asterisk users - Jan 2006 - 3RD REQUEST - Any Help Is Appreciated

Is there a protocol I'm supposed to use here?  It seems that people are
asking 100 questions a day and SOMEONE is helping them, and I've posted this
three times and not even an "I Don't Know".

My third repost:

Ok, I've been trying to figure out why my A@H won't answer the lines
when I can call out and the panel shows the call coming in - well something
bizarre has happened.
I set up inbound routing to ring my extension if a call comes in - and my
extension rings but when I pick it up I get a dial tone. The whole time after I
answer I hear the phone I originated the call on just ring and ring and ring,
even though I answer the IP phone....
Ok, so then I set it to go to VM, and it does - but it's just a dial tone. 
So, why would the originating phone ring and ring if the PBX is picking up and
routing? And why would I get dial tone on the answering phone when the incoming
call rings to it?
Bizarre! 
Here is the real time status from CLI: 
asterisk1*CLI> 
-- Starting simple switch on 'Zap/2-1' 
-- Executing SetVar("Zap/2-1", "FROM_DID=s") in new stack 
-- Executing Answer("Zap/2-1", "") in new stack 
-- Executing Wait("Zap/2-1", "0") in new stack 
-- Executing Goto("Zap/2-1", "ext-local|*101|1") in new
stack
-- Goto (ext-local,*101,1) 
-- Executing Macro("Zap/2-1", "vm|101") in new stack 
-- Executing Macro("Zap/2-1", "user-callerid") in new stack 
-- Executing DBget("Zap/2-1", "AMPUSER=DEVICE//user") in new
stack
-- DBget: varname=AMPUSER, family=DEVICE, key=/user 
-- DBget: Value not found in database. 
-- Executing DBget("Zap/2-1",
"AMPUSERCIDNAME=AMPUSER//cidname") in new stack
-- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=/cidname 
-- DBget: Value not found in database. 
-- Executing GotoIf("Zap/2-1", "1?5") in new stack 
-- Goto (macro-user-callerid,s,5) 
-- Executing NoOp("Zap/2-1", "Using CallerID ") in new stack
-- Executing Goto("Zap/2-1", "s-|1") in new stack 
-- Goto (macro-vm,s-,1) 
-- Executing VoiceMail("Zap/2-1", "u101") in new stack 
-- Playing '/var/spool/asterisk/voicemail/default/101/unavail' (language
'en')
-- Playing 'vm-intro' (language 'en') 
-- Playing 'beep' (language 'en') 
-- Recording the message 
-- x=0, open writing: /var/spool/asterisk/voicemail/default/101/INBOX/msg0000
format: wav49, 0x9f56790
-- x=1, open writing: /var/spool/asterisk/voicemail/default/101/INBOX/msg0000
format: wav, 0x9f73680
Any clues?


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casasterisk@valnet.com a ?crit :
>Is there a protocol I'm supposed to use here?  It seems that people are
asking 100 questions a day and SOMEONE is helping them, and I've posted this
three times and not even an "I Don't Know".
>  
>
You know, if thoushands of people had to answer "I don't know", it
would
blow a bit. Your other options are to check #asterisk on freenode, or 
hire a consultant.

BTW: I don't know. Sorry :(

Cheers,
Jean-Michel.

On Fri, 2006-01-06 at 08:47 -0700, casasterisk@valnet.com
wrote:
> Is there a protocol I'm supposed to use here?  It seems that people are
asking 100 questions a day and SOMEONE is helping them, and I've posted this
three times and not even an "I Don't Know".
> 
> My third repost:
> 
> Ok, I've been trying to figure out why my A@H won't answer the
lines when I can call out and the panel shows the call coming in - well
something bizarre has happened.
> I set up inbound routing to ring my extension if a call comes in - and my
extension rings but when I pick it up I get a dial tone. The whole time after I
answer I hear the phone I originated the call on just ring and ring and ring,
even though I answer the IP phone....
> Ok, so then I set it to go to VM, and it does - but it's just a dial
tone.
> So, why would the originating phone ring and ring if the PBX is picking up
and routing? And why would I get dial tone on the answering phone when the
incoming call rings to it?
> Bizarre! 
> Here is the real time status from CLI: 
> asterisk1*CLI> 
> -- Starting simple switch on 'Zap/2-1' 
> -- Executing SetVar("Zap/2-1", "FROM_DID=s") in new
stack
> -- Executing Answer("Zap/2-1", "") in new stack 
> -- Executing Wait("Zap/2-1", "0") in new stack 
> -- Executing Goto("Zap/2-1", "ext-local|*101|1") in new
stack
> -- Goto (ext-local,*101,1) 
> -- Executing Macro("Zap/2-1", "vm|101") in new stack 
> -- Executing Macro("Zap/2-1", "user-callerid") in new
stack
> -- Executing DBget("Zap/2-1", "AMPUSER=DEVICE//user")
in new stack
> -- DBget: varname=AMPUSER, family=DEVICE, key=/user 
> -- DBget: Value not found in database. 
> -- Executing DBget("Zap/2-1",
"AMPUSERCIDNAME=AMPUSER//cidname") in new stack
> -- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=/cidname 
> -- DBget: Value not found in database. 
> -- Executing GotoIf("Zap/2-1", "1?5") in new stack 
> -- Goto (macro-user-callerid,s,5) 
> -- Executing NoOp("Zap/2-1", "Using CallerID ") in new
stack
> -- Executing Goto("Zap/2-1", "s-|1") in new stack 
> -- Goto (macro-vm,s-,1) 
> -- Executing VoiceMail("Zap/2-1", "u101") in new stack 
> -- Playing '/var/spool/asterisk/voicemail/default/101/unavail'
(language 'en')
> -- Playing 'vm-intro' (language 'en') 
> -- Playing 'beep' (language 'en') 
> -- Recording the message 
> -- x=0, open writing:
/var/spool/asterisk/voicemail/default/101/INBOX/msg0000 format: wav49, 0x9f56790
> -- x=1, open writing:
/var/spool/asterisk/voicemail/default/101/INBOX/msg0000 format: wav, 0x9f73680
> Any clues?
You'd probably do better to ask on the A@H list.

Pete

People don't usually respond with "I don't know". They just
don't respond
unless they can help. This helps reduce the clutter on the list. And for the
record I do not have an answer to this issue.

casasterisk@valnet.com wrote:
>Is there a protocol I'm supposed to use here?  It seems that people are
asking 100 questions a day and SOMEONE is helping them, and I've posted this
three times and not even an "I Don't Know".
>
>My third repost:
>
>Ok, I've been trying to figure out why my A@H won't answer the lines
when I can call out and the panel shows the call coming in - well something
bizarre has happened.
>I set up inbound routing to ring my extension if a call comes in - and my
extension rings but when I pick it up I get a dial tone. The whole time after I
answer I hear the phone I originated the call on just ring and ring and ring,
even though I answer the IP phone....
>Ok, so then I set it to go to VM, and it does - but it's just a dial
tone.
>So, why would the originating phone ring and ring if the PBX is picking up
and routing? And why would I get dial tone on the answering phone when the
incoming call rings to it?
>Bizarre! 
>Here is the real time status from CLI: 
>asterisk1*CLI> 
>-- Starting simple switch on 'Zap/2-1' 
>-- Executing SetVar("Zap/2-1", "FROM_DID=s") in new
stack
>-- Executing Answer("Zap/2-1", "") in new stack 
>-- Executing Wait("Zap/2-1", "0") in new stack 
>-- Executing Goto("Zap/2-1", "ext-local|*101|1") in new
stack
>-- Goto (ext-local,*101,1) 
>-- Executing Macro("Zap/2-1", "vm|101") in new stack 
>-- Executing Macro("Zap/2-1", "user-callerid") in new
stack
>-- Executing DBget("Zap/2-1", "AMPUSER=DEVICE//user") in
new stack
>-- DBget: varname=AMPUSER, family=DEVICE, key=/user 
>-- DBget: Value not found in database. 
>-- Executing DBget("Zap/2-1",
"AMPUSERCIDNAME=AMPUSER//cidname") in new stack
>-- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=/cidname 
>-- DBget: Value not found in database. 
>-- Executing GotoIf("Zap/2-1", "1?5") in new stack 
>-- Goto (macro-user-callerid,s,5) 
>-- Executing NoOp("Zap/2-1", "Using CallerID ") in new
stack
>-- Executing Goto("Zap/2-1", "s-|1") in new stack 
>-- Goto (macro-vm,s-,1) 
>-- Executing VoiceMail("Zap/2-1", "u101") in new stack 
>-- Playing '/var/spool/asterisk/voicemail/default/101/unavail'
(language 'en')
>-- Playing 'vm-intro' (language 'en') 
>-- Playing 'beep' (language 'en') 
>-- Recording the message 
>-- x=0, open writing:
/var/spool/asterisk/voicemail/default/101/INBOX/msg0000 format: wav49, 0x9f56790
>-- x=1, open writing:
/var/spool/asterisk/voicemail/default/101/INBOX/msg0000 format: wav, 0x9f73680
>Any clues?
>
>
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-- 
  
ISC Network Engineering
The University of Pennsylvania
3401 Walnut Street, Suite 221A
Philadelphia, PA 19104  


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casasterisk@valnet.com wrote:
> Is there a protocol I'm supposed to use here?  It seems that people are
asking 100 questions a day and SOMEONE is helping them, and I've posted this
three times and not even an "I Don't Know".
> 
> My third repost:
> 
> Ok, I've been trying to figure out why my A@H won't answer the
lines when I can call out and the panel shows the call coming in - well
something bizarre has happened.
You should be posting to the A@H Help forum:

http://sourceforge.net/forum/forum.php?forum_id=420324

or the AMP Help forum:

http://sourceforge.net/forum/forum.php?forum_id=414452

or amportal-users mailing list:

http://lists.sourceforge.net/lists/listinfo/amportal-users

Regards,

-- 
Jason Becker
Director & CEO
Coalescent Systems Inc.
Enabling Open Source Telephony
403.244.8089
www.coalescentsystems.ca
www.gabcast.com

> >-- Executing Goto("Zap/2-1", "ext-local|*101|1") in
new stack
I think the problem here is that you have the timeout set to one second
and I am not sure what the * is before 101.

My interpretation is "ext-local" specifies local context. *101 means
dial extension 101 but I am unsure of what the * is for. And the final 1
means a one second timeout.

-or-

I don't know.

Thanks,
Steve