Hi, Firstly let me start off by saying that asterisk is one of the most amazing pieces of open source I have seen, it rates right up there with Apache, OpenOffice, MySQL and even Linux itself.. Nice work!! I have just installed my first server, thanks to the astinstall script.. and I have read the Handbook (ver 1) and the white paper PDF's.. and I have managed to setup 2 extentions and make calls between them using MSN Messenger, nothing fantastic but its a start.. One answer is still missing.. How do I transfer a call to another ext?? I am looking at only using IP phones and so for the test system I am using MSN Messenger.. The final solution will probably use a linux softphone line gnophone or linphone.. All I have been able to find in the docs about call transfer is using a normal phone handset and hook-flash (not quite sure what that it, I am new to telephony).. So I guess what I am asking is what do I need to configure or do to be able to transfer a call from one IP ext to another?? Thanks.. -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze
If you search the archives you would find that for IP phone you need to add a 't' option to the end of your dial command. The 't' option will let the user dial '#' to get the systems attention, then dial an extention for the transfer. On Fri, 2003-03-14 at 03:32, =?iso-8859-1?B?V2lwZU91dCCZ ?= wrote:> Hi, > > Firstly let me start off by saying that asterisk is one of the most amazing pieces of open source I have seen, it rates right up there with Apache, OpenOffice, MySQL and even Linux itself.. Nice work!! > > I have just installed my first server, thanks to the astinstall script.. and I have read the Handbook (ver 1) and the white paper PDF's.. and I have managed to setup 2 extentions and make calls between them using MSN Messenger, nothing fantastic but its a start.. > > One answer is still missing.. How do I transfer a call to another ext?? I am looking at only using IP phones and so for the test system I am using MSN Messenger.. The final solution will probably use a linux softphone line gnophone or linphone.. > > All I have been able to find in the docs about call transfer is using a normal phone handset and hook-flash (not quite sure what that it, I am new to telephony).. > > So I guess what I am asking is what do I need to configure or do to be able to transfer a call from one IP ext to another?? > > Thanks..-- Steven Critchfield <critch@basesys.com>
What is the correct syntax to use the 't' option?? This is the current line in my extensions.conf exten => 9998,1,Dial,SIP/9998 So would I change it to exten => 9998,1,Dial,SIP/9998,t Thanks. ----- Original Message ----- From: Pertti Pikkarainen <ppik@lanwan.fi> Date: Fri, 14 Mar 2003 13:50:21 +0200 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] How to transfer a call??> > Negative side effect with 't' option: all the local SIP-to-SIP media > streams travel trough Asterisk instead of going direct. > > Right now I'm using SNOM's transfer option instead. > But now I can't use * call parking because of that. Using # is > probably better > if there are no scaling problems. > > Regards Pertti > > > > Steven Critchfield wrote: > > >If you search the archives you would find that for IP phone you need to > >add a 't' option to the end of your dial command. The 't' option will > >let the user dial '#' to get the systems attention, then dial an > >extention for the transfer. > > > >On Fri, 2003-03-14 at 03:32, =?iso-8859-1?B?V2lwZU91dCCZ ?= wrote: > > > > > >>Hi, > >> > >>Firstly let me start off by saying that asterisk is one of the most amazing pieces of open source I have seen, it rates right up there with Apache, OpenOffice, MySQL and even Linux itself.. Nice work!! > >> > >>I have just installed my first server, thanks to the astinstall script.. and I have read the Handbook (ver 1) and the white paper PDF's.. and I have managed to setup 2 extentions and make calls between them using MSN Messenger, nothing fantastic but its a start.. > >> > >>One answer is still missing.. How do I transfer a call to another ext?? I am looking at only using IP phones and so for the test system I am using MSN Messenger.. The final solution will probably use a linux softphone line gnophone or linphone.. > >> > >>All I have been able to find in the docs about call transfer is using a normal phone handset and hook-flash (not quite sure what that it, I am new to telephony).. > >> > >>So I guess what I am asking is what do I need to configure or do to be able to transfer a call from one IP ext to another?? > >> > >>Thanks.. > >> > >> > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users-- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze
Thanks the 'show application dial' was useful.. Can multiple options be specified? eg. exten => 9998,1,Dial,SIP/9998|30|t|T ----- Original Message ----- From: Pertti Pikkarainen <ppik@lanwan.fi> Date: Fri, 14 Mar 2003 15:15:14 +0200 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] How to transfer a call??> > I have it like this > > exten => 9998,1,Dial,SIP/9998|30|t > > 30 is a timeout value > Check 'show application dial' > > > WipeOut ™ wrote: > > >What is the correct syntax to use the 't' option?? > > > >This is the current line in my extensions.conf > >exten => 9998,1,Dial,SIP/9998 > >So would I change it to > >exten => 9998,1,Dial,SIP/9998,t > > > >Thanks. > > > >----- Original Message ----- > >From: Pertti Pikkarainen <ppik@lanwan.fi> > >Date: Fri, 14 Mar 2003 13:50:21 +0200 > >To: asterisk-users@lists.digium.com > >Subject: Re: [Asterisk-Users] How to transfer a call?? > > > > > > > >>Negative side effect with 't' option: all the local SIP-to-SIP media > >>streams travel trough Asterisk instead of going direct. > >> > >>Right now I'm using SNOM's transfer option instead. > >>But now I can't use * call parking because of that. Using # is > >>probably better > >>if there are no scaling problems. > >> > >>Regards Pertti > >> > >> > >> > >>Steven Critchfield wrote: > >> > >> > >> > >>>If you search the archives you would find that for IP phone you need to > >>>add a 't' option to the end of your dial command. The 't' option will > >>>let the user dial '#' to get the systems attention, then dial an > >>>extention for the transfer. > >>> > >>>On Fri, 2003-03-14 at 03:32, =?iso-8859-1?B?V2lwZU91dCCZ ?= wrote: > >>> > >>> > >>> > >>> > >>>>Hi, > >>>> > >>>>Firstly let me start off by saying that asterisk is one of the most amazing pieces of open source I have seen, it rates right up there with Apache, OpenOffice, MySQL and even Linux itself.. Nice work!! > >>>> > >>>>I have just installed my first server, thanks to the astinstall script.. and I have read the Handbook (ver 1) and the white paper PDF's.. and I have managed to setup 2 extentions and make calls between them using MSN Messenger, nothing fantastic but its a start.. > >>>> > >>>>One answer is still missing.. How do I transfer a call to another ext?? I am looking at only using IP phones and so for the test system I am using MSN Messenger.. The final solution will probably use a linux softphone line gnophone or linphone.. > >>>> > >>>>All I have been able to find in the docs about call transfer is using a normal phone handset and hook-flash (not quite sure what that it, I am new to telephony).. > >>>> > >>>>So I guess what I am asking is what do I need to configure or do to be able to transfer a call from one IP ext to another?? > >>>> > >>>>Thanks.. > >>>> > >>>> > >>>> > >>>> > >>_______________________________________________ > >>Asterisk-Users mailing list > >>Asterisk-Users@lists.digium.com > >>http://lists.digium.com/mailman/listinfo/asterisk-users > >> > >> > > > > > > > > -- > > ********************************************************************** > Nordic LAN&WAN Communication Oy > Pertti Pikkarainen > vp of engineering > E-Mail: ppik@lanwan.fi > tel: +358-9-5024100 > fax: +358-9-5023840 > gsm: +358-500-511467 > > Sinikalliontie 16 > 02630 Espoo > FINLAND > > ********************************************************************** > > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users-- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze
>I have T working here.Is your definition of T, that the caller can transfer ?? ie I P/U Zap 1 call Zap 2, & Now I press #701 & I can Park Zap2 If so is this a patch ??? pls post
I was petty sure that t and T worked from calls from one extension to another. I did notice that the caller can not transfer a call that goes to an outside line. I can double check tomorrow. --On Friday, March 14, 2003 11:45 AM -0800 TC <trclark@shaw.ca> wrote:> > >> I have T working here. > Is your definition of T, that the caller can transfer ?? > ie I P/U Zap 1 call Zap 2, & Now I press #701 & I can Park Zap2 > If so is this a patch ??? pls post > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users