[Asterisk-Users] Zap & ANSWER the Call
Robinson Tim-W10277
Tim.Robinson at motorola.com
Wed Sep 1 06:08:25 MST 2004
Some telcos execute a line polarity reversal to indicate Answer
Supervision. This might be detectable on the TDM400P cards. ISTR that
Mark said this was available although not implemented.
Other exchanges will sometimes send meter charge pulsing at 50Hz
longitudinal or 12 or 16 Khz. These options are the only clean way of
getting answer supervision on analogue loop-start circuits.
If you need to be certain of call answer supervision the only tidy
solution I know of is to use ISDN. I think some of the US T1 systems
answer supervision but I have no knowledge of this. No doubt others on
this list from 'u-law land' can contribute more wisdom.
Rgds
Tim
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Rodrigo P.
Telles
Sent: 01 September 2004 13:52
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Zap & ANSWER the Call
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Lyle,
Lyle Giese wrote:
| The standard for loop start does not send answer supervision, so * and
| all other telcom devices that do CDR records have to 'assume' that the
| call was answered.
So, it means that is impossible to say if the call was answered or not?
Someone have any idea to solve that? Is that problem is specific for
TDM400P?
Thanks for your answer.
| Lyle
|
| ----- Original Message -----
| From: "Rodrigo P. Telles" <telles at devel-it.com.br>
| To: "Asterisk Users Mailing List - Non-Commercial Discussion"
| <asterisk-users at lists.digium.com>
| Sent: Tuesday, August 31, 2004 5:31 PM
| Subject: Re: [Asterisk-Users] Zap & ANSWER the Call
|
|
|
| Hi,
|
| Nobody knows about that strange "behaviour" of Zap channels or at
| least if is that right?
|
| Thanks in advance.
|
| Rodrigo P. Telles wrote:
| | Hi,
| |
| | I'm using a TDM400 with one FXS and one FXO module (developer kit)
| | and I've been testing termination from SIP phones to PSTN and it
| | works fine, but asterisk accounting is doing something strange (for
| | me).
| | Scenario:
| | 1 - extension 1009 (SIP phone - BT101)
| | 2 - Zap/4-1 (TDM400 FXO module)
| |
| | extensions.conf:
| | [dialout]
| | exten => _9.,1,Dial(Zap/4/${EXTEN:1}|25|r)
| | exten => _9.,2,Congestion
| |
| | [sip]
| | include => dialout
| | exten => 1009,1,Dial(SIP/1009,20,rt)
| |
| | So, when I dial "9something" from 1009, something rings and then I
| | hangup the phone. I realised that asterisk thought that "something"
| | ANSWERED the call:
| |
| |
|
|> "","1009","9something","sip","""Tests""","SIP/1009-42fb","Zap/4-1","D
|> ial",
|
| | "Zap/4/something|25","2004-08-27 20:15:34","200
| | 4-08-27 20:15:36","2004-08-27 20:15:43",9,7,"ANSWERED","BILLING"
| |
| | Is that right?
| |
| | Version: Asterisk 0.9.0
| |
| | Thanks in advance.
| |
| | --
| | ============================================
| | Rodrigo P. Telles <telles at devel-it.com.br>
| | Project Manager
| | Devel-IT - http://www.devel-it.com.br
| | TDKOM Group
| | ============================================
| _______________________________________________
| Asterisk-Users mailing list
| Asterisk-Users at lists.digium.com
| http://lists.digium.com/mailman/listinfo/asterisk-users
| To UNSUBSCRIBE or update options visit:
| ~ http://lists.digium.com/mailman/listinfo/asterisk-users
|
| --
| ============================================
| Rodrigo P. Telles <telles at devel-it.com.br>
| Project Manager
| Devel-IT - http://www.devel-it.com.br
| TDKOM Group
| ============================================
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- --
============================================
Rodrigo P. Telles <telles at devel-it.com.br>
Project Manager
Devel-IT - http://www.devel-it.com.br
TDKOM Group
============================================
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