: 416-568-6548 E-mail: ianp@technologyassociates.ca WWW: http://www.technologyassociates.ca>>> akrall-lists@intruder.com.mx 02/05/2005 14:44 >>>Guys. Im having NAT problems. Any good tips on how to debug remote SIPS, how to see which ports are been sent and received, etc? _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -------------- next part -------------- BEGIN:VCARD VERSION:2.1 FN:Ian Pattison EMAIL;WORK;PREF:ianp@technologyassociates.ca TEL;WORK:416-657-2464 ext. 204 N:Pattison;Ian TITLE:Senior Analyst ADR;INTL;WORK;PARCEL;POSTAL:;;9052 Creditview Rd.;Brampton;Ontario;L6V 1A1;Canada LABEL;INTL;WORK;PARCEL;POSTAL;ENCODING=QUOTED-PRINTABLE:Ian Pattison=0A9052 Creditview Rd.=0ABrampton, Ontario L6V 1A1=0ACanada LABEL;DOM;WORK;PARCEL;POSTAL;ENCODING=QUOTED-PRINTABLE:Ian Pattison=0A9052 Creditview Rd.=0ABrampton, Ontario L6V 1A1 TEL;CELL:416-568-6548 TEL;PREF:416-657-2464 ext. 204 TEL;WORK:905-459-2100 ext. 204 ORG:Technology Associates Inc. END:VCARD