On Tue, Oct 25, 2005 at 12:31:41PM -0200, [EMAIL PROTECTED] wrote: > Hi Pablo!
ok. i do all the changes but now i have this error -- Channel 0/1, span 1 got hangup Oct 25 11:46:40 WARNING[3639]: app_dial.c:416 wait_for_answer: Unable to forward voice -- Hungup 'Zap/1-1' == No one is available to answer at this time -- Executing Playback("SIP/205-0014", "invalid") in new stack -- Playing 'invalid' (language 'en') == Spawn extension (from-internal, 9122, 2) exited non-zero on 'SIP/205-0014' maybe is a extensions.conf ?? can you paste your extensions.conf here please? > > I understood your problem. It is related to Siemens PBX. > > With this topology, Asterisk is acting as a PSTN Central Office (a Public > Central). What you asking is something like this: > > Asterisk acting as Central Office -> HiPath -> Public Central Office > > That is: the SIP devices connected to the Asterisk are not HI-Path's > extensions! They seem "external" terminal/lines. > > So... > > You will have to enable, at HiPath, something called "Transit" or "External > traffic". In other words, it is a feature that you enable on HiPath allowing > traffic between two trunks (the trunk connected to Asterisk and the trunk > connected to the PSTN Central Office). > > Here we had to create a "trunk access code". So, if a Asterisk user wants to > call the outside number 5555-1234, he/she will dial: > 9 + 5555-1234 > Asterisk with then route this call to HiPath prefixing the trunk access > code, for example, "88". So, asterisk will dial: > 88 + 5555-1234 > > Hope this helps, > > --hg > ----- Original Message ----- > From: <[EMAIL PROTECTED]> > To: "Pablo Allietti" <[EMAIL PROTECTED]> > Sent: Tuesday, October 25, 2005 11:52 AM > Subject: Re: Siemens HI-path to ASTERISK > > > >Hi Pablo! > > > >I understood your problem. It is related to Siemens PBX. > > > >With this topology, Asterisk is acting as a PSTN Central Office (a Public > >Central). What you asking is something like this: > > > >Asterisk acting as Central Office -> HiPath -> Public Central Office > > > >That is: the SIP devices connected to the Asterisk are not HI-Path's > >extensions! They seem "external" terminal/lines. > > > >So... > > > >You will have to enable, at HiPath, something called "Transit" or > >"External traffic". In other words, it is a feature that you enable on > >HiPath allowing traffic between two trunks (the trunk connected to > >Asterisk and the trunk connected to the PSTN Central Office). > > > >Here we had to create a "trunk access code". So, if a Asterisk user wants > >to call the outside number 5555-1234, he/she will dial: > >9 + 5555-1234 > >Asterisk with then route this call to HiPath prefixing the trunk access > >code, for example, "88". So, asterisk will dial: > >88 + 5555-1234 > > > >Hope this helps, > > > >Huelbe. > > > >----- Original Message ----- > >From: "Pablo Allietti" <[EMAIL PROTECTED]> > >To: <[EMAIL PROTECTED]> > >Sent: Tuesday, October 25, 2005 12:41 PM > >Subject: Re: Siemens HI-path to ASTERISK > > > > > >>On Mon, Oct 24, 2005 at 06:42:02PM -0200, [EMAIL PROTECTED] > >>wrote: > >>>Yes, for sure, it works. With TE110P from Digium using E1/ISDN/Pri > >>>signalling. > >>> > >>>By heart, I remember the following: > >>> > >>>1. Configure Siemens E1 port as "station" and Asterisk as "Pri_Net" (or > >>>Central Office). > >>> > >>>2. At Siemens, set the E1 port as "S2 Point-to-Point net line without > >>>CRC4" > >>>or something like this. > >> > >> > >>yep done. i only have a problem i can call any extension in the pbx but > >>i can't take outside line with the 9 > >> > >>you can send to me the extensions.conf please???? please///// > >> > >>> > >>>3. At Asterisk, put these lines (/etc/zaptel.conf): > >>>span=1,1,0,ccs,hdb3 > >>>bchan=1-15 > >>>dchan=16 > >>>bchan=17-31 > >>> > >>>You have to study the rest of * conf file, but these ones are the > >>>important > >>>ones. > >>> > >>>Regards, > >>> > >>>--hg > >>> > >>>----- Original Message ----- > >>>From: "Pablo Allietti" <[EMAIL PROTECTED]> > >>>To: <asterisk-users@lists.digium.com> > >>>Sent: Monday, October 24, 2005 6:55 PM > >>>Subject: [Asterisk-Users] Siemens HI-path to ASTERISK > >>> > >>> > >>>>anybody can connect a Siemens HI-PATH to ASterisk via e1 ? > >>>> > >>>>i need your help please. > >>>>_______________________________________________ > >>>>--Bandwidth and Colocation sponsored by Easynews.com -- > >>>> > >>>>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 > >>>> > >>> > >>>_______________________________________________ > >>>--Bandwidth and Colocation sponsored by Easynews.com -- > >>> > >>>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 > >>---end quoted text--- > >> > >>-- > >> > >>.- > >> > >>Pablo Allietti > >>LACNIC > >> > >> > > > > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > 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 ---end quoted text--- -- .- Pablo Allietti LACNIC _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- 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