[Asterisk-Users] Tranferring UniCall lines

[Guillermo Freige]

Steve:
This means the only way to use Transfer (or Hook and DTMFSend) in a E1 is 
using it as a channel bank trunk using FXO signaling?. I really need to free 
those channels.
I'm glad the outgoing problem will be solved soon. If Transfer don't work, 
it's the only way to call the operator via a second channel.
BTW, I'm in Argentina using the local R2 variant against a Meridian 1 Option 
11C via a DTI2 card,  Asterisk is using a 410P card in E1 mode.

>From: Steve Underwood <steveu at coppice.org>
>Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion 
><asterisk-users at lists.digium.com>
>To: Asterisk Users Mailing List - Non-Commercial Discussion 
><asterisk-users at lists.digium.com>
>Subject: Re: [Asterisk-Users] Tranferring UniCall lines
>Date: Wed, 20 Oct 2004 08:28:09 +0800
>MIME-Version: 1.0
>Received: from mc9-f29.hotmail.com ([65.54.166.36]) by mc9-s21.hotmail.com 
>with Microsoft SMTPSVC(5.0.2195.6824); Tue, 19 Oct 2004 17:35:40 -0700
>Received: from lists.digium.com ([69.16.138.164]) by mc9-f29.hotmail.com 
>with Microsoft SMTPSVC(5.0.2195.6824); Tue, 19 Oct 2004 17:35:39 -0700
>Received: from [69.16.138.164] (localhost [127.0.0.1])by lists.digium.com 
>(Postfix) with ESMTPid 8AD582FE89F; Tue, 19 Oct 2004 19:30:26 -0500 (CDT)
>Received: from psmtp.com (exprod5mx120.postini.com [64.18.0.34])by 
>lists.digium.com (Postfix) with SMTP id 829382FDBD3for 
><asterisk-users at lists.digium.com>;Tue, 19 Oct 2004 19:30:23 -0500 (CDT)
>Received: from source ([202.76.92.172]) by 
>exprod5mx120.postini.com([64.18.4.10]) with SMTP; Tue, 19 Oct 2004 19:30:28 
>CDT
>Received: from [192.168.2.50] ([192.168.2.50]) (authenticated bits=0)by 
>main.coppice.org (8.12.11/8.12.8) with ESMTP id i9K0S9LA029957for 
><asterisk-users at lists.digium.com>; Wed, 20 Oct 2004 08:28:10 +0800
>X-Message-Info: N4u0pqWW+O2Y5O/k9cqyEY5B9GfpUqLNW6oUwpdkovk=
>X-Original-To: asterisk-users at lists.digium.com
>Delivered-To: asterisk-users at lists.digium.com
>Message-ID: <4175B119.7020504 at coppice.org>
>User-Agent: Mozilla/5.0 (X11; U; Linux i686; en-US; rv:1.7.3) 
>Gecko/20040922
>X-Accept-Language: en, en-us
>References: <BAY14-F36IBXEHt3V5g0000cc18 at hotmail.com>
>In-Reply-To: <BAY14-F36IBXEHt3V5g0000cc18 at hotmail.com>
>X-pstn-levels: (S:99.90000/99.90000 )
>X-pstn-settings: 1 (0.1500:0.1500) gt3 gt2 gt1 X-pstn-addresses: from 
><steveu at coppice.org> [65/3] X-BeenThere: asterisk-users at lists.digium.com
>X-Mailman-Version: 2.1.5
>Precedence: list
>List-Id: Asterisk Users Mailing List - Non-Commercial 
>Discussion<asterisk-users.lists.digium.com>
>List-Unsubscribe: 
><http://lists.digium.com/mailman/listinfo/asterisk-users>,<mailto:asterisk-users-request at lists.digium.com?subject=unsubscribe>
>List-Archive: <http://lists.digium.com/pipermail/asterisk-users>
>List-Post: <mailto:asterisk-users at lists.digium.com>
>List-Help: <mailto:asterisk-users-request at lists.digium.com?subject=help>
>List-Subscribe: 
><http://lists.digium.com/mailman/listinfo/asterisk-users>,<mailto:asterisk-users-request at lists.digium.com?subject=subscribe>
>Errors-To: asterisk-users-bounces at lists.digium.com
>Return-Path: asterisk-users-bounces at lists.digium.com
>X-OriginalArrivalTime: 20 Oct 2004 00:35:39.0673 (UTC) 
>FILETIME=[B952F090:01C4B63C]
>
>Hi Guillermo,
>
>Guillermo Freige wrote:
>
>>It's possible to use the "Transfer" function in UniCall MFC/R2 lines?. The 
>>command seems to do nothing when called from a R2 call, but it works fine 
>>from a SIP phone. Transfer and 3waycalling options are set to "Yes" in 
>>unicall.conf. I've tried the "hook" command but it didn't work either.
>>I need to make a blind transfer from a incoming line of our R2-connected 
>>PBX to another extension in the PBX, to not to waste 2 channels in a 
>>second Dial command.
>>Also I'm having trouble with outgoing calls generated in Asterisk to the 
>>PBX via the R2 line. The call hangs after a couple of seconds if answered 
>>in the first ring, or dies after the first ring if unnattended, and the 
>>channel remains in "Call" state in asterisk. Any indea?. Incoming calls 
>>work ok.
>>
>>Guillermo
>
>It is good to hear you are getting some success with my R2 code.
>
>R2 does not allow transfers of that kind. It is a protocol limitation, not 
>something about my implementation. You cannot free up the circuits. In 
>general, only the modern common channel message oriented protocols, like 
>ISDN and SS7, can do that.
>
>The outgoing call problem has been reported already. The will be a fix in a 
>few days for that, and some polishing.
>
>Which country are you in? I am trying to find which national variants of 
>the R2 protocol are being tested.
>
>Regards,
>Steve
>
>_______________________________________________
>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

_________________________________________________________________
Charla con tus amigos en línea mediante MSN Messenger: 
http://messenger.latam.msn.com/