search for: ip4

I am inquiring about how to setup a proper SPF record. I know there are SPF wizards/generators available but each seem to have a different "opinion" of what should be included and what should not be included. Let me give you a scenario of my setup, and hopefully someone can help me out. My domain is: test.com My mailserver hostname is: mail.host.com which also has a MATCHING PTR

...cording from the asterisk installation, although in fact the streaming is supposed to have started. When debugging with tcpdump, I have seen that all the successful calls have SDP negotiation that look like this: (inside INVITE request body from SIP carrier) v=0 o=msw.chance4minutes.net 1234 0 IN IP4 38.126.208.46 s=sip call c=IN IP4 38.126.208.46 t=0 0 m=audio 30552 RTP/AVP 18 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:20 (inside 200 OK response body from asterisk) v=0 o=ro...

...00036283-3383986922 User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled CSeq: 101 INVITE Max-Forwards: 6 Timestamp: 1066825189 Contact: <sip:52880472@200.61.32.142:5060;user=phone> Expires: 180 Content-Type: application/sdp Content-Length: 166 v=0 o=CiscoSystemsSIP-GW-UserAgent 624 4121 IN IP4 200.61.32.142 s=SIP Call c=IN IP4 200.61.32.142 t=0 0 m=audio 20476 RTP/AVP 8 0 18 65535 65535 65535 4 65535 Oct 22 09:19:49.930: Received: SIP/2.0 100 Trying Via: SIP/2.0/UDP 200.61.32.142:5060 From: "52880472" <sip:52880472@200.61.32.142> To: <sip:2003@200.61.32.238;user=phon...

...lems. I have not specified a transport in the endpoint section, so that the appropriate transport which corresponds to the registration can be used. Now I have noticed, if an phone is registered from an ipv4 only endpoint and is performing an outgoing call, my asterisk server is answering with an IP4 RTP IPv6 address: Example: <--- Received SIP request (1235 bytes) from UDP:157.161.4.172:5060 ---> [...] v=0 o=MxSIP 0 20 IN IP4 157.161.4.172 s=SIP Call c=IN IP4 157.161.4.172 t=0 0 m=audio 6018 RTP/AVP 9 8 97 91 101 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 CLEARMODE/8000 a=rt...

...9868731-3383986922 User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled CSeq: 101 INVITE Max-Forwards: 6 Timestamp: 1064519388 Contact: <sip:52880472@172.16.254.96:5060;user=phone> Expires: 180 Content-Type: application/sdp Content-Length: 167 v=0 o=CiscoSystemsSIP-GW-UserAgent 8010 6925 IN IP4 172.16.254.96 s=SIP Call c=IN IP4 172.16.254.96 t=0 0 m=audio 20388 RTP/AVP 8 0 18 65535 65535 65535 4 65535 15 headers, 6 lines Using latest request as basis request Sending to 172.16.254.96 : 5060 (non-NAT) Found audio format ALAW Found audio format UNKN Found audio format UNKN Found audio forma...

...ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.0.3.0127 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 245 v=0 o=- 978307756 978307756 IN IP4 10.253.4.50 s=Polycom IP Phone c=IN IP4 10.253.4.50 t=0 0 m=audio 2228 RTP/AVP 0 18 8 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 --- (14 headers 11 lines) --- Using INVITE request as basis request - a857d7ac-36f29d46-4d6ef889@...

....5>> From: "Do Nguyen Ha"<sip:test at 192.168.1.5 <sip%3Atest at 192.168.1.5> >;tag=f543a140 Call-ID: MjFlODY2MmE0OTBmMzE2ZDJhMWEzN2ZmYzUwM2ZkODg. CSeq: 2 INVITE Content-Type: application/sdp User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 261 v=0 o=- 8 2 IN IP4 192.168.1.4 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.1.4 t=0 0 m=audio 50420 RTP/AVP 107 0 8 101 <--- Transmitting (no NAT) to 192.168.1.4:18341 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.4:18341 ;branch=z9hG4bK-d8754z-da695e1f167deb68-1---d8754z-;received=192.168.1.4;rport=18341 Fr...

...60000 ms Keepalive : 0 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs RTP Engine : asterisk Parkinglot : Use Reason : No Encryption : No When the call comes in the SDP contains :- v=0. o=root 973184584 973184584 IN IP4 81.x.x.x s=session. c=IN IP4 81.x.x.x t=0 0. m=audio 11370 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. and we reply back with :- v=0. o=root 822402971 822402971 IN IP4 88.x.x.x s=Asterisk PBX 11.2-cert...

....0.1> Call-ID: 44f37f304731f0ef212d4ffd38846d1c@192.168.0.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sat, 24 Jun 2006 16:12:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 235 v=0 o=root 3131 3131 IN IP4 192.168.0.1 s=session c=IN IP4 192.168.0.1 t=0 0 m=audio 12580 RTP/AVP 18 101 a=rtpmap:18 H723/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- asterisk1*CLI> Retransmitting #1 (no NAT) to 192.168.0.254:5060: INVITE sip:165622270602000@192...

Hi Guys, Something I have noticed while dealing with T.38 and re-invites in Asterisk 1.4.22. I have a provider who re-invites with the following sdp (message flow PROVIDER_EQPMT -> ASTERISK): """ . v=0. o=SIP_5F9 123456 654322 IN IP4 CONN_IP_PROVIDER. s=-. c=IN IP4 CONN_IP_PROVIDER. t=0 0. m=audio 0 RTP/AVP 0. m=image 26858 udptl t38. a=T38FaxMaxBuffer:288. a=T38FaxRateManagement:transferredTCF. a=T38FaxUdpEC:t38UDPRedundancy. """ The answer coming from asterisk in this case is: """ . v=0. o=root...

...at sipgate.de CSeq: 103 INVITE Contact: <sip:0xxxxxxxx9 at 217.10.77.115:5060> max-forwards: 66 supported: replaces Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Content-Type: application/sdp Content-Length: 394 v=0 o=root 15363811 15363812 IN IP4 192.168.2.1 s=sipgate VoIP GW c=IN IP4 192.168.2.1 t=0 0 m=audio 7070 RTP/AVP 8 0 3 97 18 112 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:112 G726-32/8000 a=rtpmap:101 telephone-event/80...

...:3E:FA:CE:4A GENERAL.MTU: 1500 GENERAL.STATE: 100 (connected) GENERAL.CONNECTION: System-eth0 GENERAL.CON-PATH: /org/freedesktop/NetworkManager/ActiveConnection/2 WIRED-PROPERTIES.CARRIER: on IP4.ADDRESS[1]: 192.168.1.5/24 IP4.ADDRESS[2]: 153.153.xxx.xxx/32 IP4.GATEWAY: 192.168.1.1 IP4.DNS[1]: 8.8.8.8 IP4.DNS[2]: 8.8.4.4 IP6.ADDRESS[1]: f...

Hi, I have 2 systems (A and B). I have an 800 number... when someone calls the 800 number it goes: IAX2-->A---IAX---B--->SIP PHONE However.. if the user calling the 800 number is a SIP user that is registered to A it goes: SIP--->A---IAX---B--->SIP PHONE This is the problem... when a call comes in from the IAX2 800 provider, things work fine... however if a SIP user registered to

...001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:Voicemail at 10.2.0.2> Content-Type: application/sdp Content-Length: 256 v=0 o=root 27452 27452 IN IP4 10.2.0.2 s=session c=IN IP4 10.2.0.2 t=0 0 m=audio 13256 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> Retransmitting #1 (no NAT) to 10.2.0.203:5060: SIP/2.0 200 OK Vi...

...192.168.1.223> Call-ID: fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223 CSeq: 62159 INVITE User-Agent: Grandstream SIP UA 1.0.3.81 Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Content-Length: 263 v=0 o=budgtone 0 0 IN IP4 192.168.1.223 s=- c=IN IP4 192.168.1.223 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 15 a=ptime:20 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 12 headers, 13 lines Using latest request as basis request Sending to 1...

with wireshark i need decrypt traffic every call which is time consuming. get debug from pjnat through asterisk is not possible because of technical reasons or nobody did it? in my case its strange that ice candidates are the same good call v=0 o=- 3669976329745317845 2 IN IP4 127.0.0.1 s=- t=0 0 a=msid-semantic: WMS EoNIdKcMZvWBLULGqGPJTDe12ujjFEemeapo m=audio 52421 RTP/SAVPF 8 0 101 c=IN IP4 10.2.152.36 a=rtcp:9 IN IP4 0.0.0.0 a=candidate:3607370648 1 udp 2122260223 10.2.152.36 52421 typ host generation 0 network-id 1 network-cost 10 a=candidate:2575820648 1 tcp 15182...

...p>;tag=24957277 To: <sip:3004@<asterisk ip>> Via: SIP/2.0/UDP 172.16.1.54:5060;branch=z9hG4bK3f113fc0c05ec1deece622bd0ed4a521 Max-Forwards: 70 Contact: "3002" <sip:<gateway1>:5060;transport=udp> Content-Type: application/sdp Content-Length: 148 v=0 o=par 0 0 IN IP4 <gateway1> s=- c=IN IP4 <gateway1> t=0 0 m=audio 22224 RTP/AVP 0 3 4 5 6 8 15 18 m=video 22222 RTP/AVP 26 34 31 10 headers, 7 lines Using latest request as basis request Sending to 172.16.1.54 : 5060 (non-NAT) Found audio format UNKN Found audio format UNKN Found audio format ULAW Foun...

...2.168.1.100>;tag=as476eacfd To: <sip:16507148980@natrelay.deltathree.com> Contact: <sip:asterisk@192.168.1.100> Call-ID: 2a3fe9bc1d8dd93a400263c775c63f5b@192.168.1.100 CSeq: 102 INVITE User-Agent: Asterisk PBX Content-Type: application/sdp Content-Length: 236 v=0 o=root 3296 3296 IN IP4 192.168.1.100 s=session c=IN IP4 192.168.1.100 t=0 0 m=audio 10860 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 (no NAT) to 213.137.73.140:5060 -- Called 16507148980@iconnect Retransmitting #1 (no NAT): INVITE...

...ine => aaln/1 line => aaln/2 line => aaln/3 line => aaln/4 extension.conf [local] ignorepat => 9 exten => _9XXXXXXX,1,Dial,SIP/${EXTEN:1}@IP.IP.IP.IP Some logs from Asterisk: First mgcp CRCX after hang up: Posting Request: CRCX 323 aaln/1@DLINK MGCP 1.0 v=0 o=root 23577 23577 IN IP4 X.X.X.X s=session c=IN IP4 X.X.X.X t=0 0 m=audio 14548 RTP/AVP 18 a=rtpmap:18 G729/8000 After that I enter phone number and sent call to sip server: -- Executing Dial("MGCP/aaln/1@DLINK-0", "SIP/3632034@IP.IP.IP.IP") in new stack INVITE sip:3632034@IP.IP.IP.IP SIP/2.0 &lt...

...CSeq: 101 INVITE Max-Forwards: 70 Contact: PAP 220 <sip:220@192.168.254.44:5060> Expires: 240 User-Agent: Linksys/PAP2-3.1.3(LS) Content-Length: 246 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 261305180 261305180 IN IP4 192.168.254.44 s=- c=IN IP4 192.168.254.44 t=0 0 m=audio 16392 RTP/AVP 18 100 101 a=rtpmap:18 G729a/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv INVITE sip:203@192.168.121.20 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.44:5060;branch=z9hG4bK-3fda1e15...