Calls between sip extensions, don't work.

seaq

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#1
I've configured some SIP extensions, but when try to call between extensions, these messages appear in the CLI>:

-- Executing [104@from-internal:1] Macro("SIP/100-096edbe0", "exten-vm|104|104") in new stack
-- Executing [s@macro-exten-vm:1] Macro("SIP/100-096edbe0", "user-callerid") in new stack
-- Executing [s@macro-user-callerid:1] NoOp("SIP/100-096edbe0", "user-callerid: device 100") in new stack
-- Executing [s@macro-user-callerid:2] Set("SIP/100-096edbe0", "AMPUSER=100") in new stack
-- Executing [s@macro-user-callerid:3] GotoIf("SIP/100-096edbe0", "0?report") in new stack
-- Executing [s@macro-user-callerid:4] ExecIf("SIP/100-096edbe0", "1|Set|REALCALLERIDNUM=100") in new stack
-- Executing [s@macro-user-callerid:5] NoOp("SIP/100-096edbe0", "REALCALLERIDNUM is 100") in new stack
-- Executing [s@macro-user-callerid:6] Set("SIP/100-096edbe0", "AMPUSER=") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/100-096edbe0", "AMPUSERCIDNAME=") in new stack
-- Executing [s@macro-user-callerid:8] GotoIf("SIP/100-096edbe0", "1?report") in new stack
-- Goto (macro-user-callerid,s,13)
-- Executing [s@macro-user-callerid:13] NoOp("SIP/100-096edbe0", "TTL: ARG1: 104") in new stack
-- Executing [s@macro-user-callerid:14] GotoIf("SIP/100-096edbe0", "0?continue") in new stack
-- Executing [s@macro-user-callerid:15] Set("SIP/100-096edbe0", "__TTL=64") in new stack
-- Executing [s@macro-user-callerid:16] GotoIf("SIP/100-096edbe0", "1?continue") in new stack
-- Goto (macro-user-callerid,s,23)
-- Executing [s@macro-user-callerid:23] NoOp("SIP/100-096edbe0", "Using CallerID "device" <100>") in new stack
-- Executing [s@macro-exten-vm:2] Set("SIP/100-096edbe0", "FROMCONTEXT=exten-vm") in new stack
-- Executing [s@macro-exten-vm:3] Set("SIP/100-096edbe0", "VMBOX=104") in new stack
-- Executing [s@macro-exten-vm:4] Set("SIP/100-096edbe0", "EXTTOCALL=104") in new stack
-- Executing [s@macro-exten-vm:5] Set("SIP/100-096edbe0", "CFUEXT=") in new stack
-- Executing [s@macro-exten-vm:6] Set("SIP/100-096edbe0", "CFBEXT=") in new stack
-- Executing [s@macro-exten-vm:7] Set("SIP/100-096edbe0", "RT=15") in new stack
-- Executing [s@macro-exten-vm:8] Macro("SIP/100-096edbe0", "record-enable|104|IN") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/100-096edbe0", "0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] AGI("SIP/100-096edbe0", "recordingcheck|20081029-100553|1225292753.8") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20081029-100553|1225292753.8: No AMPUSER db entry for 104. Not recording
-- AGI Script recordingcheck completed, returning 0
-- Executing [s@macro-record-enable:5] NoOp("SIP/100-096edbe0", "No recording needed") in new stack
-- Executing [s@macro-exten-vm:9] Macro("SIP/100-096edbe0", "dial|15|tr|104") in new stack
-- Executing [s@macro-dial:1] GotoIf("SIP/100-096edbe0", "1?dial") in new stack
-- Goto (macro-dial,s,3)
-- Executing [s@macro-dial:3] AGI("SIP/100-096edbe0", "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
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
dialparties.agi: Caller ID name is 'device' number is '100'
dialparties.agi: Methodology of ring is 'none'
-- dialparties.agi: Added extension 104 to extension map
-- dialparties.agi: Extension 104 cf is disabled
-- dialparties.agi: Extension 104 do not disturb is disabled
dialparties.agi: Extension 104 has ExtensionState: -1
-- dialparties.agi: Checking CW and CFB status for extension 104
dialparties.agi: Failed to DbSet CALLTRACE/104 to 100 (0)
-- dialparties.agi: Filtered ARG3: 104
dialparties.agi: Setting default NOANSWER DIALSTATUS since no extensions available
== Manager 'admin' logged off from 127.0.0.1
-- AGI Script dialparties.agi completed, returning 0
-- Executing [s@macro-dial:4] NoOp("SIP/100-096edbe0", "Returned from dialparties with no extensions to call and DIALSTATUS: NOANSWER") in new stack
-- Executing [s@macro-exten-vm:10] Set("SIP/100-096edbe0", "SV_DIALSTATUS=NOANSWER") in new stack
-- Executing [s@macro-exten-vm:11] GosubIf("SIP/100-096edbe0", "0?docfu|1") in new stack
-- Executing [s@macro-exten-vm:12] GosubIf("SIP/100-096edbe0", "0?docfb|1") in new stack
-- Executing [s@macro-exten-vm:13] Set("SIP/100-096edbe0", "DIALSTATUS=NOANSWER") in new stack
-- Executing [s@macro-exten-vm:14] NoOp("SIP/100-096edbe0", "Voicemail is 104") in new stack
-- Executing [s@macro-exten-vm:15] GotoIf("SIP/100-096edbe0", "0?s-NOANSWER|1") in new stack
-- Executing [s@macro-exten-vm:16] NoOp("SIP/100-096edbe0", "Sending to Voicemail box 104") in new stack
-- Executing [s@macro-exten-vm:17] Macro("SIP/100-096edbe0", "vm|104|NOANSWER") in new stack
-- Executing [s@macro-vm:1] Macro("SIP/100-096edbe0", "user-callerid|SKIPTTL") in new stack
-- Executing [s@macro-user-callerid:1] NoOp("SIP/100-096edbe0", "user-callerid: device 100") in new stack
-- Executing [s@macro-user-callerid:2] Set("SIP/100-096edbe0", "AMPUSER=100") in new stack
-- Executing [s@macro-user-callerid:3] GotoIf("SIP/100-096edbe0", "0?report") in new stack
-- Executing [s@macro-user-callerid:4] ExecIf("SIP/100-096edbe0", "0|Set|REALCALLERIDNUM=100") in new stack
-- Executing [s@macro-user-callerid:5] NoOp("SIP/100-096edbe0", "REALCALLERIDNUM is 100") in new stack
-- Executing [s@macro-user-callerid:6] Set("SIP/100-096edbe0", "AMPUSER=") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/100-096edbe0", "AMPUSERCIDNAME=") in new stack
-- Executing [s@macro-user-callerid:8] GotoIf("SIP/100-096edbe0", "1?report") in new stack
-- Goto (macro-user-callerid,s,13)
-- Executing [s@macro-user-callerid:13] NoOp("SIP/100-096edbe0", "TTL: 64 ARG1: SKIPTTL") in new stack
-- Executing [s@macro-user-callerid:14] GotoIf("SIP/100-096edbe0", "1?continue") in new stack
-- Goto (macro-user-callerid,s,23)
-- Executing [s@macro-user-callerid:23] NoOp("SIP/100-096edbe0", "Using CallerID "device" <100>") in new stack
-- Executing [s@macro-vm:2] Set("SIP/100-096edbe0", "VMGAIN=""") in new stack
-- Executing [s@macro-vm:3] GotoIf("SIP/100-096edbe0", "1?vmx|1") in new stack
-- Goto (macro-vm,vmx,1)
-- Executing [vmx@macro-vm:1] GotoIf("SIP/100-096edbe0", "0?s-NOANSWER|1") in new stack
-- Executing [vmx@macro-vm:2] Set("SIP/100-096edbe0", "MODE=unavail") in new stack
-- Executing [vmx@macro-vm:3] GotoIf("SIP/100-096edbe0", "1?notdirect") in new stack
-- Goto (macro-vm,vmx,5)
-- Executing [vmx@macro-vm:5] NoOp("SIP/100-096edbe0", "Checking if ext 104 is enabled: ") in new stack
-- Executing [vmx@macro-vm:6] GotoIf("SIP/100-096edbe0", "1?s-NOANSWER|1") in new stack
-- Goto (macro-vm,s-NOANSWER,1)
-- Executing [s-NOANSWER@macro-vm:1] Macro("SIP/100-096edbe0", "get-vmcontext|104") in new stack
-- Executing [s@macro-get-vmcontext:1] Set("SIP/100-096edbe0", "VMCONTEXT=") in new stack
-- Executing [s@macro-get-vmcontext:2] GotoIf("SIP/100-096edbe0", "1?200:300") in new stack
-- Goto (macro-get-vmcontext,s,200)
-- Executing [s@macro-get-vmcontext:200] Set("SIP/100-096edbe0", "VMCONTEXT=default") in new stack
-- Executing [s-NOANSWER@macro-vm:2] VoiceMail("SIP/100-096edbe0", "104@default|u") in new stack
-- <SIP/100-096edbe0> Playing 'vm-theperson' (language 'en')
-- <SIP/100-096edbe0> Playing 'digits/1' (language 'en')
-- <SIP/100-096edbe0> Playing 'digits/0' (language 'en')
-- <SIP/100-096edbe0> Playing 'digits/4' (language 'en')
-- <SIP/100-096edbe0> Playing 'vm-isunavail' (language 'en')
-- <SIP/100-096edbe0> Playing 'vm-intro' (language 'en')
== Spawn extension (macro-vm, s-NOANSWER, 2) exited non-zero on 'SIP/100-096edbe0' in macro 'vm'
== Spawn extension (macro-vm, s-NOANSWER, 2) exited non-zero on 'SIP/100-096edbe0' in macro 'exten-vm'
== Spawn extension (macro-vm, s-NOANSWER, 2) exited non-zero on 'SIP/100-096edbe0'

And the call is lost.

What's happen?

Thank you in advance.
 

danardf

Joined
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#2
Hi.

Try to enable the 'Call waiting' on each extension.
Or maybe you don't set the codecs. (ulaw, gsm, or other)
Set disallow all , and allow ulaw on eache extension.
 

seaq

Joined
Sep 17, 2008
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#3
Hi;

I do this; but the same message continues appearing and the call is lost:

-- Executing [505@from-internal:1] Macro("SIP/606-b7701450", "exten-vm|novm|505") in new stack
-- Executing [s@macro-exten-vm:1] Macro("SIP/606-b7701450", "user-callerid") in new stack
-- Executing [s@macro-user-callerid:1] NoOp("SIP/606-b7701450", "user-callerid: device 606") in new stack
-- Executing [s@macro-user-callerid:2] Set("SIP/606-b7701450", "AMPUSER=606") in new stack
-- Executing [s@macro-user-callerid:3] GotoIf("SIP/606-b7701450", "0?report") in new stack
-- Executing [s@macro-user-callerid:4] ExecIf("SIP/606-b7701450", "1|Set|REALCALLERIDNUM=606") in new stack
-- Executing [s@macro-user-callerid:5] NoOp("SIP/606-b7701450", "REALCALLERIDNUM is 606") in new stack
-- Executing [s@macro-user-callerid:6] Set("SIP/606-b7701450", "AMPUSER=") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/606-b7701450", "AMPUSERCIDNAME=") in new stack
-- Executing [s@macro-user-callerid:8] GotoIf("SIP/606-b7701450", "1?report") in new stack
-- Goto (macro-user-callerid,s,13)
-- Executing [s@macro-user-callerid:13] NoOp("SIP/606-b7701450", "TTL: ARG1: novm") in new stack
-- Executing [s@macro-user-callerid:14] GotoIf("SIP/606-b7701450", "0?continue") in new stack
-- Executing [s@macro-user-callerid:15] Set("SIP/606-b7701450", "__TTL=64") in new stack
-- Executing [s@macro-user-callerid:16] GotoIf("SIP/606-b7701450", "1?continue") in new stack
-- Goto (macro-user-callerid,s,23)
-- Executing [s@macro-user-callerid:23] NoOp("SIP/606-b7701450", "Using CallerID "device" <606>") in new stack
-- Executing [s@macro-exten-vm:2] Set("SIP/606-b7701450", "FROMCONTEXT=exten-vm") in new stack
-- Executing [s@macro-exten-vm:3] Set("SIP/606-b7701450", "VMBOX=novm") in new stack
-- Executing [s@macro-exten-vm:4] Set("SIP/606-b7701450", "EXTTOCALL=505") in new stack
-- Executing [s@macro-exten-vm:5] Set("SIP/606-b7701450", "CFUEXT=") in new stack
-- Executing [s@macro-exten-vm:6] Set("SIP/606-b7701450", "CFBEXT=") in new stack
-- Executing [s@macro-exten-vm:7] Set("SIP/606-b7701450", "RT=""") in new stack
-- Executing [s@macro-exten-vm:8] Macro("SIP/606-b7701450", "record-enable|505|IN") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/606-b7701450", "0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] AGI("SIP/606-b7701450", "recordingcheck|20081029-171624|1225318584.16") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20081029-171624|1225318584.16: No AMPUSER db entry for 505. Not recording
-- AGI Script recordingcheck completed, returning 0
-- Executing [s@macro-record-enable:5] NoOp("SIP/606-b7701450", "No recording needed") in new stack
-- Executing [s@macro-exten-vm:9] Macro("SIP/606-b7701450", "dial||tr|505") in new stack
-- Executing [s@macro-dial:1] GotoIf("SIP/606-b7701450", "1?dial") in new stack
-- Goto (macro-dial,s,3)
-- Executing [s@macro-dial:3] AGI("SIP/606-b7701450", "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
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
dialparties.agi: Caller ID name is 'device' number is '606'
dialparties.agi: Methodology of ring is 'none'
-- dialparties.agi: Added extension 505 to extension map
-- dialparties.agi: Extension 505 cf is disabled
-- dialparties.agi: Extension 505 do not disturb is disabled
dialparties.agi: Extension 505 has ExtensionState: -1
-- dialparties.agi: Checking CW and CFB status for extension 505
dialparties.agi: Failed to DbSet CALLTRACE/505 to 606 (0)
-- dialparties.agi: Filtered ARG3: 505
dialparties.agi: Setting default NOANSWER DIALSTATUS since no extensions available
== Manager 'admin' logged off from 127.0.0.1
-- AGI Script dialparties.agi completed, returning 0
-- Executing [s@macro-dial:4] NoOp("SIP/606-b7701450", "Returned from dialparties with no extensions to call and DIALSTATUS: NOANSWER") in new stack
-- Executing [s@macro-exten-vm:10] Set("SIP/606-b7701450", "SV_DIALSTATUS=NOANSWER") in new stack
-- Executing [s@macro-exten-vm:11] GosubIf("SIP/606-b7701450", "0?docfu|1") in new stack
-- Executing [s@macro-exten-vm:12] GosubIf("SIP/606-b7701450", "0?docfb|1") in new stack
-- Executing [s@macro-exten-vm:13] Set("SIP/606-b7701450", "DIALSTATUS=NOANSWER") in new stack
-- Executing [s@macro-exten-vm:14] NoOp("SIP/606-b7701450", "Voicemail is novm") in new stack
-- Executing [s@macro-exten-vm:15] GotoIf("SIP/606-b7701450", "1?s-NOANSWER|1") in new stack
-- Goto (macro-exten-vm,s-NOANSWER,1)
-- Executing [s-NOANSWER@macro-exten-vm:1] PlayTones("SIP/606-b7701450", "congestion") in new stack
-- Executing [s-NOANSWER@macro-exten-vm:2] Congestion("SIP/606-b7701450", "10") in new stack
== Spawn extension (macro-exten-vm, s-NOANSWER, 2) exited non-zero on 'SIP/606-b7701450' in macro 'exten-vm'
== Spawn extension (macro-exten-vm, s-NOANSWER, 2) exited non-zero on 'SIP/606-b7701450'


Thanks again.
 

danardf

Joined
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#4
Some questions:

  • what do you use like extension?
  • What's your config (sip.conf, sip_additional.conf)
  • What's your config network?
 

seaq

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#5
Hi;

- I'm using Softphones Zoiper and X-Lite, and ip phones Linksys SPA922 and Polycom IP550 .

- LAN: 192.168.249.0/24. Gateway: 192.168.249.249. No VLAN.

- I've reinstalled the Elastix.

I will to try again with this new installation, and i will comment you about that.

Thanks in advance.
 

danardf

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#6
Normally, it's good!

You must just configure the network with Elastix GUI (default gatway, DNS, host name).

That's all.
 

seaq

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#7
Hi;

I've intalled the Elastix 1.1 release; because with the version 1.3 don't was possible to work in a HP Proliant DL385G2 server. I think that the Elastix 1.3 maybe require a hardware more compliant.

Zaptel load without problem, and Asterisk too.

Now, i don't have troubles in the calls between extensions, and to/from PSTN.

I'll comment you about that.

Thanks in advance.
 

seaq

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#8
It's a possible bug in Elastix 1.3?


Thanks again.
 

danardf

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#9
I don't know why!

I made a update from 1.2 to 1.3.2, it's ok.

Maybe there is some problems with the hard configuration.
If the version 1.1 work fine, you can try to update simply to 1.3.2 version.

Centos can do not have all the drivers. :unsure:
 

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