# SIP Configuration in Asterisk and SIP Phone **URL:** \ **Category:** Asterisk Support\ **Created:** [July 23, 2013, 6:27pm UTC](https://community.asterisk.org/t/sip-configuration-in-asterisk-and-sip-phone/43446 "2013-07-23T18:27:22Z")\ **Posts on this page:** 9\ **Page:** 1
**Author:** ![KBennett](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/kbennett/32/146_2.png) [@KBennett](https://community.asterisk.org/u/KBennett)\ **Post date:** [July 23, 2013, 6:27pm UTC](https://community.asterisk.org/t/sip-configuration-in-asterisk-and-sip-phone/43446/1 "2013-07-23T18:27:22Z")
Hi, I’m running Asterisk 1.8.4.4. on Ubuntu 11.10. I am trying to get my Nexus 4 to sink with my Asterisk box via SIP. I can’t seem to figure out where the problem is. My phone keeps saying “Account Registration is unsuccessful”. Here are my details. Any suggestions or directions would be appreciated! Thanks for your time! **_sip.conf_** [nx4] type = friend allowguest = yes context = incoming host = dynamic nat = yes udpbindaddr = 0.0.0.0 username = nx4 secret = XXXXXX bindport = 5060 canreinvite = no insecure = invite **_dialplan.conf_** \* [incoming] exten =\> 4242, 1, Goto(google, start, 1) [google] exten =\> start, 1, Answer() same =\> n, Playback(tt-monkeysintro) same =\> n, Playback(hello-world) same =\> n, Playback(vm-goodbye) same =\> n, Hangup() \ ***Sip settings on phone** Username: nx4 password: XXXXXX Server: [xxx.xx.xxx.xxx](http://xxx.xx.xxx.xxx) ;ip for my asterisk box Outbound proxy address: [xxx.xx.xxx.xxx](http://xxx.xx.xxx.xxx) ;ip for my asterisk box Port Number: 5060 Transportation type: UDP Send keep-alive: Automatic **_CLI\> sip show peers_** Name/username Host Dyn Forcerport ACL Port Status 1001/1001 (Unspecified) D N 0 UNKNOWN kbeezy/0000000070 xx.xx.xxx.xx N 5060 OK (55 ms) nx4/nx4 (Unspecified) D N 0 Unmonitored office-phone (Unspecified) D N 0 UNKNOWN zoiper/zoiper [xx.xx.xx.xxx](http://xx.xx.xx.xxx) D 5060 Unmonitored 5 sip peers [Monitored: 1 online, 2 offline Unmonitored: 1 online, 1 offline] \*\*\*\*\*\*\* sip debug\*\*\*\*\*\*\*\*\* \<— SIP read from UDP:xx.xx.77.xxx:5060 —\> \<-------------\> Reliably Transmitting (NAT) to xx.x4.xx0.xx:5060: OPTIONS sip:xx.x4.xx0.xx SIP/2.0 Via: SIP/2.0/UDP [xxx.xx.xxx.xxx:5060](http://xxx.xx.xxx.xxx:5060);branch=z9hG4bK3f47219a;rport Max-Forwards: 70 From: “asterisk” sip:asterisk@1xxx.xx.xxx.xxx;tag=as7313814a To: sip:xx.x4.xx0.xx Contact: sip:asterisk@xxx.xx.xxx.xxx:5060 Call-ID: [4d7bb95b6b6904094c4eda5510a8c7ac@xxx.xx.xxx.xxx](mailto:4d7bb95b6b6904094c4eda5510a8c7ac@xxx.xx.xxx.xxx):5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1.1 Date: Tue, 23 Jul 2013 18:03:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 * * * \<— SIP read from UDP:xx.x4.xx0.xx:5060 —\> SIP/2.0 404 Not Found Via: SIP/2.0/UDP [xxx.xx.xxx.xxx:5060](http://xxx.xx.xxx.xxx:5060);branch=z9hG4bK3f47219a;[received=xxx.xx.xxx.xxx](http://received=xxx.xx.xxx.xxx);rport=5060 From: “asterisk” sip:asterisk@xxx.xx.xxx.xxx;tag=as7313814a To: sip:xx.x4.xx0.xx;tag=as6793089d Call-ID: [4d7bb95b6b6904094c4eda5510a8c7ac@xxx.xx.xxx.xxx](mailto:4d7bb95b6b6904094c4eda5510a8c7ac@xxx.xx.xxx.xxx):5060 CSeq: 102 OPTIONS User-Agent: IPKall Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Accept: application/sdp Content-Length: 0 \<-------------\> — (11 headers 0 lines) — Really destroying SIP dialog '4d7bb95b6b6904094c4eda5510a8c7ac@xxx.xx.xxx.xxx:5060’ Method: OPTIONS Really destroying SIP dialog ‘4eba1837cab8596cf822587f2b8f46b0@:xx.xx.77.xxx’ Method: REGISTER \<— SIP read from UDP::xx.xx.77.xxx:57120 —\> REGISTER sip:xxx.xx.xxx.xxx SIP/2.0 Call-ID: 2ed25e42a5dbecde0244eba5c6c1c637@:xx.xx.77.xxx CSeq: 7516 REGISTER From: “nx4” sip:nx4@xxx.xx.xxx.xxx;tag=1453620334 To: “nx4” sip:nx4@xxx.xx.xxx.xxx Via: SIP/2.0/UDP:xx.xx.77.xxx:57120;branch=z9hG4bK015b3fd1093f830d5c3963d7449cda6c323939;rport Max-Forwards: 70 User-Agent: SIPAUA/0.1.001 Contact: “nx4” sip:nx4@:57120:xx.xx.77.xxx;transport=udp Expires: 3600 Content-Length: 0 \<-------------\> — (11 headers 0 lines) — Sending to 😠x.xx.77.xxx:57120 (no NAT) \<— Transmitting (NAT) to 😠x.xx.77.xxx:57120 —\> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 24.90.77.182:57120;branch=z9hG4bK015b3fd1093f830d5c3963d7449cda6c323939;received=:xx.xx.77.xxx;rport=57120 From: “nx4” sip:nx4@xxx.xx.xxx.xxx;tag=1453620334 To: “nx4” sip:nx4@xxx.xx.xxx.xxx;tag=as7a11e87e Call-ID: 2ed25e42a5dbecde0244eba5c6c1c637@:xx.xx.77.xxx CSeq: 7516 REGISTER Server: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="21f9bff1" Content-Length: 0 \<------------\> Scheduling destruction of SIP dialog ‘2ed25e42a5dbecde0244eba5c6c1c637@:xx.xx.77.xxx’ in 32000 ms (Method: REGISTER) Really destroying SIP dialog ‘9795944343362fbb2ff976c7b5838f86@:xx.xx.77.xxx’ Method: REGISTER \<— SIP read from UDP::xx.xx.77.xxx:5060 —\> \<-------------\> Really destroying SIP dialog ‘2ed25e42a5dbecde0244eba5c6c1c637@:xx.xx.77.xxx’ Method: REGISTER \<— SIP read from UDP:24.90.77.182:5060 —\> \<-------------\> **EDIT** \*\*\*\*\*\* I made the following changes, but still cannot get the sip phone to register. When I take the secret off, the IP shows up, but the device is still unreachable. It looks like the ports are different, despite what I set them up to port 5060. I added this to my sip and reloaded (not shown here) and still have trouble with registering. help! **_SIP_** \*\*\*\* [general] context = unauthenticated language = en allowguest = no srvlookup = yes udpbindaddr = 0.0.0.0 tcpenable = no pedantic = no host = dynamic registertimeout = 20 ; allowoverlap = no ; bindport = 5060 ; qualify = yes ; nat = force\_rport, comedia disallow= all allow = ulaw allow = alaw allow = gsm allow = g729 [nx4] ; call out deny = 0.0.0.0/0.0.0.0 type = friend host = dynamic context = incoming dtmfmode = inband ; changed from rfc… and worked username = nx4 ; second part of sip peers --\> channel/username fromuser = nx4 user = nx4 outboundproxy = xxx.x1.xx8.x7x secret = xxxxxxxx canreinvite = no nat = yes transport = udp permit = 0.0.0.0/0.0.0.0 callcounter = yes faxdetect = no disallow = all allow = ulaw **_Sip show peers_** \* Really destroying SIP dialog ‘70c72cc928c9254a313579c64ed53d04@ xxx.x1.xx8.x7x:5060’ Method: OPTIONS asterisk\*CLI\> sip show peers Name/username Host Dyn Forcerport ACL Port Status 1001/1001 (Unspecified) D N 0 UNKNOWN kbeezy/x0xxxx000 xx.xx.xx0.xx N 5060 OK (55 ms) nx4/nx4 (Unspecified) D N A 0 UNKNOWN office-phone (Unspecified) D N 0 UNKNOWN zoiper/zoiper [xx.xx.x7.xxx](http://xx.xx.x7.xxx) D 5060 OK (40 ms) 5 sip peers [Monitored: 2 online, 3 offline Unmonitored: 0 online, 0 offline] Really destroying SIP dialog ‘a0bbbefba59005b27993cb0f4a957ee6@24.90.77.182’ Method: REGISTER **_Sip debug_** \*\*\*\* Retransmitting #3 (NAT) to [xx.xx.xx.xxx:58078](http://xx.xx.xx.xxx:58078): OPTIONS sip:nx4@xx.xx.xx.xxx:58078;transport=udp SIP/2.0 Via: SIP/2.0/UDP xxx.x1.xx8.x7x:5060;branch=z9hG4bK2185d521;rport Max-Forwards: 70 From: “asterisk” sip:asterisk@xxx.x1.xx8.x7x;tag=as032e6299 To: sip:nx4@xx.xx.xx.xxx:58078;transport=udp Contact: sip:asterisk@xxx.x1.xx8.x7x:5060 Call-ID: [68192523592fa2f0161ad7037f02a62a@xxx.x1.xx8.x](mailto:68192523592fa2f0161ad7037f02a62a@xxx.x1.xx8.x)7x:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1.1 Date: Tue, 23 Jul 2013 22:32:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 * * * Retransmitting #4 (NAT) to [xx.xx.xx.xxx:58078](http://xx.xx.xx.xxx:58078): OPTIONS sip:nx4@xx.xx.xx.xxx:58078;transport=udp SIP/2.0 Via: SIP/2.0/UDP xxx.x1.xx8.x7x:5060;branch=z9hG4bK2185d521;rport Max-Forwards: 70 From: “asterisk” sip:asterisk@xxx.x1.xx8.x7x;tag=as032e6299 To: sip:nx4@xx.xx.xx.xxx:58078;transport=udp Contact: sip:asterisk@xxx.x1.xx8.x7x:5060 Call-ID: [68192523592fa2f0161ad7037f02a62a@xxx.x1.xx8.x](mailto:68192523592fa2f0161ad7037f02a62a@xxx.x1.xx8.x)7x:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1.1 Date: Tue, 23 Jul 2013 22:32:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 * * * Really destroying SIP dialog '68192523592fa2f0161ad7037f02a62a@1xxx.x1.xx8.x7x:5060’ Method: OPTIONS Really destroying SIP dialog '1a7a3d03c36ec8bbf28723c76d86e901@xx.xx.xx.xxx’ Method: REGISTER Reliably Transmitting (NAT) to [xx.xx.xx.xxx:58078](http://xx.xx.xx.xxx:58078): OPTIONS sip:nx4@xx.xx.xx.xxx:58078;transport=udp SIP/2.0 Via: SIP/2.0/UDP xxx.x1.xx8.x7x:5060;branch=z9hG4bK43c75f32;rport Max-Forwards: 70 From: “asterisk” sip:asterisk@xxx.x1.xx8.x7x;tag=as3263bd55 To: sip:nx4@xx.xx.xx.xxx:58078;transport=udp Contact: sip:asterisk@xxx.x1.xx8.x7x:5060 Call-ID: [2f8c231e0c7271603f3f93332fdb50db@xxx.x1.xx8.x](mailto:2f8c231e0c7271603f3f93332fdb50db@xxx.x1.xx8.x)7x:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1.1 Date: Tue, 23 Jul 2013 22:32:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 ---
**Author:** ![david55](https://avatars.discourse-cdn.com/v4/letter/d/e19adc/32.png) [@david55](https://community.asterisk.org/u/david55)\ **Post date:** [July 24, 2013, 3:30pm UTC](https://community.asterisk.org/t/sip-configuration-in-asterisk-and-sip-phone/43446/2 "2013-07-24T15:30:19Z")
Never set insecure=invite on devices that may be given toll calling rights. As well as normally being bad practice, allowguest does not work in device sections. The phone should resend the register with authentication data. If it doesn’t, it suggests it doesn’t believe it has a password set (or cannot do basic authentication). ---
**Author:** ![KBennett](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/kbennett/32/146_2.png) [@KBennett](https://community.asterisk.org/u/KBennett)\ **Post date:** [July 25, 2013, 3:46pm UTC](https://community.asterisk.org/t/sip-configuration-in-asterisk-and-sip-phone/43446/3 "2013-07-25T15:46:41Z")
Hi David55, thanks for responding… I took out allowguest and insecure, but I still can’t get the phone to register. I can see that the phone is trying to register (due to the feedback on its interface). I’m still not sure what the problem is. ☹ Can you take another look? Could it be something else that isn’t set correctly? Or is set incorrectly? I’d appreciate it! Here is what I have on the phone itself, the sip.conf, debug and sip show peers: \*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\* settings on actual phone\*\*\*\*\*\*\*\*\*\*\*\* Username: nx4 password: mypass server: my.asterisk.box outbound proxy address: my.asterisk.box port number: 5060 transportation type: UDP Send keep-alive: automatic display name: authentication username: **_sip conf_** \*\*\* [general] context = unauthenticated language = en allowguest = no srvlookup = yes udpbindaddr = 0.0.0.0 tcpenable = no pedantic = no host = dynamic registertimeout = 20 ; allowoverlap = no ; bindport = 5060 ; qualify = yes ; nat = force\_rport, comedia disallow= all allow = ulaw allow = alaw allow = gsm allow = g729 [nx4] deny = 0.0.0.0/0.0.0.0 type = friend host = dynamic context = incoming dtmfmode = inband username = nx4 ; second part of sip peers --\> channel/username fromuser = nx4 user = nx4 outboundproxy = [xxx.xx.xxx.xxx](http://xxx.xx.xxx.xxx) ; asterisk box secret =mypass nat = yes transport = udp permit = 0.0.0.0/0.0.0.0 callcounter = yes faxdetect = no disallow = all allow = ulaw bindport =5060 **_ship show peers_** Really destroying SIP dialog ‘1edaf14b7bd613b16bb168941aa7891a@198.61.168.179:5060’ Method: OPTIONS asterisk_CLI\> sip show peers Name/username Host Dyn Forcerport ACL Port Status 1001/1001 (Unspecified) D N 0 UNKNOWN kbeezy/xxxxxxxxx xx.xx.xxx.xx N 5060 OK (56 ms) nx4/nx4 (Unspecified) D N A 0 UNKNOWN office-phone (Unspecified) D N 0 UNKNOWN zoiper/zoiper [xx.xx.xx.xxx](http://xx.xx.xx.xxx) D 5060 OK (46 ms) 5 sip peers [Monitored: 2 online, 3 offline Unmonitored: 0 online, 0 offline] asterisk_CLI\> ## **_sip debug output_** \*\*\*\* Reliably Transmitting (NAT) to xx.xx.xxx.xx :5060: OPTIONS sip:xx.xx.xxx.xx SIP/2.0 Via: SIP/2.0/UDP [xxx.xx.xxx.xxx:5060](http://xxx.xx.xxx.xxx:5060);branch=z9hG4bK20f3ad4a;rport Max-Forwards: 70 From: “asterisk” sip:asterisk@xxx.xx.xxx.xxx;tag=as35b1d57f To: sip:xx.xx.xxx.xx Contact: sip:asterisk@xxx.xx.xxx.xxx:5060 Call-ID: [12652ac42817c7205206be18601df5ce@xxx.xx.xxx.xxx](mailto:12652ac42817c7205206be18601df5ce@xxx.xx.xxx.xxx):5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1.1 Date: Thu, 25 Jul 2013 15:20:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 \<-------------\> — (11 headers 0 lines) — Really destroying SIP dialog '12652ac42817c7205206be18601df5ce@xxx.xx.xxx.xxx:5060’ Method: OPTIONS Really destroying SIP dialog '305cf6cf6ea54ff642d864c5ebbf9bb3@xx.xx.xxx.xx’ Method: REGISTER ---
**Author:** ![david55](https://avatars.discourse-cdn.com/v4/letter/d/e19adc/32.png) [@david55](https://community.asterisk.org/u/david55)\ **Post date:** [July 25, 2013, 4:58pm UTC](https://community.asterisk.org/t/sip-configuration-in-asterisk-and-sip-phone/43446/4 "2013-07-25T16:58:57Z")
There is no register attempt in the current debug output. The previous one showed an abandoned one. It should go REGISTER, 401, REGISTER, 200, but stopped after the 401. That means that either you provided incomplete logs, or the phone realised that it was unable to authenticate, and gave up. ---
**Author:** ![KBennett](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/kbennett/32/146_2.png) [@KBennett](https://community.asterisk.org/u/KBennett)\ **Post date:** [July 25, 2013, 5:22pm UTC](https://community.asterisk.org/t/sip-configuration-in-asterisk-and-sip-phone/43446/5 "2013-07-25T17:22:03Z")
My apologies! Zoiper and IPKall work fine, so far. Nx4 is the one I’m having issues with. Here’s a full one: \<— SIP read from UDP:xx.xx.xx.xxx:57753 —\> REGISTER sip:xxx.xx.xxx.xxx SIP/2.0 Call-ID: [50d6618f51cfb430c1e047a0f470fd8f@xx.xx.xx.xxx](mailto:50d6618f51cfb430c1e047a0f470fd8f@xx.xx.xx.xxx) CSeq: 6386 REGISTER From: “nx4” sip:nx4@xxx.xx.xxx.xxx;tag=2778056207 To: “nx4” sip:nx4@xxx.xx.xxx.xxx Via: SIP/2.0/UDP [xx.xx.xx.xxx:57753](http://xx.xx.xx.xxx:57753);branch=z9hG4bK225c148413dd14575ca6993ad7a5484d313631;rport Max-Forwards: 70 User-Agent: SIPAUA/0.1.001 Contact: \* Expires: 0 Content-Length: 0 \<-------------\> — (11 headers 0 lines) — Sending to [xx.xx.xx.xxx:57753](http://xx.xx.xx.xxx:57753) (no NAT) \<— Transmitting (NAT) to [xx.xx.xx.xxx:57753](http://xx.xx.xx.xxx:57753) —\> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP [xx.xx.xx.xxx:57753](http://xx.xx.xx.xxx:57753);branch=z9hG4bK225c148413dd14575ca6993ad7a5484d313631;[received=xx.xx.xx.xxx](http://received=xx.xx.xx.xxx);rport=57753 From: “nx4” sip:nx4@xxx.xx.xxx.xxx;tag=2778056207 To: “nx4” sip:nx4@xxx.xx.xxx.xxx;tag=as02009c6f Call-ID: [50d6618f51cfb430c1e047a0f470fd8f@xx.xx.xx.xxx](mailto:50d6618f51cfb430c1e047a0f470fd8f@xx.xx.xx.xxx) CSeq: 6386 REGISTER Server: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="227edcda" Content-Length: 0 \<------------\> Scheduling destruction of SIP dialog '50d6618f51cfb430c1e047a0f470fd8f@xx.xx.xx.xxx’ in 32000 ms (Method: REGISTER) \<— SIP read from UDP:xx.xx.xx.xxx:5060 —\> \<-------------\> Reliably Transmitting (NAT) toxx.xx.xxx.xx:5060: OPTIONS sip:xx.xx.xxx.xx SIP/2.0 Via: SIP/2.0/UDP [xxx.xx.xxx.xxx:5060](http://xxx.xx.xxx.xxx:5060);branch=z9hG4bK192b5810;rport Max-Forwards: 70 From: “asterisk” sip:asterisk@xxx.xx.xxx.xxx;tag=as701fca91 To: sip:xx.xx.xxx.xx Contact: sip:asterisk@xxx.xx.xxx.xxx:5060 Call-ID: [77208fab2f871c5d04e1382768d71bda@xxx.xx.xxx.xxx](mailto:77208fab2f871c5d04e1382768d71bda@xxx.xx.xxx.xxx):5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1.1 Date: Thu, 25 Jul 2013 17:15:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 * * * Reliably Transmitting (no NAT) to [xx.xx.xx.xxx:5060](http://xx.xx.xx.xxx:5060): OPTIONS sip:zoiper@xx.xx.xx.xxx:5060;rinstance=7bc586e32d632a74;transport=UDP SIP/2.0 Via: SIP/2.0/UDP [xxx.xx.xxx.xxx:5060](http://xxx.xx.xxx.xxx:5060);branch=z9hG4bK21403b8c Max-Forwards: 70 From: “asterisk” sip:asterisk@xxx.xx.xxx.xxx;tag=as56d43c49 To: sip:zoiper@xx.xx.xx.xxx:5060;rinstance=7bc586e32d632a74;transport=UDP Contact: sip:asterisk@xxx.xx.xxx.xxx:5060 Call-ID: [2bf05a3a6b9b931f5678e44d342d8dfd@xxx.xx.xxx.xxx](mailto:2bf05a3a6b9b931f5678e44d342d8dfd@xxx.xx.xxx.xxx):5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1.1 Date: Thu, 25 Jul 2013 17:15:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 * * * \<— SIP read from UDP:xx.xx.xxx.xx:5060 —\> SIP/2.0 404 Not Found Via: SIP/2.0/UDP [xxx.xx.xxx.xxx:5060](http://xxx.xx.xxx.xxx:5060);branch=z9hG4bK192b5810;[received=xxx.xx.xxx.xxx](http://received=xxx.xx.xxx.xxx);rport=5060 From: “asterisk” sip:asterisk@xxx.xx.xxx.xxx;tag=as701fca91 To: sip:xx.xx.xxx.xx;tag=as3505656d Call-ID: [77208fab2f871c5d04e1382768d71bda@xxx.xx.xxx.xxx](mailto:77208fab2f871c5d04e1382768d71bda@xxx.xx.xxx.xxx):5060 CSeq: 102 OPTIONS User-Agent: IPKall Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Accept: application/sdp Content-Length: 0 \<-------------\> — (11 headers 0 lines) — Really destroying SIP dialog '77208fab2f871c5d04e1382768d71bda@xxx.xx.xxx.xxx:5060’ Method: OPTIONS \<— SIP read from UDP:xx.xx.xx.xxx:5060 —\> SIP/2.0 200 OK Via: SIP/2.0/UDP [xxx.xx.xxx.xxx:5060](http://xxx.xx.xxx.xxx:5060);branch=z9hG4bK21403b8c Contact: sip:xx.xx.xx.xxx:5060 To: sip:zoiper@xx.xx.xx.xxx:5060;rinstance=7bc586e32d632a74;transport=UDP;tag=b7a22458 From: "asterisk"sip:asterisk@xxx.xx.xxx.xxx;tag=as56d43c49 Call-ID: [2bf05a3a6b9b931f5678e44d342d8dfd@xxx.xx.xxx.xxx](mailto:2bf05a3a6b9b931f5678e44d342d8dfd@xxx.xx.xxx.xxx):5060 CSeq: 102 OPTIONS Accept: application/sdp, application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri User-Agent: Zoiper rev.11619 Allow-Events: presence, kpml Content-Length: 0 \<-------------\> — (14 headers 0 lines) — Really destroying SIP dialog '2bf05a3a6b9b931f5678e44d342d8dfd@xxx.xx.xxx.xxx:5060’ Method: OPTIONS \<— SIP read from UDP:xx.xx.xx.xxx:57753 —\> REGISTER sip:xxx.xx.xxx.xxx SIP/2.0 Call-ID: [f1ad591c984e6c1fd7e79d09a9596748@xx.xx.xx.xxx](mailto:f1ad591c984e6c1fd7e79d09a9596748@xx.xx.xx.xxx) CSeq: 8575 REGISTER From: “nx4” sip:nx4@xxx.xx.xxx.xxx;tag=96168691 To: “nx4” sip:nx4@xxx.xx.xxx.xxx Via: SIP/2.0/UDP [xx.xx.xx.xxx:57753](http://xx.xx.xx.xxx:57753);branch=z9hG4bKb63f244a27efe1d3ff6996a361ffac24313631;rport Max-Forwards: 70 User-Agent: SIPAUA/0.1.001 Contact: “nx4” sip:nx4@xx.xx.xx.xxx:57753;transport=udp Expires: 3600 Content-Length: 0 \<-------------\> — (11 headers 0 lines) — Sending to [xx.xx.xx.xxx:57753](http://xx.xx.xx.xxx:57753) (no NAT) \<— Transmitting (NAT) to [xx.xx.xx.xxx:57753](http://xx.xx.xx.xxx:57753) —\> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP [xx.xx.xx.xxx:57753](http://xx.xx.xx.xxx:57753);branch=z9hG4bKb63f244a27efe1d3ff6996a361ffac24313631;[received=xx.xx.xx.xxx](http://received=xx.xx.xx.xxx);rport=57753 From: “nx4” sip:nx4@xxx.xx.xxx.xxx;tag=96168691 To: “nx4” sip:nx4@xxx.xx.xxx.xxx;tag=as6a01b7cd Call-ID: [f1ad591c984e6c1fd7e79d09a9596748@xx.xx.xx.xxx](mailto:f1ad591c984e6c1fd7e79d09a9596748@xx.xx.xx.xxx) CSeq: 8575 REGISTER Server: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="5b605158" Content-Length: 0 \<------------\> Scheduling destruction of SIP dialog 'f1ad591c984e6c1fd7e79d09a9596748@xx.xx.xx.xxx’ in 32000 ms (Method: REGISTER) Really destroying SIP dialog '50d6618f51cfb430c1e047a0f470fd8f@xx.xx.xx.xxx’ Method: REGISTER \<— SIP read from UDP:xx.xx.xx.xxx:5060 —\> \<-------------\> \<— SIP read from UDP:xx.xx.xx.xxx:57753 —\> REGISTER sip:xxx.xx.xxx.xxx SIP/2.0 Call-ID: [10bf44298552e260bc84d1d276c99788@xx.xx.xx.xxx](mailto:10bf44298552e260bc84d1d276c99788@xx.xx.xx.xxx) CSeq: 7849 REGISTER From: “nx4” sip:nx4@xxx.xx.xxx.xxx;tag=4020576262 To: “nx4” sip:nx4@xxx.xx.xxx.xxx Via: SIP/2.0/UDP [xx.xx.xx.xxx:57753](http://xx.xx.xx.xxx:57753);branch=z9hG4bKd474bf3983b685f00d33eae6216a04c2313631;rport Max-Forwards: 70 User-Agent: SIPAUA/0.1.001 Contact: “nx4” sip:nx4@xx.xx.xx.xxx:57753;transport=udp Expires: 3600 Content-Length: 0 \<-------------\> — (11 headers 0 lines) — Sending to [xx.xx.xx.xxx:57753](http://xx.xx.xx.xxx:57753) (no NAT) \<— Transmitting (NAT) to [xx.xx.xx.xxx:57753](http://xx.xx.xx.xxx:57753) —\> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP [xx.xx.xx.xxx:57753](http://xx.xx.xx.xxx:57753);branch=z9hG4bKd474bf3983b685f00d33eae6216a04c2313631;[received=xx.xx.xx.xxx](http://received=xx.xx.xx.xxx);rport=57753 From: “nx4” sip:nx4@xxx.xx.xxx.xxx;tag=4020576262 To: “nx4” sip:nx4@xxx.xx.xxx.xxx;tag=as245ea5e6 Call-ID: [10bf44298552e260bc84d1d276c99788@xx.xx.xx.xxx](mailto:10bf44298552e260bc84d1d276c99788@xx.xx.xx.xxx) CSeq: 7849 REGISTER Server: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="6357a9e2" Content-Length: 0 \<------------\> Scheduling destruction of SIP dialog '10bf44298552e260bc84d1d276c99788@xx.xx.xx.xxx’ in 32000 ms (Method: REGISTER) \<— SIP read from UDP:xx.xx.xx.xxx:57753 —\> REGISTER sip:xxx.xx.xxx.xxx SIP/2.0 Call-ID: [10bf44298552e260bc84d1d276c99788@xx.xx.xx.xxx](mailto:10bf44298552e260bc84d1d276c99788@xx.xx.xx.xxx) CSeq: 7849 REGISTER From: “nx4” sip:nx4@xxx.xx.xxx.xxx;tag=4020576262 To: “nx4” sip:nx4@xxx.xx.xxx.xxx Via: SIP/2.0/UDP [xx.xx.xx.xxx:57753](http://xx.xx.xx.xxx:57753);branch=z9hG4bKd474bf3983b685f00d33eae6216a04c2313631;rport Max-Forwards: 70 User-Agent: SIPAUA/0.1.001 Contact: “nx4” sip:nx4@xx.xx.xx.xxx:57753;transport=udp Expires: 3600 Content-Length: 0 \<-------------\> — (11 headers 0 lines) — Sending to [xx.xx.xx.xxx:57753](http://xx.xx.xx.xxx:57753) (NAT) \<— Transmitting (NAT) to [xx.xx.xx.xxx:57753](http://xx.xx.xx.xxx:57753) —\> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP [xx.xx.xx.xxx:57753](http://xx.xx.xx.xxx:57753);branch=z9hG4bKd474bf3983b685f00d33eae6216a04c2313631;[received=xx.xx.xx.xxx](http://received=xx.xx.xx.xxx);rport=57753 From: “nx4” sip:nx4@xxx.xx.xxx.xxx;tag=4020576262 To: “nx4” sip:nx4@xxx.xx.xxx.xxx;tag=as245ea5e6 Call-ID: [10bf44298552e260bc84d1d276c99788@xx.xx.xx.xxx](mailto:10bf44298552e260bc84d1d276c99788@xx.xx.xx.xxx) CSeq: 7849 REGISTER Server: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="6357a9e2" Content-Length: 0 ---
**Author:** ![david55](https://avatars.discourse-cdn.com/v4/letter/d/e19adc/32.png) [@david55](https://community.asterisk.org/u/david55)\ **Post date:** [July 25, 2013, 6:18pm UTC](https://community.asterisk.org/t/sip-configuration-in-asterisk-and-sip-phone/43446/6 "2013-07-25T18:18:36Z")
As well as the problems that I already mentioned, in relation to the phone, [xx.xx.77.xxx:57120](http://xx.xx.77.xxx:57120) may not be routable from the Asterisk side. In all cases, the phone is not making an appropriate response to the 401. ---
**Author:** ![KBennett](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/kbennett/32/146_2.png) [@KBennett](https://community.asterisk.org/u/KBennett)\ **Post date:** [July 25, 2013, 6:30pm UTC](https://community.asterisk.org/t/sip-configuration-in-asterisk-and-sip-phone/43446/7 "2013-07-25T18:30:41Z")
Hi David55, Thanks again for taking a look. Would a different SIP app on the phone help? I’m currently trying SIPDroid versus the native SIP settings via the phone. Or, does this mean that the phone won’t work at all? Can anyone suggest some work-arounds? Again, thanks for the time and the suggestions! I appreciate it! ---
**Author:** ![KBennett](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/kbennett/32/146_2.png) [@KBennett](https://community.asterisk.org/u/KBennett)\ **Post date:** [July 25, 2013, 10:43pm UTC](https://community.asterisk.org/t/sip-configuration-in-asterisk-and-sip-phone/43446/8 "2013-07-25T22:43:50Z")
Well… I changed “secret” to “remotesecret”, reloaded my sip settings and checked my peers. SipDroid only works with [pbxes.org](http://pbxes.org) to tunnel VOIP calls. The Zoiper VoIP app works great! On to the next step! Wooo-who! 8) ---
**Author:** ![david55](https://avatars.discourse-cdn.com/v4/letter/d/e19adc/32.png) [@david55](https://community.asterisk.org/u/david55)\ **Post date:** [July 26, 2013, 7:04am UTC](https://community.asterisk.org/t/sip-configuration-in-asterisk-and-sip-phone/43446/9 "2013-07-26T07:04:36Z")
Using only remotesecret for a peer that is registering with you effectively disables authentication for that registration!