search for: rfc3581

Ok, this is highly confusing. hestia*CLI> sip show users Username Secret Accountcode Def.Context ACL NAT 2944030 2944030 oneeighty_start No RFC3581 2944035 2944035 oneeighty_start No RFC3581 sip users (type=friend) are in sip.conf. I have nat=no against all of them. Why does a 'sip show users' have RFC3581 against ALL my users? (there's a lot more than I pasted here) Thanks, D...

Hi list, I have a fresh install of Asterisk 12.0.0 and I'm going to use it only as a client. I'm trying to SIP REGISTER with a remote SIP provider. The situation is that Asterisk is running in a VMware VM with a RFC IP address (192.168.1.2). The provider of the VM performs static NAT from the RFC IP address to a dedicated public IP address, however, they are rewriting ports at will.

...: Presentation Allowed, Not Screened Callgroup : Pickupgroup : Mailbox : VM Extension : asterisk LastMsgsSent : 32767/65535 Call limit : 0 Dynamic : No Callerid : "" <> Expire : -1 Insecure : port,invite Nat : RFC3581 ACL : No CanReinvite : No PromiscRedir : No User=Phone : No Trust RPID : No Send RPID : No DTMFmode : rfc2833 LastMsg : 0 ToHost : sip.stanaphone.com Addr->IP : 204.147.183.18 Port 5060 Defaddr->IP : 0.0.0.0 Port 0 Def....

...ong ? Is this a bug ? [I rebooted asterisk, and now it works.] Regards Axelle. Logs of failed registration: > sip show users Username Secret Accountcode Def.Context ACL NAT IMSI208011234567890 sip-local No RFC3581 IMSI208302141472352 sip-external No RFC3581 IMSI208304424439206 sip-external No RFC3581 [Apr 8 15:01:01] NOTICE[20626]: chan_sip.c:15642 handle_request_register: Registration from 'IMSI208300618462231 <sip:IMSI208300618462231 at 127.0.0.1>' failed for '127.0.0.1'...

...nown CallingPres : Presentation Allowed, Not Screened Callgroup : Pickupgroup : Mailbox : VM Extension : asterisk LastMsgsSent : 32767/65535 Call limit : 0 Dynamic : Yes Callerid : "" <> Expire : 113 Insecure : no Nat : RFC3581 ACL : No CanReinvite : Yes PromiscRedir : No User=Phone : No Trust RPID : No Send RPID : No DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr->IP : 10.4.5.1 Port 5060 Defaddr->IP : 0.0.0.0 Port 5060 Sock fd : 24 Transport : TCP...

...f has: sipusers => mysql,dbname,ast_sip_users sippeers => mysql,dbname,ast_sip_users When I do a 'sip show peers' I see all my phones. When I do a 'sip show users' I only see a few of them. I can't work out why this is the case. They are also coming up with NAT as 'RFC3581', eventhough I have nat set to NO for every friend in the ast_sip_users table. In short, phones make and receive calls, so they should be defined as type=friend, right? Should I point sipusers and sippeers from extconfig to the same table? Why does 'extconfig' have sipusers and sippee...

...D 5060 Unmonitored usuario1/usuario1 10.xxx.xxx.xxx D 5060 Unmonitored 2 sip peers [2 online , 0 offline] sip show users Username Secret Accountcode Def.Context ACL NAT usuario2 usuario2 default No RFC3581 usuario1 usuario1 default No RFC3581 --- (8 headers 0 lines)--- Looking for 200.xxx.xxxx.xxx in default (domain ) Transmitting (no NAT) to 10.xxx.xxx.xxx:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.xxx.xxx.xxx ;rport;branch=z9hG4bK0a0101e20000001044479388000070d300...

...pability: 1822 > Non-Codec Capability: 1 > Their Codec Capability: 262 > Joint Codec Capability: 262 > Format ulaw > Theoretical Address: 204.147.183.18:5060 > Received Address: 204.147.183.18:5060 > NAT Support: RFC3581 > Audio IP: XX.XX.XX.XX (local) > Our Tag: as78cfb201 > Their Tag: da6aae9eb017f29b6c9de270fb85c352 > SIP User agent: Sippy > Original uri: sip:204.147.183.55:1024 > Caller-ID: XXXXXXXXXX &g...

...: Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup : Pickupgroup : Mailbox : 245 at device VM Extension : *97 LastMsgsSent : 32767/65535 Call limit : 50 Dynamic : Yes Callerid : "device" <245> MaxCallBR : 384 kbps Expire : 67 Insecure : no Nat : RFC3581 ACL : No T38 pt UDPTL : No CanReinvite : No PromiscRedir : No User=Phone : No Video Support: Yes Trust RPID : No Send RPID : No Subscriptions: Yes Overlap dial : Yes DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr->IP : 192.168.0.239 Port 5060 Defaddr->IP : 0.0.0.0 Port 5060 Def. Username: 245 S...

...ion Allowed, Not Screened Callgroup : Pickupgroup : Mailbox : VM Extension : asterisk LastMsgsSent : -1 Inc. limit : 0 Outg. limit : 0 Dynamic : Yes Callerid : "" <> Expire : 386 Expiry : 900 Insecure : no Nat : RFC3581 ACL : No CanReinvite : No PromiscRedir : No User=Phone : No DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr->IP : 192.168.10.32 Port 5060 Defaddr->IP : 0.0.0.0 Port 5060 Def. Username: 2608 SIP Options : (none) Codecs : 0x4 (ulaw)...

...ut: 20 secs Outbound reg. attempts: 0 Notify ringing state: Yes Notify hold state: No SIP Transfer mode: open Max Call Bitrate: 384 kbps Auto-Framing: No Default Settings: ----------------- Context: default Nat: RFC3581 DTMF: rfc2833 Qualify: 0 Use ClientCode: No Progress inband: Never Language: (Defaults to English) MOH Interpret: default MOH Suggest: Voice Mail Extension: asterisk ---- Parsing /etc/asterisk/extconfi...

...Yes Pedantic SIP support: No Reg. max duration: 3600 secs Reg. default duration: 120 secs Outbound reg. timeout: 20 secs Outbound reg. attempts: 0 Notify ringing state: Yes Default Settings: ----------------- Context: default Nat: RFC3581 DTMF: rfc2833 Qualify: 0 Use ClientCode: No Progress inband: Never Language: (Defaults to English) Musicclass: default Voice Mail Extension: asterisk ******************sip.conf file*************************...

...allgroup : Pickupgroup : Mailbox : VM Extension : asterisk LastMsgsSent : 32767/65535 Call limit : 99 Busy level : 1 Dynamic : Yes Callerid : "Matts SIP" <6668> MaxCallBR : 384 kbps Expire : 2016 Insecure : no Nat : RFC3581 ACL : No T38 pt UDPTL : No CanReinvite : No PromiscRedir : No User=Phone : No Video Support: No Text Support : No Trust RPID : No Send RPID : No Subscriptions: Yes Overlap dial : No DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost...

...sinband=no ; If we should generate in-band ringing always ;useragent=Asterisk PBX ; Allows you to change the user agent string ;nat=no ; NAT settings ; yes = Always ignore info and assume NAT ; no = Use NAT mode only according to RFC3581 ; never = Never attempt NAT mode or RFC3581 support ;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address ; Asterisk can register as a SIP user agent to a SIP proxy (provider) ; Format for the register statement is: ; register => us...

...984ee48048d9e01c@192.168.0.22 Our Codec Capability: 524302 Non-Codec Capability: 1 Their Codec Capability: 0 Joint Codec Capability: 0 Format UNKN Theoretical Address: 192.168.0.22:5060 Received Address: 192.168.0.22:5060 NAT Support: RFC3581 Our Tag: 1190462248 Their Tag: SIP User agent: Need Destroy: 0 Last Message: Promiscuous Redir: No Route: N/A DTMF Mode: rfc283 Here's a quote from a post to an earlier question by som...

...5c2cf755-ccde1b6c@x.x.x.xq: 520 REGISTER Our Codec Capability: 12 Non-Codec Capability: 1 Their Codec Capability: 0 Joint Codec Capability: 0 Format unknown Theoretical Address: x.x.x.x:5060 Received Address: x.x.x.x:5060 NAT Support: RFC3581 Our Tag: 715659627 Their Tag: SIP User agent: Need Destroy: 0 Last Message: Promiscuous Redir: No Route: N/A DTMF Mode: rfc2833 The sequence number (ie. 520) increases by 1 every time....

...rate in-band ringing always useragent=Asterisk ; Allows you to change the user agent string nat=no ; NAT settings ; yes = Always ignore info and assume NAT ; no = Use NAT mode only according to RFC3581 ; never = Never attempt NAT mode or RFC3581 support ; route = Assume NAT, don't send rport (work around more UNIDEN bugs) ;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address ; ;...

...dbd3f@203.196.128.56 Our Codec Capability: 256 Non-Codec Capability: 1 Their Codec Capability: 256 Joint Codec Capability: 256 Format g729 Theoretical Address: 203.196.128.56:5060 Received Address: 203.196.128.56:5060 NAT Support: RFC3581 Audio IP: 220.227.174.4 (local) Our Tag: as7a55ac7a Their Tag: 29258 SIP User agent: Username: 61356251162 Peername: 90340 Original uri: sip:61356251162@216.181.122.44:5060 Need Destroy:...

...resentation Allowed, Not Screened Callgroup : 1 Pickupgroup : 1 Mailbox : VM Extension : asterisk LastMsgsSent : 32767/65535 Call limit : 0 Dynamic : No Callerid : "" <2029191> MaxCallBR : 384 kbps Expire : -1 Insecure : no Nat : RFC3581 ACL : No T38 pt UDPTL : No CanReinvite : No PromiscRedir : No User=Phone : No Video Support: No Trust RPID : No Send RPID : No Subscriptions: No Overlap dial : No DTMFmode : auto LastMsg : 0 ToHost : sf2.clarocom.net Addr->IP : 200.105.69.132 Por...

...generate in-band ringing always ;useragent=Asterisk PBX ; Allows you to change the user agent string nat=yes ; NAT settings ; yes = Always ignore info and assume NAT ; no = Use NAT mode only according to RFC3581 ; never = Never attempt NAT mode or RFC3581 support ; route = Assume NAT, don't send rport (work around more UNIDEN bugs) ;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address ; ; Note...