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 _________________________________________________________________ Charla con tus amigos en l?nea mediante MSN Messenger: http://messenger.latam.msn.com/
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. > > GuillermoIt 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
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@coppice.org> >Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion ><asterisk-users@lists.digium.com> >To: Asterisk Users Mailing List - Non-Commercial Discussion ><asterisk-users@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@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@lists.digium.com>; Wed, 20 Oct 2004 08:28:10 +0800 >X-Message-Info: N4u0pqWW+O2Y5O/k9cqyEY5B9GfpUqLNW6oUwpdkovk>X-Original-To: asterisk-users@lists.digium.com >Delivered-To: asterisk-users@lists.digium.com >Message-ID: <4175B119.7020504@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@hotmail.com> >In-Reply-To: <BAY14-F36IBXEHt3V5g0000cc18@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@coppice.org> [65/3] X-BeenThere: asterisk-users@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@lists.digium.com?subject=unsubscribe> >List-Archive: <http://lists.digium.com/pipermail/asterisk-users> >List-Post: <mailto:asterisk-users@lists.digium.com> >List-Help: <mailto:asterisk-users-request@lists.digium.com?subject=help> >List-Subscribe: ><http://lists.digium.com/mailman/listinfo/asterisk-users>,<mailto:asterisk-users-request@lists.digium.com?subject=subscribe> >Errors-To: asterisk-users-bounces@lists.digium.com >Return-Path: asterisk-users-bounces@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@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 l?nea mediante MSN Messenger: http://messenger.latam.msn.com/
Steve: Thanks about the explanation. I'm rather new to all of this digital telephony world. I'm a computer networks guy :) If I understood well, the transfer limitation isn?t a MFC/R2 one, but a PSTN one?. Can I transfer calls using the PBX call control even in R2 if the PBX support it? Flash didn?t work because it isnt a Zap FXO signalled interface, but even commenting the condition in the code it didn?t work in the UniCall channel. And Transfer, as said, does nothing at all. If some code is missing in app_flash.c (and the PBX support it) I can patch the code, but I need to know if it is supported by UniCall library (or R2). Regarding outgoing calls, I'd 2 problems, the timeout after 1 ring or aroung 4 seconds (it still remains), and a problem with the codec. Apparently alaw wasn?t selected (despite some code tracing showing codec 8 (alaw) was used) and audio was corrupted, and MF codes were sometimes misunderstood because that. I?ve solved it forcing the alaw use in the asterisk code, but now I feel dirty :) The new code will solve this problem too?. Incoming calls work fine (they even show "alaw" as the default codec after a call) and if I use a previously used channel with the codec set for an outgoing call, the sound is OK even with the unmodified code. Thanks for your time and work Guillermo> >Guillermo Freige wrote: > >>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. > >FXO signaling cannot reroute the call. You are relying on * to do that >work, as an extension of the usual capabilities of FXO signaling. If your >channel bank were connected directly to the PSTN it would have the same >limitations as your MFC/R2 setup. > >>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. > >An operator could take control of the call and reroute it, but I'm not sure >how you would alert the operator and get them involved. You say you are >using MFC/R2 with a PBX, rather than the PSTN. The PBX might be able to >offer you some help, if it supports call control by DTMF recall. An MFC/R2 >connection to the PSTN would definitely not. I've never used Merdians with >R2, so I have no idea of their capabilities. > >>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. > >Thanks. That's another data point I have about what works, and what does >not. :-) > >Regards, >Steve > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@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 l?nea mediante MSN Messenger: http://messenger.latam.msn.com/
Steve: You wrote:>An operator could take control of the call and reroute it, but I'm not sure >how you would alert the operator and get them involved.That gaves me an idea. The asterisk box will have also 12 analog FXO signalled lines, so if an operator can reroute a call, Asterisk can act as an "operator", using one of those lines to reroute the call to it (freeing the R2 channel), and then flash it and transfer to the actual operators queue (freeing the analog line). I'm right or it?s just a nonsense? Guillermo _________________________________________________________________ Charla con tus amigos en l?nea mediante MSN Messenger: http://messenger.latam.msn.com/