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IVR and SPA941's

chrisbn02
Posts: 2
 Member Since:
2008-01-17
Submitted by chrisbn02 on Fri, 07/04/2008 - 3:43am.

Hi all,

I'm very new to trixbox. I'm trying to set it up with IVR and am also using SPA400, a couple of SPA941's and some softphones (SJ Phone). I'm having trouble with my incomming calls. I've setup my incomming route to ring the IVR via the SPA400. If i transfer to the extensions being used by my softphones, I can get through without any problems. But if i transfer to the extensions used by the 941's, I immediately get a busy tone. I've also tried zeroing out all the feature codes in the 941's, to no avail.

Any help would really really be appreciated.

Thanks!

‹ Voicemail not working Freepbx update to 2.4.0 now sending email of my asterisk logs ›

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eeknz
Posts: 173
 Member Since:
2006-08-13
I love the SPA-9xx phones.
Fri, 07/04/2008 - 4:00am

I love the SPA-9xx phones. Use the endpoint manager to config the phones if you can. You'll need a DHCP server that can support the correct option for remote boot.
Factory Reset the phones and let them auto configure.
Seems to me that you're having a phone config problem that the auto config will clear up nicely.
If you can't work out the endpoint manager, go watch this instead and do it the long way:
http://www.asterisktutorials.com/videos/942/movie.html


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chrisbn01
Posts: 3
 Member Since:
2008-01-17
Sorry, this reply took a
Wed, 07/09/2008 - 7:14pm

Sorry, this reply took a while 'coz i went on travel.

@eeknz, thanks for helping out.

Anyway, i actually did use the endpoint manager to configure my SPA's. Furthermore, i've also come accross the vid eeknz showed above a few days back, to no avail.

Btw, my problem only happens when I use IVR. If i tell my incomming route to forward calls directly to the SPA, it works. Problem only occurs when I use the IVR. Might this be a DTMF setting issue on the SPA? If so, which options should i change?

Addendum: I also have a few grandstream phones, and it works well with the same IVR. Am only stuck on the SPA941's.

Please help, i'm really new to this and my boss expects me to work this out :(


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netout
Posts: 187
 Member Since:
2007-08-18
Go download PuTTy.exe. This
Wed, 07/09/2008 - 8:54pm

Go download PuTTy.exe. This will allow you to SSH into the system. After logging in as root. Run the following

# asterisk -r

This will run the asterisk console. Aftre trying one of the failed calles type

#exit

This will exit the asterisk console. Copy and past the results here and I would love to see what's going on with the system.

--

Michael D Mosier
Sr. Telecom Engineer
Network Outfitters
Houston, Austin and San Antonio
Support Available
832-715-6981


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chrisbn01
Posts: 3
 Member Since:
2008-01-17
trixbox1*CLI> trixbox1*CLI>
Wed, 07/09/2008 - 11:29pm

trixbox1*CLI>
trixbox1*CLI>
-- Executing [SPA400@from-trunk:1] Set("SIP/localhost-09a48990", "__FROM_DID=SPA400") in n
ew stack
-- Executing [SPA400@from-trunk:2] GotoIf("SIP/localhost-09a48990", "1 ?cidok") in new sta
ck
-- Goto (from-trunk,SPA400,4)
-- Executing [SPA400@from-trunk:4] NoOp("SIP/localhost-09a48990", "CallerID is "- FXO_Port
_ID_1" ") in new stack
-- Executing [SPA400@from-trunk:5] Set("SIP/localhost-09a48990", "__CALLINGPRES_SV=allowed
_not_screened") in new stack
-- Executing [SPA400@from-trunk:6] SetCallerPres("SIP/localhost-09a48990", "allowed_not_sc
reened") in new stack
-- Executing [SPA400@from-trunk:7] Goto("SIP/localhost-09a48990", "ivr-2|s|1") in new stac
k
-- Goto (ivr-2,s,1)
-- Executing [s@ivr-2:1] Set("SIP/localhost-09a48990", "LOOPCOUNT=0") in new stack
-- Executing [s@ivr-2:2] Set("SIP/localhost-09a48990", "__DIR-CONTEXT=") in new stack
-- Executing [s@ivr-2:3] Set("SIP/localhost-09a48990", "_IVR_CONTEXT_ivr-2=") in new stack
-- Executing [s@ivr-2:4] Set("SIP/localhost-09a48990", "_IVR_CONTEXT=ivr-2") in new stack
-- Executing [s@ivr-2:5] GotoIf("SIP/localhost-09a48990", "0?begin") in new stack
-- Executing [s@ivr-2:6] Answer("SIP/localhost-09a48990", "") in new stack
-- Executing [s@ivr-2:7] Wait("SIP/localhost-09a48990", "1") in new stack
-- Executing [s@ivr-2:8] Set("SIP/localhost-09a48990", "TIMEOUT(digit)=3") in new stack
-- Digit timeout set to 3
-- Executing [s@ivr-2:9] Set("SIP/localhost-09a48990", "TIMEOUT(response)=10") in new stac
k
-- Response timeout set to 10
-- Executing [s@ivr-2:10] BackGround("SIP/localhost-09a48990", "custom/Pacific") in new st
ack
-- Playing 'custom/Pacific' (language 'en')
== CDR updated on SIP/localhost-09a48990
-- Executing [1@ivr-2:1] DBdel("SIP/localhost-09a48990", "") in new stack
-- Executing [1@ivr-2:2] Set("SIP/localhost-09a48990", "__NODEST=") in new stack
-- Executing [1@ivr-2:3] Goto("SIP/localhost-09a48990", "from-did-direct|2001|1") in new stack
-- Goto (from-did-direct,2001,1)
-- Executing [2001@from-did-direct:1] Macro("SIP/localhost-09a48990", "exten-vm|novm|2001") in new stack
-- Executing [s@macro-exten-vm:1] Macro("SIP/localhost-09a48990", "user-callerid") in new stack
-- Executing [s@macro-user-callerid:1] NoOp("SIP/localhost-09a48990", "user-callerid: - FXO_Port_ID_1 anonymous") in new stack
-- Executing [s@macro-user-callerid:2] Set("SIP/localhost-09a48990", "AMPUSER=anonymous") in new stack
-- Executing [s@macro-user-callerid:3] GotoIf("SIP/localhost-09a48990", "0?report") in new stack
-- Executing [s@macro-user-callerid:4] ExecIf("SIP/localhost-09a48990", "1|Set|REALCALLERIDNUM=anonymous") in new stack
-- Executing [s@macro-user-callerid:5] NoOp("SIP/localhost-09a48990", "REALCALLERIDNUM is anonymous") in new stack
-- Executing [s@macro-user-callerid:6] Set("SIP/localhost-09a48990", "AMPUSER=") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/localhost-09a48990", "AMPUSERCIDNAME=") in new stack
-- Executing [s@macro-user-callerid:8] GotoIf("SIP/localhost-09a48990", "1?report") in new stack
-- Goto (macro-user-callerid,s,13)
-- Executing [s@macro-user-callerid:13] NoOp("SIP/localhost-09a48990", "TTL: ARG1: novm") in new stack
-- Executing [s@macro-user-callerid:14] GotoIf("SIP/localhost-09a48990", "0?continue") in new stack
-- Executing [s@macro-user-callerid:15] Set("SIP/localhost-09a48990", "__TTL=64") in new stack
-- Executing [s@macro-user-callerid:16] GotoIf("SIP/localhost-09a48990", "1?continue") in new stack
-- Goto (macro-user-callerid,s,23)
-- Executing [s@macro-user-callerid:23] NoOp("SIP/localhost-09a48990", "Using CallerID "- FXO_Port_ID_1" ") in new stack
-- Executing [s@macro-exten-vm:2] Set("SIP/localhost-09a48990", "FROMCONTEXT=exten-vm") in new stack
-- Executing [s@macro-exten-vm:3] Set("SIP/localhost-09a48990", "VMBOX=novm") in new stack
-- Executing [s@macro-exten-vm:4] Set("SIP/localhost-09a48990", "EXTTOCALL=2001") in new stack
-- Executing [s@macro-exten-vm:5] Set("SIP/localhost-09a48990", "CFUEXT=") in new stack
-- Executing [s@macro-exten-vm:6] Set("SIP/localhost-09a48990", "CFBEXT=") in new stack
-- Executing [s@macro-exten-vm:7] Set("SIP/localhost-09a48990", "RT=""") in new stack
-- Executing [s@macro-exten-vm:8] Macro("SIP/localhost-09a48990", "record-enable|2001|IN") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/localhost-09a48990", "0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] AGI("SIP/localhost-09a48990", "recordingcheck|20080710-142201|1215670916.44") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20080710-142201|1215670916.44: Inbound recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing [s@macro-record-enable:5] NoOp("SIP/localhost-09a48990", "No recording needed") in new stack
-- Executing [s@macro-exten-vm:9] Macro("SIP/localhost-09a48990", "dial||tr|2001") in new stack
-- Executing [s@macro-dial:1] GotoIf("SIP/localhost-09a48990", "1?dial") in new stack
-- Goto (macro-dial,s,3)
-- Executing [s@macro-dial:3] AGI("SIP/localhost-09a48990", "dialparties.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'admin' logged on from 127.0.0.1
dialparties.agi: Caller ID name is '- FXO_Port_ID_1' number is 'anonymous'
dialparties.agi: Methodology of ring is 'none'
-- dialparties.agi: Added extension 2001 to extension map
-- dialparties.agi: Extension 2001 cf is disabled
-- dialparties.agi: Extension 2001 do not disturb is disabled
dialparties.agi: Extension 2001 has ExtensionState: 0
-- dialparties.agi: Checking CW and CFB status for extension 2001
-- dialparties.agi: DbDel CALLTRACE/2001 - Caller ID is not defined
-- dialparties.agi: Filtered ARG3: 2001
== Manager 'admin' logged off from 127.0.0.1
-- AGI Script dialparties.agi completed, returning 0
-- Executing [s@macro-dial:7] Dial("SIP/localhost-09a48990", "SIP/2001||tr") in new stack
-- Called 2001
-- Got SIP response 406 "Not Acceptable" back from 61.9.108.31
-- No one is available to answer at this time (1:0/0/0)
-- Executing [s@macro-dial:8] Set("SIP/localhost-09a48990", "DIALSTATUS=NOANSWER") in new stack
-- Executing [s@macro-exten-vm:10] Set("SIP/localhost-09a48990", "SV_DIALSTATUS=NOANSWER") in new stack
-- Executing [s@macro-exten-vm:11] GosubIf("SIP/localhost-09a48990", "0?docfu|1") in new stack
-- Executing [s@macro-exten-vm:12] GosubIf("SIP/localhost-09a48990", "0?docfb|1") in new stack
-- Executing [s@macro-exten-vm:13] Set("SIP/localhost-09a48990", "DIALSTATUS=NOANSWER") in new stack
-- Executing [s@macro-exten-vm:14] NoOp("SIP/localhost-09a48990", "Voicemail is novm") in new stack
-- Executing [s@macro-exten-vm:15] GotoIf("SIP/localhost-09a48990", "1?s-NOANSWER|1") in new stack
-- Goto (macro-exten-vm,s-NOANSWER,1)
-- Executing [s-NOANSWER@macro-exten-vm:1] PlayTones("SIP/localhost-09a48990", "congestion") in new stack
-- Executing [s-NOANSWER@macro-exten-vm:2] Congestion("SIP/localhost-09a48990", "10") in new stack
trixbox1*CLI>
trixbox1*CLI> exit
[trixbox1.localdomain ~]#

@M2Mcom, thanks so much for the attention. Hopefully you could help me.


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chrisbn01
Posts: 3
 Member Since:
2008-01-17
Help anyone? Pls....
Sun, 07/13/2008 - 7:54pm

Help anyone? Pls....


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stechnique
Posts: 733
 Member Since:
2008-02-21
Login to asterisk CLI again
Mon, 07/14/2008 - 4:50am

Login to asterisk CLI again and do:

# sip show peers

Is the 941 registered?
Seems it isn't:

-- Called 2001
-- Got SIP response 406 "Not Acceptable" back from 61.9.108.31

You can turn on SIP debugging for that extension and get more detailed logs:

sip set debug peer 2001

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chrisbn02
Posts: 2
 Member Since:
2008-01-17
Hmm.. left my voip project
Fri, 09/12/2008 - 1:15am

Hmm.. left my voip project for some time since I had to travel again. Now, am back on it.. and going back to my problem, here it is....

-------------------------------------------------------------------------------

Login to asterisk CLI again and do:

# sip show peers
Is the 941 registered?
-------------------------------------------------------------------------------

Yup it is.. look:

*CLI>
*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
SPA400 192.168.111.204 5060 Unmonitored
2004/2004 192.168.111.149 D N 5060 OK (1 ms)
2003 (Unspecified) D N 0 UNKNOWN
2002 (Unspecified) D N 0 UNKNOWN
2001/2001 192.168.111.203 D N 5060 OK (12 ms)
5 sip peers [Monitored: 2 online, 2 offline Unmonitored: 1 online, 0 offline]
*CLI>

SPA941 in question is the 2001/2001 (192.168.111.203).

Here's the debug:

-------------------------------------------------------------------------------

*CLI> -- Executing [SPA400@from-trunk:1] Set("SIP/localhost-0914b100", "__FROM_DID=SPA400") in new stack
-- Executing [SPA400@from-trunk:2] GotoIf("SIP/localhost-0914b100", "1 ?cidok") in new stack
-- Goto (from-trunk,SPA400,4)
-- Executing [SPA400@from-trunk:4] NoOp("SIP/localhost-0914b100", "CallerID is "- FXO_Port_ID_1" ") in new stack
-- Executing [SPA400@from-trunk:5] Set("SIP/localhost-0914b100", "__CALLINGPRES_SV=allowed_not_screened") in new stack
-- Executing [SPA400@from-trunk:6] SetCallerPres("SIP/localhost-0914b100", "allowed_not_screened") in new stack
-- Executing [SPA400@from-trunk:7] Goto("SIP/localhost-0914b100", "ivr-2|s|1") in new stack
-- Goto (ivr-2,s,1)
-- Executing [s@ivr-2:1] Set("SIP/localhost-0914b100", "LOOPCOUNT=0") in new stack
-- Executing [s@ivr-2:2] Set("SIP/localhost-0914b100", "__DIR-CONTEXT=default") in new stack
-- Executing [s@ivr-2:3] Set("SIP/localhost-0914b100", "_IVR_CONTEXT_ivr-2=") in new stack
-- Executing [s@ivr-2:4] Set("SIP/localhost-0914b100", "_IVR_CONTEXT=ivr-2") in new stack
-- Executing [s@ivr-2:5] GotoIf("SIP/localhost-0914b100", "0?begin") in new stack
-- Executing [s@ivr-2:6] Answer("SIP/localhost-0914b100", "") in new stack
-- Executing [s@ivr-2:7] Wait("SIP/localhost-0914b100", "1") in new stack
-- Executing [s@ivr-2:8] Set("SIP/localhost-0914b100", "TIMEOUT(digit)=3") in new stack
-- Digit timeout set to 3
-- Executing [s@ivr-2:9] Set("SIP/localhost-0914b100", "TIMEOUT(response)=10") in new stack
-- Response timeout set to 10
-- Executing [s@ivr-2:10] BackGround("SIP/localhost-0914b100", "custom/Pacific") in new stack
-- Playing 'custom/Pacific' (language 'en')
== CDR updated on SIP/localhost-0914b100
-- Executing [1@ivr-2:1] DBdel("SIP/localhost-0914b100", "") in new stack
-- Executing [1@ivr-2:2] Set("SIP/localhost-0914b100", "__NODEST=") in new stack
-- Executing [1@ivr-2:3] Goto("SIP/localhost-0914b100", "from-did-direct|2001|1") in new stack
-- Goto (from-did-direct,2001,1)
-- Executing [2001@from-did-direct:1] Macro("SIP/localhost-0914b100", "exten-vm|2001|2001") in new stack
-- Executing [s@macro-exten-vm:1] Macro("SIP/localhost-0914b100", "user-callerid") in new stack
-- Executing [s@macro-user-callerid:1] NoOp("SIP/localhost-0914b100", "user-callerid: - FXO_Port_ID_1 anonymous") in new stack
-- Executing [s@macro-user-callerid:2] Set("SIP/localhost-0914b100", "AMPUSER=anonymous") in new stack
-- Executing [s@macro-user-callerid:3] GotoIf("SIP/localhost-0914b100", "0?report") in new stack
-- Executing [s@macro-user-callerid:4] ExecIf("SIP/localhost-0914b100", "1|Set|REALCALLERIDNUM=anonymous") in new stack
-- Executing [s@macro-user-callerid:5] NoOp("SIP/localhost-0914b100", "REALCALLERIDNUM is anonymous") in new stack
-- Executing [s@macro-user-callerid:6] Set("SIP/localhost-0914b100", "AMPUSER=") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/localhost-0914b100", "AMPUSERCIDNAME=") in new stack
-- Executing [s@macro-user-callerid:8] GotoIf("SIP/localhost-0914b100", "1?report") in new stack
-- Goto (macro-user-callerid,s,13)
-- Executing [s@macro-user-callerid:13] NoOp("SIP/localhost-0914b100", "TTL: ARG1: 2001") in new stack
-- Executing [s@macro-user-callerid:14] GotoIf("SIP/localhost-0914b100", "0?continue") in new stack
-- Executing [s@macro-user-callerid:15] Set("SIP/localhost-0914b100", "__TTL=64") in new stack
-- Executing [s@macro-user-callerid:16] GotoIf("SIP/localhost-0914b100", "1?continue") in new stack
-- Goto (macro-user-callerid,s,23)
-- Executing [s@macro-user-callerid:23] NoOp("SIP/localhost-0914b100", "Using CallerID "- FXO_Port_ID_1" ") in new stack
-- Executing [s@macro-exten-vm:2] Set("SIP/localhost-0914b100", "FROMCONTEXT=exten-vm") in new stack
-- Executing [s@macro-exten-vm:3] Set("SIP/localhost-0914b100", "VMBOX=2001") in new stack
-- Executing [s@macro-exten-vm:4] Set("SIP/localhost-0914b100", "EXTTOCALL=2001") in new stack
-- Executing [s@macro-exten-vm:5] Set("SIP/localhost-0914b100", "CFUEXT=") in new stack
-- Executing [s@macro-exten-vm:6] Set("SIP/localhost-0914b100", "CFBEXT=") in new stack
-- Executing [s@macro-exten-vm:7] Set("SIP/localhost-0914b100", "RT=15") in new stack
-- Executing [s@macro-exten-vm:8] Macro("SIP/localhost-0914b100", "record-enable|2001|IN") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/localhost-0914b100", "0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] AGI("SIP/localhost-0914b100", "recordingcheck|20080912-161215|1221207124.12") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20080912-161215|1221207124.12: Inbound recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing [s@macro-record-enable:5] NoOp("SIP/localhost-0914b100", "No recording needed") in new stack
-- Executing [s@macro-exten-vm:9] Macro("SIP/localhost-0914b100", "dial|15|tr|2001") in new stack
-- Executing [s@macro-dial:1] GotoIf("SIP/localhost-0914b100", "1?dial") in new stack
-- Goto (macro-dial,s,3)
-- Executing [s@macro-dial:3] AGI("SIP/localhost-0914b100", "dialparties.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'admin' logged on from 127.0.0.1
dialparties.agi: Caller ID name is '- FXO_Port_ID_1' number is 'anonymous'
dialparties.agi: Methodology of ring is 'none'
-- dialparties.agi: Added extension 2001 to extension map
-- dialparties.agi: Extension 2001 cf is disabled
-- dialparties.agi: Extension 2001 do not disturb is disabled
dialparties.agi: Extension 2001 has ExtensionState: 0
-- dialparties.agi: Checking CW and CFB status for extension 2001
-- dialparties.agi: DbDel CALLTRACE/2001 - Caller ID is not defined
-- dialparties.agi: Filtered ARG3: 2001
== Manager 'admin' logged off from 127.0.0.1
-- AGI Script dialparties.agi completed, returning 0
-- Executing [s@macro-dial:7] Dial("SIP/localhost-0914b100", "SIP/2001|15|tr") in new stack
Audio is at 192.168.111.200 port 16370
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.111.203:5060:
INVITE sip:2001@192.168.111.203:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.200:5060;branch=z9hG4bK3af3b0fc;rport
From: "- FXO_Port_ID_1" ;tag=as2585da52
To:
Contact:
Call-ID: 67587a1975525ed833f9d7bd72565555@192.168.111.200
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 12 Sep 2008 08:12:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 270

v=0
o=root 10403 10403 IN IP4 192.168.111.200
s=session
c=IN IP4 192.168.111.200
t=0 0
m=audio 16370 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
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 2001


SIP/2.0 100 Trying
To:
From: "- FXO_Port_ID_1" ;tag=as2585da52
Call-ID: 67587a1975525ed833f9d7bd72565555@192.168.111.200
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.111.200:5060;branch=z9hG4bK3af3b0fc
Server: Linksys/SPA941-5.1.8
Content-Length: 0


--- (8 headers 0 lines) ---


SIP/2.0 406 Not Acceptable
To: ;tag=465c2f4ab158be67i0
From: "- FXO_Port_ID_1" ;tag=as2585da52
Call-ID: 67587a1975525ed833f9d7bd72565555@192.168.111.200
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.111.200:5060;branch=z9hG4bK3af3b0fc
Server: Linksys/SPA941-5.1.8
Content-Length: 0


--- (8 headers 0 lines) ---
-- Got SIP response 406 "Not Acceptable" back from 192.168.111.203
Transmitting (NAT) to 192.168.111.203:5060:
ACK sip:2001@192.168.111.203:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.200:5060;branch=z9hG4bK3af3b0fc;rport
From: "- FXO_Port_ID_1" ;tag=as2585da52
To: ;tag=465c2f4ab158be67i0
Contact:
Call-ID: 67587a1975525ed833f9d7bd72565555@192.168.111.200
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

---
Really destroying SIP dialog '67587a1975525ed833f9d7bd72565555@192.168.111.200' Method: INVITE
-- No one is available to answer at this time (1:0/0/0)
-- Executing [s@macro-dial:8] Set("SIP/localhost-0914b100", "DIALSTATUS=NOANSWER") in new stack
-- Executing [s@macro-exten-vm:10] Set("SIP/localhost-0914b100", "SV_DIALSTATUS=NOANSWER") in new stack
-- Executing [s@macro-exten-vm:11] GosubIf("SIP/localhost-0914b100", "0?docfu|1") in new stack
-- Executing [s@macro-exten-vm:12] GosubIf("SIP/localhost-0914b100", "0?docfb|1") in new stack
-- Executing [s@macro-exten-vm:13] Set("SIP/localhost-0914b100", "DIALSTATUS=NOANSWER") in new stack
-- Executing [s@macro-exten-vm:14] NoOp("SIP/localhost-0914b100", "Voicemail is 2001") in new stack
-- Executing [s@macro-exten-vm:15] GotoIf("SIP/localhost-0914b100", "0?s-NOANSWER|1") in new stack
-- Executing [s@macro-exten-vm:16] NoOp("SIP/localhost-0914b100", "Sending to Voicemail box 2001") in new stack
-- Executing [s@macro-exten-vm:17] Macro("SIP/localhost-0914b100", "vm|2001|NOANSWER") in new stack
-- Executing [s@macro-vm:1] Macro("SIP/localhost-0914b100", "user-callerid|SKIPTTL") in new stack
-- Executing [s@macro-user-callerid:1] NoOp("SIP/localhost-0914b100", "user-callerid: - FXO_Port_ID_1 anonymous") in new stack
-- Executing [s@macro-user-callerid:2] Set("SIP/localhost-0914b100", "AMPUSER=anonymous") in new stack
-- Executing [s@macro-user-callerid:3] GotoIf("SIP/localhost-0914b100", "0?report") in new stack
-- Executing [s@macro-user-callerid:4] ExecIf("SIP/localhost-0914b100", "0|Set|REALCALLERIDNUM=anonymous") in new stack
-- Executing [s@macro-user-callerid:5] NoOp("SIP/localhost-0914b100", "REALCALLERIDNUM is anonymous") in new stack
-- Executing [s@macro-user-callerid:6] Set("SIP/localhost-0914b100", "AMPUSER=") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/localhost-0914b100", "AMPUSERCIDNAME=") in new stack
-- Executing [s@macro-user-callerid:8] GotoIf("SIP/localhost-0914b100", "1?report") in new stack
-- Goto (macro-user-callerid,s,13)
-- Executing [s@macro-user-callerid:13] NoOp("SIP/localhost-0914b100", "TTL: 64 ARG1: SKIPTTL") in new stack
-- Executing [s@macro-user-callerid:14] GotoIf("SIP/localhost-0914b100", "1?continue") in new stack
-- Goto (macro-user-callerid,s,23)
-- Executing [s@macro-user-callerid:23] NoOp("SIP/localhost-0914b100", "Using CallerID "- FXO_Port_ID_1" ") in new stack
-- Executing [s@macro-vm:2] Set("SIP/localhost-0914b100", "VMGAIN=""") in new stack
-- Executing [s@macro-vm:3] GotoIf("SIP/localhost-0914b100", "1?vmx|1") in new stack
-- Goto (macro-vm,vmx,1)
-- Executing [vmx@macro-vm:1] GotoIf("SIP/localhost-0914b100", "0?s-NOANSWER|1") in new stack
-- Executing [vmx@macro-vm:2] Set("SIP/localhost-0914b100", "MODE=unavail") in new stack
-- Executing [vmx@macro-vm:3] GotoIf("SIP/localhost-0914b100", "1?notdirect") in new stack
-- Goto (macro-vm,vmx,5)
-- Executing [vmx@macro-vm:5] NoOp("SIP/localhost-0914b100", "Checking if ext 2001 is enabled: ") in new stack
-- Executing [vmx@macro-vm:6] GotoIf("SIP/localhost-0914b100", "1?s-NOANSWER|1") in new stack
-- Goto (macro-vm,s-NOANSWER,1)
-- Executing [s-NOANSWER@macro-vm:1] Macro("SIP/localhost-0914b100", "get-vmcontext|2001") in new stack
-- Executing [s@macro-get-vmcontext:1] Set("SIP/localhost-0914b100", "VMCONTEXT=default") in new stack
-- Executing [s@macro-get-vmcontext:2] GotoIf("SIP/localhost-0914b100", "0?200:300") in new stack
-- Goto (macro-get-vmcontext,s,300)
-- Executing [s@macro-get-vmcontext:300] NoOp("SIP/localhost-0914b100", "") in new stack
-- Executing [s-NOANSWER@macro-vm:2] VoiceMail("SIP/localhost-0914b100", "2001@default|u") in new stack
-- Playing 'vm-theperson' (language 'en')
-- Playing 'digits/2' (language 'en')
-- Playing 'digits/0' (language 'en')
-- Playing 'digits/0' (language 'en')
-- Playing 'digits/1' (language 'en')
-- Playing 'vm-isunavail' (language 'en')
-- Playing 'vm-intro' (language 'en')
-- Playing 'beep' (language 'en')
-- Recording the message
-- x=0, open writing: /var/spool/asterisk/voicemail/default/2001/tmp/JiOolJ format: wav49, 0x91493b0
-- x=1, open writing: /var/spool/asterisk/voicemail/default/2001/tmp


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leeph
Posts: 2
 Member Since:
2009-07-22
fixed it
Wed, 07/22/2009 - 8:04am

Hi there,

I had this problem.

I hate finding posts about an exact problem and there being no solution.

Anyway, after much digging and checking of logs etc, I discovered that the SPA941 by default blocks
SIP calls with no callerID. Since I can see from your debug that the call coming in is anonymous, I think
you have the same problem I did.

To resolve, log into the phone's web config and go to the User tab and set Block ANC Setting to NO.

It's a pretty rubbish feature as like you said, it only seemed to block calls coming in from a Ring Group or IVR. Calls made directly to the extension would ultimately get through, despite the 406 unacceptable message appearing in the Asterisk console trace.

Hope that helps you and any others having this same annoying problem!!

leeph


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