Are snom phones fully compatible with asterisk/trixbox

[Theguru]

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??

[tomdemoor]

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

[Theguru]

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

any ideas?? this is driving me nuts!!

[omid.mohajerani]

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 .

[Theguru]

ok here it goes this is using the sip show peer 6816 command

does this tell you anything that is wrong??

* Name : 6816
Secret : <Set>
MD5Secret : <Not set>
Context : from-internal
Subscr.Cont. : <Not set>
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

[omid.mohajerani]

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

[Theguru]

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>

[omid.mohajerani]

== 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 )

[miguellinux]

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

[Theguru]

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>

[Theguru]

anyone ?? :0(