# Are snom phones fully compatible with asterisk/trixbox **URL:** \ **Category:** Asterisk Support\ **Created:** [April 10, 2012, 4:21pm UTC](https://community.asterisk.org/t/are-snom-phones-fully-compatible-with-asterisk-trixbox/38868 "2012-04-10T16:21:43Z")\ **Posts on this page:** 11\ **Page:** 1
**Author:** ![system](https://global.discourse-cdn.com/asterisk/original/2X/4/475d0c889911cfb84ac4881ad6df8b523e3f9105.png) [@system](https://community.asterisk.org/u/system)\ **Post date:** [April 10, 2012, 4:21pm UTC](https://community.asterisk.org/t/are-snom-phones-fully-compatible-with-asterisk-trixbox/38868/1 "2012-04-10T16:21:43Z")
HI all, we have version 2.8.0.4 & Asterisk 1.6.0.26…and most of our phones are snom 360 and work perfectly fine but for some reason snom m9’s do not want to play…is there some special setting i have to switch on?? Currently as it stands you get a busy tone on the phone when i try to make a call… I have looked at the trixbox panel and the phone is registered, but when i log on to asterisk with -vvvrrr switch it doesn’t even see the phone making the call…i make a call to the snom m9 and it says it is busy… I have tried every thing…i know the extension is working that is configured on trixbox as i configured a snom 360 phone and it is fine. try it on a snom m9 and it doesn’t work… any ideas?? is it Compatible?? am i missing a special setting?? ---
**Author:** ![tomdemoor](https://avatars.discourse-cdn.com/v4/letter/t/82dd89/32.png) [@tomdemoor](https://community.asterisk.org/u/tomdemoor)\ **Post date:** [April 10, 2012, 4:47pm UTC](https://community.asterisk.org/t/are-snom-phones-fully-compatible-with-asterisk-trixbox/38868/2 "2012-04-10T16:47:34Z")
hi, we have snom M9’s with asterisk (even 1.4) and no problems. I have no knowledge of Snom M9’s not being comptabile with asterisk. And when you do: sip show peer $snom9 account? what do you get then? Tom ---
**Author:** ![system](https://global.discourse-cdn.com/asterisk/original/2X/4/475d0c889911cfb84ac4881ad6df8b523e3f9105.png) [@system](https://community.asterisk.org/u/system)\ **Post date:** [April 11, 2012, 1:58pm UTC](https://community.asterisk.org/t/are-snom-phones-fully-compatible-with-asterisk-trixbox/38868/3 "2012-04-11T13:58:52Z")
hi there thanks for the reply, it says peep $snom9 not found? any ideas?? this is driving me nuts!! ---
**Author:** ![omid.mohajerani](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/omid.mohajerani/32/32_2.png) [@omid.mohajerani](https://community.asterisk.org/u/omid.mohajerani)\ **Post date:** [April 11, 2012, 3:04pm UTC](https://community.asterisk.org/t/are-snom-phones-fully-compatible-with-asterisk-trixbox/38868/4 "2012-04-11T15:04:14Z")
replace $snom9 with the extension number you assign for snom extension !! Snom m9 in working without any problem with asterisk . It has a very good manual you can find their website . I think you didnt register pin of handsets . ---
**Author:** ![system](https://global.discourse-cdn.com/asterisk/original/2X/4/475d0c889911cfb84ac4881ad6df8b523e3f9105.png) [@system](https://community.asterisk.org/u/system)\ **Post date:** [April 11, 2012, 4:08pm UTC](https://community.asterisk.org/t/are-snom-phones-fully-compatible-with-asterisk-trixbox/38868/5 "2012-04-11T16:08:25Z")
ok here it goes this is using the sip show peer 6816 command does this tell you anything that is wrong?? - Name : 6816 Secret : MD5Secret : Context : from-internal Subscr.Cont. : Language : AMA flags : Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup : 1 Pickupgroup : 1 Mailbox : 6816@default VM Extension : \*97 LastMsgsSent : 32767/65535 Call limit : 50 Dynamic : Yes Callerid : “device” \<6816\> MaxCallBR : 384 kbps Expire : 3566 Insecure : no Nat : Always ACL : Yes T.38 support : No T.38 EC mode : Unknown T.38 MaxDtgrm: -1 CanReinvite : No PromiscRedir : No User=Phone : No Video Support: Yes Text Support : No Ign SDP ver : No Trust RPID : No Send RPID : No Subscriptions: Yes Overlap dial : Yes DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : Addr-\>IP : 10.50.0.144 Port 3731 Defaddr-\>IP : 0.0.0.0 Port 5060 Transport : UDP Def. Username: 6816 SIP Options : (none) Codecs : 0x4 (ulaw) Codec Order : (ulaw:20) Auto-Framing : No 100 on REG : No Status : OK (6 ms) Useragent : snom-m9/9.5.14-a Reg. Contact : sip:6816@10.50.0.144:3731;transport=udp;line=vol7di Qualify Freq : 60000 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs ---
**Author:** ![omid.mohajerani](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/omid.mohajerani/32/32_2.png) [@omid.mohajerani](https://community.asterisk.org/u/omid.mohajerani)\ **Post date:** [April 11, 2012, 7:51pm UTC](https://community.asterisk.org/t/are-snom-phones-fully-compatible-with-asterisk-trixbox/38868/6 "2012-04-11T19:51:16Z")
It seems it is successfully registered . you should put your asterisk console log here when you are calling 6816 ---
**Author:** ![system](https://global.discourse-cdn.com/asterisk/original/2X/4/475d0c889911cfb84ac4881ad6df8b523e3f9105.png) [@system](https://community.asterisk.org/u/system)\ **Post date:** [April 12, 2012, 1:24pm UTC](https://community.asterisk.org/t/are-snom-phones-fully-compatible-with-asterisk-trixbox/38868/7 "2012-04-12T13:24:00Z")
thanks for you replies feel like we are getting somewhere here is the asterisk log from when i call from 6811 (snom 360) to 6816 (snom m9) – Executing [6816@from-internal:1] Macro(“SIP/6811-00001e6f”, “exten-vm,novm,6816”) in new stack – Executing [s@macro-exten-vm:1] Macro(“SIP/6811-00001e6f”, “user-callerid”) in new stack – Executing [s@macro-user-callerid:1] Set(“SIP/6811-00001e6f”, “AMPUSER=6811”) in new stack – Executing [s@macro-user-callerid:2] GotoIf(“SIP/6811-00001e6f”, “0?report”) in new stack – Executing [s@macro-user-callerid:3] ExecIf(“SIP/6811-00001e6f”, “1?Set(REALCALLERIDNUM=6811)”) in new stack – Executing [s@macro-user-callerid:4] Set(“SIP/6811-00001e6f”, “AMPUSER=6811”) in new stack – Executing [s@macro-user-callerid:5] Set(“SIP/6811-00001e6f”, “AMPUSERCIDNAME=JH”) in new stack – Executing [s@macro-user-callerid:6] GotoIf(“SIP/6811-00001e6f”, “0?report”) in new stack – Executing [s@macro-user-callerid:7] Set(“SIP/6811-00001e6f”, “AMPUSERCID=6811”) in new stack – Executing [s@macro-user-callerid:8] Set(“SIP/6811-00001e6f”, “CALLERID(all)=“JH” \<6811\>”) in new stack – Executing [s@macro-user-callerid:9] ExecIf(“SIP/6811-00001e6f”, “0?Set(CHANNEL(language)=)”) in new stack – Executing [s@macro-user-callerid:10] GotoIf(“SIP/6811-00001e6f”, “0?continue”) in new stack – Executing [s@macro-user-callerid:11] Set(“SIP/6811-00001e6f”, “\_\_TTL=64”) in new stack – Executing [s@macro-user-callerid:12] GotoIf(“SIP/6811-00001e6f”, “1?continue”) in new stack – Goto (macro-user-callerid,s,19) – Executing [s@macro-user-callerid:19] NoOp(“SIP/6811-00001e6f”, “Using CallerID “JH” \<6811\>”) in new stack – Executing [s@macro-exten-vm:2] Set(“SIP/6811-00001e6f”, “RingGroupMethod=none”) in new stack – Executing [s@macro-exten-vm:3] Set(“SIP/6811-00001e6f”, “VMBOX=novm”) in new stack – Executing [s@macro-exten-vm:4] Set(“SIP/6811-00001e6f”, “EXTTOCALL=6816”) in new stack – Executing [s@macro-exten-vm:5] Set(“SIP/6811-00001e6f”, “CFUEXT=”) in new stack – Executing [s@macro-exten-vm:6] Set(“SIP/6811-00001e6f”, “CFBEXT=”) in new stack – Executing [s@macro-exten-vm:7] Set(“SIP/6811-00001e6f”, “RT=”"") in new stack – Executing [s@macro-exten-vm:8] Macro(“SIP/6811-00001e6f”, “record-enable,6816,IN”) in new stack – Executing [s@macro-record-enable:1] GotoIf(“SIP/6811-00001e6f”, “1?check”) in new stack – Goto (macro-record-enable,s,4) – Executing [s@macro-record-enable:4] AGI(“SIP/6811-00001e6f”, “recordingcheck,20120412-141526,1334236526.12914”) in new stack – Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck,20120412-141526,1334236526.12914: Inbound recording not enabled – \AGI Script recordingcheck completed, returning 0 – Executing [s@macro-record-enable:5] MacroExit(“SIP/6811-00001e6f”, “”) in new stack – Executing [s@macro-exten-vm:9] Macro(“SIP/6811-00001e6f”, “dial,”",trw,6816") in new stack – Executing [s@macro-dial:1] GotoIf(“SIP/6811-00001e6f”, “1?dial”) in new stack – Goto (macro-dial,s,3) – Executing [s@macro-dial:3] AGI(“SIP/6811-00001e6f”, “dialparties.agi”) in new stack – Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi dialparties.agi: Starting New Dialparties.agi dialparties.agi: Caller ID name is ‘JH’ number is ‘6811’ \> dialparties.agi: USE\_CONFIRMATION: ‘FALSE’ \> dialparties.agi: RINGGROUP\_INDEX: '' dialparties.agi: Methodology of ring is ‘none’ – dialparties.agi: Added extension 6816 to extension map – dialparties.agi: Extension 6816 cf is disabled – dialparties.agi: Extension 6816 do not disturb is disabled \> dialparties.agi: extnum 6816 has: cw: 1; hascfb: 0 [] hascfu: 0 [] dialparties.agi: EXTENSION\_STATE: 0 (NOT\_INUSE) – dialparties.agi: dbset CALLTRACE/6816 to 6811 – dialparties.agi: Filtered ARG3: 6816 – \AGI Script dialparties.agi completed, returning 0 – Executing [s@macro-dial:7] Dial(“SIP/6811-00001e6f”, “SIP/6816,”",trw") in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 == Using SIP VRTP TOS bits 136 == Using SIP VRTP CoS mark 6 – Called 6816 == Extension Changed 6816[ext-local] new state Ringing for Notify User 6833 – Got SIP response 410 “Gone” back from 10.50.0.144 – SIP/6816-00001e70 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) – Executing [s@macro-dial:8] Set(“SIP/6811-00001e6f”, “DIALSTATUS=CONGESTION”) in new stack – Executing [s@macro-dial:9] GosubIf(“SIP/6811-00001e6f”, “0?CONGESTION,1”) in new stack – Executing [s@macro-exten-vm:10] GotoIf(“SIP/6811-00001e6f”, “0?exit,return”) in new stack – Executing [s@macro-exten-vm:11] Set(“SIP/6811-00001e6f”, “SV\_DIALSTATUS=CONGESTION”) in new stack – Executing [s@macro-exten-vm:12] GosubIf(“SIP/6811-00001e6f”, “0?docfu,1”) in new stack – Executing [s@macro-exten-vm:13] GosubIf(“SIP/6811-00001e6f”, “0?docfb,1”) in new stack – Executing [s@macro-exten-vm:14] Set(“SIP/6811-00001e6f”, “DIALSTATUS=CONGESTION”) in new stack – Executing [s@macro-exten-vm:15] NoOp(“SIP/6811-00001e6f”, “Voicemail is ‘novm’”) in new stack – Executing [s@macro-exten-vm:16] GotoIf(“SIP/6811-00001e6f”, “1?s-CONGESTION,1”) in new stack – Goto (macro-exten-vm,s-CONGESTION,1) – Executing [s-CONGESTION@macro-exten-vm:1] NoOp(“SIP/6811-00001e6f”, "IVR\_RETVM: IVR\_CONTEXT: ") in new stack – Executing [s-CONGESTION@macro-exten-vm:2] GotoIf(“SIP/6811-00001e6f”, “0?exit,1”) in new stack – Executing [s-CONGESTION@macro-exten-vm:3] PlayTones(“SIP/6811-00001e6f”, “congestion”) in new stack – Executing [s-CONGESTION@macro-exten-vm:4] Congestion(“SIP/6811-00001e6f”, “10”) in new stack == Spawn extension (macro-exten-vm, s-CONGESTION, 4) exited non-zero on ‘SIP/6811-00001e6f’ in macro ‘exten-vm’ == Spawn extension (from-internal, 6816, 1) exited non-zero on ‘SIP/6811-00001e6f’ – Executing [h@from-internal:1] Macro(“SIP/6811-00001e6f”, “hangupcall”) in new stack – Executing [s@macro-hangupcall:1] GotoIf(“SIP/6811-00001e6f”, “1?skiprg”) in new stack – Goto (macro-hangupcall,s,4) – Executing [s@macro-hangupcall:4] GotoIf(“SIP/6811-00001e6f”, “1?skipblkvm”) in new stack – Goto (macro-hangupcall,s,7) – Executing [s@macro-hangupcall:7] GotoIf(“SIP/6811-00001e6f”, “1?theend”) in new stack – Goto (macro-hangupcall,s,9) – Executing [s@macro-hangupcall:9] Hangup(“SIP/6811-00001e6f”, “”) in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on ‘SIP/6811-00001e6f’ in macro ‘hangupcall’ == Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/6811-00001e6f’ == Extension Changed 6816[ext-local] new state Idle for Notify User 6833 (queued) == Extension Changed 6816[from-internal] new state Idle for Notify User 6833 pbx\*CLI\> ---
**Author:** ![omid.mohajerani](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/omid.mohajerani/32/32_2.png) [@omid.mohajerani](https://community.asterisk.org/u/omid.mohajerani)\ **Post date:** [April 12, 2012, 2:18pm UTC](https://community.asterisk.org/t/are-snom-phones-fully-compatible-with-asterisk-trixbox/38868/8 "2012-04-12T14:18:31Z")
== Extension Changed 6816[ext-local] new state Ringing for Notify User 6833 – Got SIP response 410 “Gone” back from 10.50.0.144 – SIP/6816-00001e70 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) Do you have extension 6833 ? what is it ? what is the ip address 10.50.0.144 ? Is it 6816 extension ip address or you asterisk ip address ? you are using GUI and that makes it difficult to find the problem . you can have more detailed debug by using command : sip set debug ip 10.50.0.144 ( if it is the ip address of the snom extension ) ---
**Author:** ![miguellinux](https://avatars.discourse-cdn.com/v4/letter/m/7bcc69/32.png) [@miguellinux](https://community.asterisk.org/u/miguellinux)\ **Post date:** [April 12, 2012, 2:36pm UTC](https://community.asterisk.org/t/are-snom-phones-fully-compatible-with-asterisk-trixbox/38868/9 "2012-04-12T14:36:53Z")
The SNOM m9 have a bug with Asteirsk… the trick is: Change at SIP Settings for Identitity 1 (or the one you need) Server type= TAHI project That is… it solves the inbound call problem with Asterisk. Goog Luck! Miguellinux ---
**Author:** ![system](https://global.discourse-cdn.com/asterisk/original/2X/4/475d0c889911cfb84ac4881ad6df8b523e3f9105.png) [@system](https://community.asterisk.org/u/system)\ **Post date:** [April 13, 2012, 10:43am UTC](https://community.asterisk.org/t/are-snom-phones-fully-compatible-with-asterisk-trixbox/38868/10 "2012-04-13T10:43:45Z")
thanks again guys for this… ok something strange is going on…change it to tahi project samething…turn debugging on for 10.50.0.144…and when i make an outgoing call nothing is registered on the system even with debugging on…but of course if i call the snom m9 from 6811 phone you get a log…so surely the outgoing call problem is the key…?? oh 6833 is just another extension accidently captured. – DAHDI/1-1 answered SIP/6802-00001eeb Reliably Transmitting (NAT) to 10.50.0.144:3894: OPTIONS sip:6816@10.50.0.144:3894;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.50.0.10:5060;branch=z9hG4bK5aac9428;rport Max-Forwards: 70 From: “Unknown” sip:Unknown@10.50.0.10;tag=as2d7a0e06 To: sip:6816@10.50.0.144:3894;transport=udp Contact: sip:Unknown@10.50.0.10 Call-ID: [160a884a0c75b2b716c205d40d93d105@10.50.0.10](mailto:160a884a0c75b2b716c205d40d93d105@10.50.0.10) CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78 Date: Fri, 13 Apr 2012 10:36:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 * * * pbx\*CLI\> \<— SIP read from UDP://10.50.0.144:3894 —\> SIP/2.0 200 Ok Via: SIP/2.0/UDP 10.50.0.10:5060;branch=z9hG4bK5aac9428;rport=5060 From: “Unknown” sip:Unknown@10.50.0.10;tag=as2d7a0e06 To: sip:6816@10.50.0.144:3894;transport=udp Call-ID: [160a884a0c75b2b716c205d40d93d105@10.50.0.10](mailto:160a884a0c75b2b716c205d40d93d105@10.50.0.10) CSeq: 102 OPTIONS Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Supported: 100rel, replaces Content-Length: 0 \<-------------\> — (9 headers 0 lines) — Really destroying SIP dialog ‘160a884a0c75b2b716c205d40d93d105@10.50.0.10’ Method: OPTIONS == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 == Using SIP VRTP TOS bits 136 == Using SIP VRTP CoS mark 6 – Executing [6816@from-internal:1] Macro(“SIP/6811-00001eee”, “exten-vm,novm,6816”) in new stack – Executing [s@macro-exten-vm:1] Macro(“SIP/6811-00001eee”, “user-callerid”) in new stack – Executing [s@macro-user-callerid:1] Set(“SIP/6811-00001eee”, “AMPUSER=6811”) in new stack – Executing [s@macro-user-callerid:2] GotoIf(“SIP/6811-00001eee”, “0?report”) in new stack – Executing [s@macro-user-callerid:3] ExecIf(“SIP/6811-00001eee”, “1?Set(REALCALLERIDNUM=6811)”) in new stack – Executing [s@macro-user-callerid:4] Set(“SIP/6811-00001eee”, “AMPUSER=6811”) in new stack – Executing [s@macro-user-callerid:5] Set(“SIP/6811-00001eee”, “AMPUSERCIDNAME=JH”) in new stack – Executing [s@macro-user-callerid:6] GotoIf(“SIP/6811-00001eee”, “0?report”) in new stack – Executing [s@macro-user-callerid:7] Set(“SIP/6811-00001eee”, “AMPUSERCID=6811”) in new stack – Executing [s@macro-user-callerid:8] Set(“SIP/6811-00001eee”, “CALLERID(all)=“JH” \<6811\>”) in new stack – Executing [s@macro-user-callerid:9] ExecIf(“SIP/6811-00001eee”, “0?Set(CHANNEL(language)=)”) in new stack – Executing [s@macro-user-callerid:10] GotoIf(“SIP/6811-00001eee”, “0?continue”) in new stack – Executing [s@macro-user-callerid:11] Set(“SIP/6811-00001eee”, “\_\_TTL=64”) in new stack – Executing [s@macro-user-callerid:12] GotoIf(“SIP/6811-00001eee”, “1?continue”) in new stack – Goto (macro-user-callerid,s,19) – Executing [s@macro-user-callerid:19] NoOp(“SIP/6811-00001eee”, “Using CallerID “JH” \<6811\>”) in new stack – Executing [s@macro-exten-vm:2] Set(“SIP/6811-00001eee”, “RingGroupMethod=none”) in new stack – Executing [s@macro-exten-vm:3] Set(“SIP/6811-00001eee”, “VMBOX=novm”) in new stack – Executing [s@macro-exten-vm:4] Set(“SIP/6811-00001eee”, “EXTTOCALL=6816”) in new stack – Executing [s@macro-exten-vm:5] Set(“SIP/6811-00001eee”, “CFUEXT=”) in new stack – Executing [s@macro-exten-vm:6] Set(“SIP/6811-00001eee”, “CFBEXT=”) in new stack – Executing [s@macro-exten-vm:7] Set(“SIP/6811-00001eee”, “RT=”"") in new stack – Executing [s@macro-exten-vm:8] Macro(“SIP/6811-00001eee”, “record-enable,6816,IN”) in new stack – Executing [s@macro-record-enable:1] GotoIf(“SIP/6811-00001eee”, “1?check”) in new stack – Goto (macro-record-enable,s,4) – Executing [s@macro-record-enable:4] AGI(“SIP/6811-00001eee”, “recordingcheck,20120413-113733,1334313453.13135”) in new stack – Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck,20120413-113733,1334313453.13135: Inbound recording not enabled – \AGI Script recordingcheck completed, returning 0 – Executing [s@macro-record-enable:5] MacroExit(“SIP/6811-00001eee”, “”) in new stack – Executing [s@macro-exten-vm:9] Macro(“SIP/6811-00001eee”, “dial,”",trw,6816") in new stack – Executing [s@macro-dial:1] GotoIf(“SIP/6811-00001eee”, “1?dial”) in new stack – Goto (macro-dial,s,3) – Executing [s@macro-dial:3] AGI(“SIP/6811-00001eee”, “dialparties.agi”) in new stack – Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi dialparties.agi: Starting New Dialparties.agi dialparties.agi: Caller ID name is ‘JH’ number is ‘6811’ \> dialparties.agi: USE\_CONFIRMATION: ‘FALSE’ \> dialparties.agi: RINGGROUP\_INDEX: '' dialparties.agi: Methodology of ring is ‘none’ – dialparties.agi: Added extension 6816 to extension map – dialparties.agi: Extension 6816 cf is disabled – dialparties.agi: Extension 6816 do not disturb is disabled \> dialparties.agi: extnum 6816 has: cw: 1; hascfb: 0 [] hascfu: 0 [] dialparties.agi: EXTENSION\_STATE: 0 (NOT\_INUSE) – dialparties.agi: dbset CALLTRACE/6816 to 6811 – dialparties.agi: Filtered ARG3: 6816 – \AGI Script dialparties.agi completed, returning 0 – Executing [s@macro-dial:7] Dial(“SIP/6811-00001eee”, “SIP/6816,”",trw") in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 == Using SIP VRTP TOS bits 136 == Using SIP VRTP CoS mark 6 == Extension Changed 6816[ext-local] new state Ringing for Notify User 6833 Audio is at 10.50.0.10 port 14828 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 10.50.0.144:3894: INVITE sip:6816@10.50.0.144:3894;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.50.0.10:5060;branch=z9hG4bK1f7241f2;rport Max-Forwards: 70 From: “JH” sip:6811@10.50.0.10;tag=as72861f0c To: sip:6816@10.50.0.144:3894;transport=udp Contact: sip:6811@10.50.0.10 Call-ID: [7c8231dd367227462cbdbafc38429059@10.50.0.10](mailto:7c8231dd367227462cbdbafc38429059@10.50.0.10) CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78 Date: Fri, 13 Apr 2012 10:37:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 270 v=0 o=root 139012037 139012037 IN IP4 10.50.0.10 s=Asterisk PBX 1.6.0.26-FONCORE-r78 c=IN IP4 10.50.0.10 t=0 0 m=audio 14828 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv * * * ``` -- Called 6816 ``` pbx\*CLI\> \<— SIP read from UDP://10.50.0.144:3894 —\> SIP/2.0 410 Gone Via: SIP/2.0/UDP 10.50.0.10:5060;branch=z9hG4bK1f7241f2;rport=5060 From: “JH” sip:6811@10.50.0.10;tag=as72861f0c To: sip:6816@10.50.0.144:3894;transport=udp;tag=yeykxg Call-ID: [7c8231dd367227462cbdbafc38429059@10.50.0.10](mailto:7c8231dd367227462cbdbafc38429059@10.50.0.10) CSeq: 102 INVITE Contact: sip:6816@10.50.0.144:3894;transport=udp Supported: 100rel, replaces, norefersub Content-Length: 0 \<-------------\> — (9 headers 0 lines) — – Got SIP response 410 “Gone” back from 10.50.0.144 Transmitting (NAT) to 10.50.0.144:3894: ACK sip:6816@10.50.0.144:3894;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.50.0.10:5060;branch=z9hG4bK1f7241f2;rport Max-Forwards: 70 From: “JH” sip:6811@10.50.0.10;tag=as72861f0c To: sip:6816@10.50.0.144:3894;transport=udp;tag=yeykxg Contact: sip:6811@10.50.0.10 Call-ID: [7c8231dd367227462cbdbafc38429059@10.50.0.10](mailto:7c8231dd367227462cbdbafc38429059@10.50.0.10) CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78 Content-Length: 0 * * * ``` -- SIP/6816-00001eef is circuit-busy ``` == Extension Changed 6816[ext-local] new state Idle for Notify User 6833 (queued) == Everyone is busy/congested at this time (1:0/1/0) – Executing [s@macro-dial:8] Set(“SIP/6811-00001eee”, “DIALSTATUS=CONGESTION”) in new stack – Executing [s@macro-dial:9] GosubIf(“SIP/6811-00001eee”, “0?CONGESTION,1”) in new stack – Executing [s@macro-exten-vm:10] GotoIf(“SIP/6811-00001eee”, “0?exit,return”) in new stack – Executing [s@macro-exten-vm:11] Set(“SIP/6811-00001eee”, “SV\_DIALSTATUS=CONGESTION”) in new stack – Executing [s@macro-exten-vm:12] GosubIf(“SIP/6811-00001eee”, “0?docfu,1”) in new stack – Executing [s@macro-exten-vm:13] GosubIf(“SIP/6811-00001eee”, “0?docfb,1”) in new stack – Executing [s@macro-exten-vm:14] Set(“SIP/6811-00001eee”, “DIALSTATUS=CONGESTION”) in new stack – Executing [s@macro-exten-vm:15] NoOp(“SIP/6811-00001eee”, “Voicemail is ‘novm’”) in new stack – Executing [s@macro-exten-vm:16] GotoIf(“SIP/6811-00001eee”, “1?s-CONGESTION,1”) in new stack – Goto (macro-exten-vm,s-CONGESTION,1) – Executing [s-CONGESTION@macro-exten-vm:1] NoOp(“SIP/6811-00001eee”, "IVR\_RETVM: IVR\_CONTEXT: ") in new stack – Executing [s-CONGESTION@macro-exten-vm:2] GotoIf(“SIP/6811-00001eee”, “0?exit,1”) in new stack – Executing [s-CONGESTION@macro-exten-vm:3] PlayTones(“SIP/6811-00001eee”, “congestion”) in new stack – Executing [s-CONGESTION@macro-exten-vm:4] Congestion(“SIP/6811-00001eee”, “10”) in new stack == Spawn extension (macro-exten-vm, s-CONGESTION, 4) exited non-zero on ‘SIP/6811-00001eee’ in macro ‘exten-vm’ == Spawn extension (from-internal, 6816, 1) exited non-zero on ‘SIP/6811-00001eee’ – Executing [h@from-internal:1] Macro(“SIP/6811-00001eee”, “hangupcall”) in new stack – Executing [s@macro-hangupcall:1] GotoIf(“SIP/6811-00001eee”, “1?skiprg”) in new stack – Goto (macro-hangupcall,s,4) – Executing [s@macro-hangupcall:4] GotoIf(“SIP/6811-00001eee”, “1?skipblkvm”) in new stack – Goto (macro-hangupcall,s,7) – Executing [s@macro-hangupcall:7] GotoIf(“SIP/6811-00001eee”, “1?theend”) in new stack – Goto (macro-hangupcall,s,9) – Executing [s@macro-hangupcall:9] Hangup(“SIP/6811-00001eee”, “”) in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on ‘SIP/6811-00001eee’ in macro ‘hangupcall’ == Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/6811-00001eee’ == Extension Changed 6816[from-internal] new state Idle for Notify User 6833 Really destroying SIP dialog ‘7c8231dd367227462cbdbafc38429059@10.50.0.10’ Method: INVITE pbx\*CLI\> ---
**Author:** ![system](https://global.discourse-cdn.com/asterisk/original/2X/4/475d0c889911cfb84ac4881ad6df8b523e3f9105.png) [@system](https://community.asterisk.org/u/system)\ **Post date:** [April 16, 2012, 9:08am UTC](https://community.asterisk.org/t/are-snom-phones-fully-compatible-with-asterisk-trixbox/38868/11 "2012-04-16T09:08:49Z")
anyone ?? :0(