Hi! I'm new to freeswitch, and I'm trying to make an att_xfer in a dialplan, but instead of giving a sofia/${domain}/${called_number} as <channel url>, i would like to use a loopback/${called_number}, because i would like to transfer the call not just to different extensions. Is there any way to achive this? When i'm trying to do this like that: <action application="att_xfer" data="loopback/${callednumber}"/> in a A-call->B-att_xfer->C situation, A gets hanged up, as soon as the bridge has been estabilished between B and C. Btw, i have tryed out att_xfer by giving user/${legal_user} as <channel url>. I've found att_xfer this way a kind of instable, sometimes it worked perfectly, sometimes not: A and C did not hang up, but there weren't succesfully connected (1 time from 10). The log printed the same at both cases. Cheers: Viktor Hirdetés (x) Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu - a kötelező biztosítások kiindulópontja!
Viktor, For the user channel its user/user@domain.com ... as for the rest can you show me your entire config? http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer /b On Nov 6, 2008, at 9:45 AM, x y wrote:
Hi!
I'm new to freeswitch, and I'm trying to make an att_xfer in a dialplan, but instead of giving a sofia/${domain}/${called_number} as <channel url>, i would like to use a loopback/${called_number}, because i would like to transfer the call not just to different extensions. Is there any way to achive this? When i'm trying to do this like that:
<action application="att_xfer" data="loopback/${callednumber}"/>
in a A-call->B-att_xfer->C situation, A gets hanged up, as soon as the bridge has been estabilished between B and C.
Btw, i have tryed out att_xfer by giving user/${legal_user} as <channel url>. I've found att_xfer this way a kind of instable, sometimes it worked perfectly, sometimes not: A and C did not hang up, but there weren't succesfully connected (1 time from 10). The log printed the same at both cases.
Cheers: Viktor
Hi! Thanks for the fast answer. Of course, there is user/${username}@${domain} in the config, i was just lazy or tired, so i forgot the @${domain}. So, here is the part of the config. <extension name="668([0-9]{4})"> <condition field="destination_number" expression="^668([0-9]{4})$"> <action application="bind_meta_app" data="2 b s execute_extension::xfer XML default"/> <action application="bind_meta_app" data="1 b s execute_extension::att_xfer XML default"/> <action application="bridge" data="user/$1@${domain}"/> </condition> </extension> <extension name="att_xfer"> <condition field="destination_number" expression="^att_xfer$"> <action application="read" data="1 15 /opt/freeswitch/sounds/en/us/callie/misc/8000/transfer1.wav callednumber 4000 #"/> <action application="att_xfer" data="user/${callednumber}@${domain}"/> </condition> </extension> ------------------------------------------------------------------------ or for the loopback: ------------------------------------------------------------------------ <extension name="att_xfer"> <condition field="destination_number" expression="^att_xfer$"> <action application="read" data="1 15 /opt/freeswitch/sounds/en/us/callie/misc/8000/transfer1.wav callednumber 4000 #"/> <action application="att_xfer" data="loopback/669${callednumber}"/> </condition> </extension> <extension name="connect_extension"> <condition field="destination_number" expression="^669([0-9]{4})$"> <action application="bridge" data="user/$1@${domain}"/> </condition> </extension> The connect_extension just simulates that, we execute something in the dialplan. The rest is the defult config. Btw, the xfer is a transfer with loopback channel, and it works fine. Cheers, Viktor -- Eredeti üzenet -- Feladó: Brian West <brian@freeswitch.org>Címzett: freeswitch-users@lists.freeswitch.orgMásolat: Elküldve: 2008.11.06 17:58Téma: Re: [Freeswitch-users] att_xfer+loopback Viktor, For the user channel its user/user@domain.com ... as for the rest can you show me your entire config? http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer /b On Nov 6, 2008, at 9:45 AM, x y wrote:Hi! I'm new to freeswitch, and I'm trying to make an att_xfer in a dialplan, but instead of giving a sofia/${domain}/${called_number} as <channel url>, i would like to use a loopback/${called_number}, because i would like to transfer the call not just to different extensions. Is there any way to achive this? When i'm trying to do this like that: <action application="att_xfer" data="loopback/${callednumber}"/> in a A-call->B-att_xfer->C situation, A gets hanged up, as soon as the bridge has been estabilished between B and C. Btw, i have tryed out att_xfer by giving user/${legal_user} as <channel url>. I've found att_xfer this way a kind of instable, sometimes it worked perfectly, sometimes not: A and C did not hang up, but there weren't succesfully connected (1 time from 10). The log printed the same at both cases. Cheers: Viktor _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Hirdetés (x) Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu - a kötelező biztosítások kiindulópontja!
Because I just wantet to try that the att_xfer is working with user channel. The 668 and 669 perfixes are only made to be assured about that my extensions will be processed instead of the default config's, and because of that, I could transfer the call to any dialable number in the config later on. The usage of the user channel wont help me in this matter, cos its very similar to the original sofia method in the att_xfer example on freeswitch wiki. Basicly, i just want to know that there is any option to use loopback channel in att_xfer. Cheers, Viktor Why aren't you prefixing the $1 with 668 in the first example? Hirdetés (x) Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu - a kötelező biztosítások kiindulópontja!
Didn't you say in your previous email using loopback worked? /b On Nov 7, 2008, at 8:32 AM, x y wrote:
Because I just wantet to try that the att_xfer is working with user channel. The 668 and 669 perfixes are only made to be assured about that my extensions will be processed instead of the default config's, and because of that, I could transfer the call to any dialable number in the config later on. The usage of the user channel wont help me in this matter, cos its very similar to the original sofia method in the att_xfer example on freeswitch wiki. Basicly, i just want to know that there is any option to use loopback channel in att_xfer.
Cheers, Viktor
Wich is working with loopback, is transfer: <extension name="xfer"> <condition field="destination_number" expression="^xfer$"> <action application="read" data="1 15 /opt/freeswitch/sounds/en/us/callie/misc/8000/transfer2.wav callednumber 7000 #"/> <action application="transfer" data="-bleg 664${callednumber}"/> </condition> </extension> <extension name="loop"> <condition field="destination_number" expression="^664([0-9]{4})$"> <action application="bridge" data="loopback/669$1"/> </condition> </extension> (I know there is no need on that loopback in the loop extension, it's just an experimental config about loopback and transfers.) Attended transfer is still a mistery for me. Btw, keeping up experimenting with att_xfer i have found the sofia method a little instable too. It sometimes acts like the user channel i said before. Here is the code that I use: <extension name="loop"> <condition field="destination_number" expression="^664([0-9]{4})$"> <action application="bridge" data="loopback/669$1"/> </condition> </extension> Viktor Hirdetés (x) Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu - a kötelező biztosítások kiindulópontja!
Hi! I see, I have no luck on this att_xfer with loopback channel. So, if it's not possible, is there another way to do the trick? I would like to use att_xfer just like transfer, i mean feed a phone number as a param to it, not a channel url. I tought using loopback channel will do it, but couldn't make it so far. If anybody interested, I could send a log from a failed att_xfer. I would really appreciate any help on this matter. Btw, would it be difficult to modify the att_xfer function to accept dialnumber as a param instead of channel url? I don't have any idea about this, so just asking. Cheers, Viktor Hirdetés (x) Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu - a kötelező biztosítások kiindulópontja!
to do what you want you need to use the regular "transfer" use set to set a variable say my_dest to the string you want to dial then make an extension with destination_number checks on some keyword like my_outbound <extension name="ext1"> <condition field="destination_number" expression="my_outbound"> <action application="bridge" data="${my_dest}"/> </condition> </extension> <extension name="ext1"> <condition [any condition]> <action application="set" data="my_dest=/sofia/profile/user@domain.com"/> <action application="transfer" data="my_outbound"/> </condition> </extension> On Wed, Nov 12, 2008 at 4:38 AM, x y <fs_ask_sy@citromail.hu> wrote:
Hi!
I see, I have no luck on this att_xfer with loopback channel. So, if it's not possible, is there another way to do the trick? I would like to use att_xfer just like transfer, i mean feed a phone number as a param to it, not a channel url. I tought using loopback channel will do it, but couldn't make it so far. If anybody interested, I could send a log from a failed att_xfer. I would really appreciate any help on this matter. Btw, would it be difficult to modify the att_xfer function to accept dialnumber as a param instead of channel url? I don't have any idea about this, so just asking.
Cheers, Viktor
*Hirdetés (x) * Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu<http://ad.adverticum.net/b/cl,1,2201,295040,360471/click.prm>- a kötelező biztosítások kiindulópontja!
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Thx for the answer, but i dont really understand what did u mean with transfer. Maybe I didn explain clearly enough what I would like to achive. I would like to use att_xfer, and somehow feed back the number I dial (wich I want to transfer to) back to the dialplan, and let the dialplan decide what to do with the number. I know that the regular transfer works like this, but the transfer and the att_xfer is different, and I would like to use both of them that way. With regular transfer, there was no problem. If I just misunderstood your advice, sorry. In that case, could you give me some more details? Big thanx. Cheers, Viktor Hirdetés (x) Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu - a kötelező biztosítások kiindulópontja!
put the log of the case you have a failure at http://pastebin.freeswitch.org (note the auth info is in the dialog box) On Wed, Nov 12, 2008 at 10:07 AM, x y <fs_ask_sy@citromail.hu> wrote:
Thx for the answer, but i dont really understand what did u mean with transfer. Maybe I didn explain clearly enough what I would like to achive. I would like to use att_xfer, and somehow feed back the number I dial (wich I want to transfer to) back to the dialplan, and let the dialplan decide what to do with the number. I know that the regular transfer works like this, but the transfer and the att_xfer is different, and I would like to use both of them that way. With regular transfer, there was no problem. If I just misunderstood your advice, sorry. In that case, could you give me some more details? Big thanx.
Cheers, Viktor
*Hirdetés (x) * Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu<http://ad.adverticum.net/b/cl,1,2201,295040,360471/click.prm>- a kötelező biztosítások kiindulópontja!
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
I sent the log about the failed att_xfer + loopback channel. Are u interested on failed attempts with sofia/ and user/ channel url in att_xfer too? I wrote that some mails before, sometimes it fails to connect the two channels. Hirdetés (x) Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu - a kötelező biztosítások kiindulópontja!
yes please put them on pastebin too On Wed, Nov 12, 2008 at 10:42 AM, x y <fs_ask_sy@citromail.hu> wrote:
I sent the log about the failed att_xfer + loopback channel. Are u interested on failed attempts with sofia/ and user/ channel url in att_xfer too? I wrote that some mails before, sometimes it fails to connect the two channels.
*Hirdetés (x) * Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu<http://ad.adverticum.net/b/cl,1,2201,295040,360471/click.prm>- a kötelező biztosítások kiindulópontja!
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
I've sent both of them. Hirdetés (x) Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu - a kötelező biztosítások kiindulópontja!
so you are answering the other call then pressing 0 to conference them together and then hanging up? 2008/11/12 x y <fs_ask_sy@citromail.hu>
I've sent both of them.
*Hirdetés (x) * Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu<http://ad.adverticum.net/b/cl,1,2201,295040,360471/click.prm>- a kötelező biztosítások kiindulópontja!
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
The dtmf 0 works fine with sofia/ and user/, but with loopback/ I dont even have time to make a conference like transfer, because the caller A hangs up as soon as C asnwers the call of B, instead of being hold (in a A-call->B-att_xfer->C situation), and there will be nobody to transfer to. I'm trying to do that, the two calls is bridged together when B hangs up, or A gets B back when C hangs up, depends on the situation. Hirdetés (x) Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu - a kötelező biztosítások kiindulópontja!
loopback may not be supported in every situation. You do know transfer etc is built into sip already right? you just press hold on your sip phone, call someone else and press transfer. On Thu, Nov 13, 2008 at 2:36 AM, x y <fs_ask_sy@citromail.hu> wrote:
The dtmf 0 works fine with sofia/ and user/, but with loopback/ I dont even have time to make a conference like transfer, because the caller A hangs up as soon as C asnwers the call of B, instead of being hold (in a A-call->B-att_xfer->C situation), and there will be nobody to transfer to. I'm trying to do that, the two calls is bridged together when B hangs up, or A gets B back when C hangs up, depends on the situation.
*Hirdetés (x) * Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu<http://ad.adverticum.net/b/cl,1,2201,295040,360471/click.prm>- a kötelező biztosítások kiindulópontja! _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Thx for the advices. Unfortunately, using transfer with hold wont work for me. For example, with a sip phone, wich dont have built in support for transfer. The whole situaion is that I'm using softphone, wich has a client freeswitch running in the background, and there is another freeswitch on the server, witch is responsible for the main part. I would like to process the att_xfer procedure in the client freeswitch, then just send the numbers to the server's dialplan, and let to the server's freeswitch decide what to do with it. I tried the regular transfer, and it worked fine. With att_xfer, the problem is that the client's freeswitch doesnt know where will be the call transfered, it just knows the dialnumber of it. Thats why I have tried to use loopback with att_xfer. Is there another way to that? Would it worth to create a similar att_xfer application, wich accepts dialnumbers like transfer? I checked the source both of transfer and att_xfer, hoping they are similar, but they werent for the first look, and I dont really know how much work would it cost. Hirdetés (x) Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu - a kötelező biztosítások kiindulópontja!
Wrong code has been sent before, here is the right one for sofia: <extension name="att_xfer"> <condition field="destination_number" expression="^att_xfer$"> <action application="read" data="1 15 /opt/freeswitch/sounds/en/us/callie/misc/8000/transfer1.wav callednumber 7000 #"/> <action application="att_xfer" data="sofia/${domain}/${callednumber}"/> </condition> </extension> Hirdetés (x) Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu - a kötelező biztosítások kiindulópontja!
participants (3)
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Anthony Minessale
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Brian West
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x y