Alec Davis
2009-Nov-19 17:36 UTC
[asterisk-users] Can asterisk PRI/BRI support redirect calls
Previously incorrectly sent to asterisk-dev list, sorry. I tried today while connected to a Jtec QSIG E1 card, with DAHDISendCallreroutingFacility with the following test dialplan: Extension 4888 is on the Fujitsu [incoming] exten => 8688,1,Answer() exten => 8688,n,Playback(connecting) exten => 8688,n,DAHDISendCallreroutingFacility(4888,8688) exten => 8688,n,Playback(goodbye) With the following in chan_dahdi.conf ... context=incoming facilityenable=yes transfer=yes switchtype=qsig signalling=pri_cpe channel => 1-15,17-31 Is this how DAHDISendCallreroutingFacility is expected to be setup? After dialing into the E1 and hearing 'connecting' the result was an immediate hangup as the transfer was started, the only a warning regarding 'reason' and defaulting to unknown. The Facilty messaqge sent to the Jtec was 86 bytes long, is there a way to construct a minimal facilty message, as the Jtec debug, although I don't have it tonight, reported an error with one of the 'message types' in the facility message. I have the option of swapping out the QSIG card in the Jtec for a non QSIG card, and change to switchtype=euroisdn in chan_dahdi.conf. Would DAHDISendCallreroutingFacility then do the equivalent ETSI methods to reroute the call? I may be able to test this over the weekend, in the mean time, I thought I'd ask, if this was the correct way, or if mattf, rmudgett or others had 'team' branch that is a work in progress that we can perhaps have a look at. Alec Davis -----Original Message----- From: asterisk-dev-bounces at lists.digium.com [mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Sa?l Ibarra Sent: Thursday, 19 November 2009 12:28 a.m. To: Asterisk Developers Mailing List Subject: Re: [asterisk-dev] Can asterisk PRI/BRI support redirect calls Hi Alec, On Wed, Nov 18, 2009 at 10:42 AM, Alec Davis <sivad.a at paradise.net.nz> wrote:> We have asterisk for a small group of users in our head office, and > still a Fujtisu PBX for the majority of users. > > The request have been can we get asterisk to be the Automated > Attendant for incoming calls from the PSTN. > IE. Press > ? ?1 for sales > ? ?2 for service > ? ?3 for admin > ? ?... > ? ?0 for reception > > The answer so far is, of course asterisk can. But as I understand it > will bridge the call to the Fujitsu PABX. > > We need to transfer the call back out of asterisk down the E1 line and > to the MAIN PABX, and free up the 2 trunks used. As I understand this, > redirect the call. > > Tthe setup is as below. > > SALES ? ? ? SERVICE/ADMIN/... > ASTERISK ? ?FJPABX > ? | ? ? ? ?| > ? E1 ? ? ? E1 > ? | ? ? ? ?| > ? ISDN SWITCH (Jtec 5015 - Nice but obsolete) > ? ? ? ?| > ? ? ? ?E1 > ? ? ? ?| > ? ? ? TELCO > ? ? ? PSTN > > Developers: Guide me in the right direction, and if it's not > supported, what's the likely hood? Or do I need rmudgett on the case. >If I understood correctly what you need is called "Call path replacement" which is not currently supported in Asterisk. However, I contacted Dialogic as their Diva cards seemed to support this (according to their website). For that you need chan_dialogicdiva, which at the time I checked it did support call path replacement but NOT in NT mode. You may ask again if support has been added. Regards, -- /Sa?l http://www.saghul.net | http://www.sipdoc.net _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev