Sponsor: VoiceMeUp - Corporate & Wholesale VoIP Services

VoIP Mailing List Archives
Mailing list archives for the VoIP community
 SearchSearch 

[Freeswitch-users] att_xfer+loopback

Goto page 1, 2  Next
 
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> freeSWITCH Users
View previous topic :: View next topic  
Author Message
fs_ask_sy at citromail.hu
Guest





PostPosted: Thu Nov 06, 2008 10:53 am    Post subject: [Freeswitch-users] att_xfer+loopback Reply with quote

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
Back to top
brian at freeswitch.org
Guest





PostPosted: Thu Nov 06, 2008 11:49 am    Post subject: [Freeswitch-users] att_xfer+loopback Reply with quote

Viktor, For the user channel its user/user@domain.com ([email]user/user@domain.com[/email]) ... 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:
Quote:
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
Back to top
fs_ask_sy at citromail.hu
Guest





PostPosted: Fri Nov 07, 2008 3:07 am    Post subject: [Freeswitch-users] att_xfer+loopback Reply with quote

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
Quote:
-- Eredeti üzenet --
Feladó: Brian West <brian@freeswitch.org> ([email]Brian%20West%20%3Cbrian@freeswitch.org%3E[/email])
Címzett: freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)
Másolat: ([email][/email])
Elküldve: 2008.11.06 17:58
Téma: Re: [Freeswitch-users] att_xfer+loopback


Viktor, For the user channel its user/user@domain.com ([email]user/user@domain.com[/email]) ... 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:
Quote:
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 (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

Back to top
brian at freeswitch.org
Guest





PostPosted: Fri Nov 07, 2008 8:58 am    Post subject: [Freeswitch-users] att_xfer+loopback Reply with quote

Why aren't you prefixing the $1 with 668 in the first example?

/b

On Nov 7, 2008, at 2:03 AM, x y wrote:
Quote:
Btw, the xfer is a transfer with loopback channel, and it works fine.
Back to top
fs_ask_sy at citromail.hu
Guest





PostPosted: Fri Nov 07, 2008 9:51 am    Post subject: [Freeswitch-users] att_xfer+loopback Reply with quote

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
Quote:
Why aren't you prefixing the $1 with 668 in the first example?


Back to top
brian at freeswitch.org
Guest





PostPosted: Fri Nov 07, 2008 10:07 am    Post subject: [Freeswitch-users] att_xfer+loopback Reply with quote

Didn't you say in your previous email using loopback worked?

/b

On Nov 7, 2008, at 8:32 AM, x y wrote:
Quote:

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
Back to top
fs_ask_sy at citromail.hu
Guest





PostPosted: Fri Nov 07, 2008 10:18 am    Post subject: [Freeswitch-users] att_xfer+loopback Reply with quote

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>
Back to top
fs_ask_sy at citromail.hu
Guest





PostPosted: Fri Nov 07, 2008 10:30 am    Post subject: [Freeswitch-users] att_xfer+loopback Reply with quote

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
Back to top
fs_ask_sy at citromail.hu
Guest





PostPosted: Wed Nov 12, 2008 5:41 am    Post subject: [Freeswitch-users] att_xfer+loopback Reply with quote

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
Back to top
anthony.minessale at g...
Guest





PostPosted: Wed Nov 12, 2008 9:28 am    Post subject: [Freeswitch-users] att_xfer+loopback Reply with quote

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 (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 (fs_ask_sy@citromail.hu)> wrote:
Quote:

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!


_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (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 ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400
Back to top
fs_ask_sy at citromail.hu
Guest





PostPosted: Wed Nov 12, 2008 11:09 am    Post subject: [Freeswitch-users] att_xfer+loopback Reply with quote

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
Back to top
anthony.minessale at g...
Guest





PostPosted: Wed Nov 12, 2008 11:40 am    Post subject: [Freeswitch-users] att_xfer+loopback Reply with quote

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 (fs_ask_sy@citromail.hu)> wrote:
Quote:

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!


_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (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 ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400
Back to top
fs_ask_sy at citromail.hu
Guest





PostPosted: Wed Nov 12, 2008 11:45 am    Post subject: [Freeswitch-users] att_xfer+loopback Reply with quote

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.
Back to top
anthony.minessale at g...
Guest





PostPosted: Wed Nov 12, 2008 11:59 am    Post subject: [Freeswitch-users] att_xfer+loopback Reply with quote

yes please put them on pastebin too


On Wed, Nov 12, 2008 at 10:42 AM, x y <fs_ask_sy@citromail.hu (fs_ask_sy@citromail.hu)> wrote:
Quote:
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!


_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (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 ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400
Back to top
fs_ask_sy at citromail.hu
Guest





PostPosted: Wed Nov 12, 2008 12:01 pm    Post subject: [Freeswitch-users] att_xfer+loopback Reply with quote

I've sent both of them.
Back to top
Display posts from previous:   
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> freeSWITCH Users All times are GMT - 5 Hours
Goto page 1, 2  Next
Page 1 of 2

 
Jump to:  
You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot edit your posts in this forum
You cannot delete your posts in this forum
You cannot vote in polls in this forum


Powered by phpBB © 2001, 2005 phpBB Group

VoiceMeUp - Corporate & Wholesale VoIP Services