Are snom phones fully compatible with asterisk/trixbox

Asterisk Asterisk Support
1

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??

2

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

3

hi there thanks for the reply, it says peep $snom9 not found?

any ideas?? this is driving me nuts!!

4

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 .

5

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
6

It seems it is successfully registered . you should put your asterisk console log here when you are calling 6816

7

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
– <SIP/6811-00001e6f>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
– <SIP/6811-00001e6f>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>

8

== 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 )

9

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

10

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
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
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
– <SIP/6811-00001eee>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
– <SIP/6811-00001eee>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
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
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
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>

11

anyone ?? :0(