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VoIP Billing solution


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 Post subject: DID not working Please Help
PostPosted: Thu Dec 28, 2006 3:46 am 
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Joined: Sun Nov 12, 2006 5:07 am
Posts: 20
Hi all,
I have been trying to get my head around this for many days so if someone can help point me in the right direction, it would be appreciated.

I want to be able to ring into an internal Sip Friend through an ISDN card.
When I dial in at present, I see in the CLI,

-- Accepting voice call from '' to '01039683440' on channel 0/1, span 1
-- Executing Set("Zap/1-1", "DID=01039683440") in new stack
-- Executing Goto("Zap/1-1", "s|1") in new stack
-- Goto (from-zaptel,s,1)
-- Executing NoOp("Zap/1-1", "Entering from-zaptel with DID == 01039683440") in new stack
-- Executing Set("Zap/1-1", "DID=01039683440") in new stack
-- Executing NoOp("Zap/1-1", "DID is now 01039683440") in new stack
-- Executing GotoIf("Zap/1-1", "1?zapok:notzap") in new stack
-- Goto (from-zaptel,s,7)
-- Executing NoOp("Zap/1-1", "Is a Zaptel Channel") in new stack
-- Executing Set("Zap/1-1", "CHAN=1-1") in new stack
-- Executing Set("Zap/1-1", "CHAN=1") in new stack
-- Executing Macro("Zap/1-1", "from-zaptel-1|01039683440|1") in new stack
-- Executing NoOp("Zap/1-1", "Returned from Macro from-zaptel-1") in new stack
-- Executing Goto("Zap/1-1", "ext-did|01039683440|1") in new stack
-- Goto (ext-did,01039683440,1)
-- Executing NoOp("Zap/1-1", "Catch-All DID Match - Found 01039683440 - You probably want a DID for this.") in new stack
-- Executing Goto("Zap/1-1", "ext-did|s|1") in new stack
-- Goto (ext-did,s,1)
-- Executing Set("Zap/1-1", "FROM_DID=s") in new stack
-- Executing Set("Zap/1-1", "FAX_RX=disabled") in new stack
-- Executing Goto("Zap/1-1", "custom-callingcard|s|1") in new stack
-- Goto (custom-callingcard,s,1)
-- Hungup 'Zap/1-1'

So from this I can see that the inbound call is going to "from-zaptel" and is then passed to "ext-did"
In ext-did in extensions.additional.conf I find
[ext-did]
include => ext-did-custom
exten => s,1,Set(FROM_DID=s)
exten => s,n,Set(FAX_RX=disabled)
exten => s,n,Goto(custom-callingcard,s,1)
exten => _X.,1,Noop(Catch-All DID Match - Found ${EXTEN} - You probably want a DID for this.)
exten => _X.,n,Goto(ext-did,s,1)

So I have setup "ext-did-custom" in extensions.custom.conf
[ext-did-custom]
exten => 01039683440,1,Set(FROM_DID=01039683440)
exten => 01039683445,1,Goto(my-did-custom,01039683440,1)

[my-did-custom]
exten => _X.,1,DeadAGI(a2billing.php|2|did)
exten => _X.,2,Hangup

But it doesn't work. Can someone Please help.
I am still getting the exact same messages in the Asterisk CLI as previously posted. I did try putting the above extensions directly into the ext-did and this then still did the same but in the Asterisk CLI it doesn't say You probably want a DID for this, it just seems to carry on and end the same way.
When I dial this number from an external phone, it rings twice and hangs up. The Sip extension it should ring never does ring at all.
I have also been into the Asterisk A2Billing pages on setup the DID and the List Destination.

Any help would be appreciated.


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 Post subject:
PostPosted: Sat Dec 30, 2006 12:23 am 
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Joined: Sun Nov 12, 2006 5:07 am
Posts: 20
Hi all,
I found that if I put this line at the top of the
[ext-did]
exten => _X.,n,Goto(custom-callingcard,${EXTEN},1)

It now works, can't hear the conversation but that is probably a nat issue.
Nothing is ever as easy as it seems.
Anyway just thought this may be useful for someone else with similar issue.


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 Post subject:
PostPosted: Sat Dec 30, 2006 4:54 am 
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Joined: Sun Nov 12, 2006 5:07 am
Posts: 20
Help,
Hi all, when I use a remote phone - I can now dial anyone and it works ok.
But when I use DID and dial a number which connects to a sip friend, communications is only one way. The external user can talk to the sip friend but the external user cannot hear anything the sip friend says.

If I use the sip friend's phone then I can call anyone and the conversation works both ways.
This tells me that the ports must be working.

I don't understand what could be wrong,
Does anyone please have any idea's


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 Post subject:
PostPosted: Sat Dec 30, 2006 5:34 am 
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Joined: Mon May 29, 2006 7:07 pm
Posts: 287
Location: Denver
Strange maybe check the codec's being used. Look at debug logs to see if you have any transcoding problems.


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VoIP Billing solution


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