Thanks Anthony it works great. The problem was my version, I was using the precompiled one but got the latest git just for that and it worked. Although to display the variable caller id you have to set on the dialplan (not on fifo.conf).<br>
<br><div class="gmail_quote">On Thu, Jan 27, 2011 at 6:35 AM, Anthony Minessale <span dir="ltr"><<a href="mailto:anthony.minessale@gmail.com">anthony.minessale@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
not the dialplan,<br>
the fifo config file:<br>
<br>
autoload_configs/fifo.conf.xml<br>
<br>
<br>
configuration name="fifo.conf" description="FIFO Configuration"><br>
<settings><br>
<param name="delete-all-outbound-member-on-startup" value="false"/><br>
</settings><br>
<fifos><br>
<fifo name="test_fifo@$${domain}" importance="0" outbound_name="sales"><br>
<member timeout="60" simo="1"<br>
lag="20">{origination_caller_id_name=fred,origination_caller_id_number=1234}user/1005@$${domain}</member><br>
</fifo><br>
</fifos><br>
</configuration><br>
<div><div></div><div class="h5"><br>
<br>
<br>
<br>
On Thu, Jan 27, 2011 at 12:55 AM, Marc de Corny <<a href="mailto:marcdecorny@gmail.com">marcdecorny@gmail.com</a>> wrote:<br>
> Hi Anthony,<br>
><br>
> I updated it the other day.<br>
><br>
> This is my version :<br>
> freeswitch@internal> version<br>
> FreeSWITCH Version 1.0.head (git-6faa4c9 2010-12-02 17-11-04 -0600)<br>
><br>
> Having said that I am using remote SIP endpoints on another SIP platform,<br>
> maybe that is why. I will take a closer look at the signalling and see.<br>
><br>
> thanks<br>
> Marc<br>
><br>
> On Thu, Jan 27, 2011 at 1:50 AM, Anthony Minessale<br>
> <<a href="mailto:anthony.minessale@gmail.com">anthony.minessale@gmail.com</a>> wrote:<br>
>><br>
>> If they made no difference you are not on latest GIT HEAD<br>
>><br>
>><br>
>> On Wed, Jan 26, 2011 at 11:38 AM, Marc de Corny <<a href="mailto:marcdecorny@gmail.com">marcdecorny@gmail.com</a>><br>
>> wrote:<br>
>> > I can see that a number of us are interested in this.<br>
>> > I have tried to set that outbound_name and fifo_outbound_name before<br>
>> > send<br>
>> > ing the call the queue but they made no difference.<br>
>> ><br>
>> > and when I do fifo list, I cannot see any variables with that name<br>
>> > available to set. Am I not looking in the right place.<br>
>> ><br>
>> > I'm looking into <action application="set"<br>
>> > data="fifo_orbit_dialplan=XML"/><br>
>> > as a potential way out, whereby as the call enters the FIFO I could<br>
>> > record<br>
>> > the outbound_name requested and then if I can control the call on the<br>
>> > way<br>
>> > out I can set it again. How does this command above allow me to send the<br>
>> > calls out a certain way and treat them. what are the options for that<br>
>> > fifo_orbit_dialplan<br>
>> ><br>
>> > thanks<br>
>> > Marc<br>
>> > On Tue, Jan 25, 2011 at 2:40 AM, liuyp2 <<a href="mailto:liuyp2@asiainfo-linkage.com">liuyp2@asiainfo-linkage.com</a>><br>
>> > wrote:<br>
>> >><br>
>> >> mod_fifo can't transfer sip header message which defined by<br>
>> >> myself(sip_h_X-xxx) to b-leg also.<br>
>> >><br>
>> >> Is there any solution in latest version?<br>
>> >><br>
>> >> ________________________________<br>
>> >> liuyp2<br>
>> >> 2011-01-25<br>
>> >> ________________________________<br>
>> >><br>
>> >> 发件人： Anthony Minessale<br>
>> >> 发送时间： 2011-01-25 09:28:25<br>
>> >> 收件人： FreeSWITCH Users Help<br>
>> >> 抄送：<br>
>> >> 主题： Re: [Freeswitch-users] Caller ID using Fifo<br>
>> >><br>
>> >><br>
>> >> You should all confer to make sure you are all using fs latest git<br>
>> >> because<br>
>> >> that is the version I am talking about. Fifo has some major new<br>
>> >> features in<br>
>> >> latest that do not exist in older versions including showing the<br>
>> >> customers<br>
>> >> cid when it calls agents. The dilemma jm describes used to be true but<br>
>> >> is<br>
>> >> no longer the case with the default ringall strategy on latest git.<br>
>> >><br>
>> >> The customers cid is sent to the agent and if the fifo xml defines<br>
>> >> outbound_name param that will be included as well.<br>
>> >><br>
>> >> If you want to override it you must do what you quoted in the wiki in<br>
>> >> the<br>
>> >> dialstring contained in the member tag of the xml for that membership<br>
>> >> not in<br>
>> >> the dialplan.<br>
>> >><br>
>> >> On Jan 14, 2011 10:36 AM, "Marc de Corny" <<a href="mailto:marcdecorny@gmail.com">marcdecorny@gmail.com</a>><br>
>> >> wrote:<br>
>> >> ><br>
>> >> > Just to follow up on this subject.<br>
>> >> ><br>
>> >> > I have done a lot of testing on the fifo trying to get the<br>
>> >> > caller_id_name changed on the outbound call to the agent and to be<br>
>> >> > honest I<br>
>> >> > cannot understand the explanation.<br>
>> >> ><br>
>> >> > If mod_fifo does not know which call it will connect until the agent<br>
>> >> > answers, how come it displays the CLI correctly, jsut won;t let me<br>
>> >> > change<br>
>> >> > it.<br>
>> >> ><br>
>> >> > Still seems strange. I am looking into the Mod_callcentre to check if<br>
>> >> > it<br>
>> >> > sends caller_id information. but the same logic if valid could apply<br>
>> >> ><br>
>> >> > Also maybe someone should change the Wiki ( I would but do not have<br>
>> >> > enough expertise on the subject) because the following is a bit<br>
>> >> > misleading<br>
>> >> ><br>
>> >> > "Note: If you wish to specify the caller ID presented when a fifo<br>
>> >> > calls<br>
>> >> > an agent, set the origination_caller_id_name and<br>
>> >> > origination_caller_id_num<br>
>> >> > variables to the values desired. These could be set within the {} of<br>
>> >> > the<br>
>> >> > dialstring, or they could be set using the set application in the<br>
>> >> > dialplan<br>
>> >> > which places the caller into the fifo (before the 'fifo in' executed<br>
>> >> > on the<br>
>> >> > caller). "<br>
>> >> > thanks<br>
>> >> > Marc<br>
>> >> > On Thu, Jan 13, 2011 at 10:47 PM, Joao Leme<br>
>> >> > <<a href="mailto:joaocarlosleme@gmail.com">joaocarlosleme@gmail.com</a>><br>
>> >> > wrote:<br>
>> >> >><br>
>> >> >> What about showing the Caller ID after it is answered? Any way to do<br>
>> >> >> that?<br>
>> >> >><br>
>> >> >> 2011/1/12 João Mesquita <<a href="mailto:jmesquita@freeswitch.org">jmesquita@freeswitch.org</a>><br>
>> >> >><br>
>> >> >>> João Leme,<br>
>> >> >>><br>
>> >> >>> The caller id is not passed when the phone is ringing because<br>
>> >> >>> mod_fifo<br>
>> >> >>> does not know which call is going to be sent to that channel once<br>
>> >> >>> it is<br>
>> >> >>> answered until it is really answered. I don't know if<br>
>> >> >>> mod_callcenter does<br>
>> >> >>> show anything but you should consider looking at the documentation<br>
>> >> >>> if you<br>
>> >> >>> really need this feature.<br>
>> >> >>><br>
>> >> >>> Regards,<br>
>> >> >>> João Mesquita<br>
>> >> >>><br>
>> >> >>><br>
>> >> >>> On Wed, Jan 12, 2011 at 9:15 PM, Joao Leme<br>
>> >> >>> <<a href="mailto:joaocarlosleme@gmail.com">joaocarlosleme@gmail.com</a>><br>
>> >> >>> wrote:<br>
>> >> >>>><br>
>> >> >>>> Hi there,<br>
>> >> >>>> I would like to know if there is a way to see the caller ID on my<br>
>> >> >>>> Sip<br>
>> >> >>>> Client (X-Lite for example) of the caller that I answear from a<br>
>> >> >>>> Fifo queue?<br>
>> >> >>>> Thanks,<br>
>> >> >>>> John<br>
>> >> >>>><br>
>> >> >>>> _______________________________________________<br>
>> >> >>>> FreeSWITCH-users mailing list<br>
>> >> >>>> <a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br>
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>> >> >>>><br>
>> >> >>>><br>
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>> >> >>>> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
>> >> >>>><br>
>> >> >>><br>
>> >> >>><br>
>> >> >>> _______________________________________________<br>
>> >> >>> FreeSWITCH-users mailing list<br>
>> >> >>> <a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br>
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>> >> >>><br>
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>> >> >>> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
>> >> >>><br>
>> >> >><br>
>> >> >><br>
>> >> >> _______________________________________________<br>
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>> >> >> <a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br>
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>> >> >><br>
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>> >> >> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
>> >> >><br>
>> >> ><br>
>> >> ><br>
>> >> > _______________________________________________<br>
>> >> > FreeSWITCH-users mailing list<br>
>> >> > <a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br>
>> >> > <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
>> >> ><br>
>> >> > UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br>
>> >> > <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
>> >> ><br>
>> >><br>
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>> >> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
>> >><br>
>> ><br>
>> ><br>
>> > _______________________________________________<br>
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>> ><br>
>> ><br>
>><br>
>><br>
>><br>
>> --<br>
>> Anthony Minessale II<br>
>><br>
>> FreeSWITCH <a href="http://www.freeswitch.org/" target="_blank">http://www.freeswitch.org/</a><br>
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>><br>
>> AIM: anthm<br>
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>><br>
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><br>
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<br>
--<br>
Anthony Minessale II<br>
<br>
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</div></div></blockquote></div><br>