- FreeSWITCH-users - FreeSWITCH Mailing Lists

Unable to Bridge to FIFO offhook agent
by Woody Dickson 24 Mar '10

24 Mar '10

Hello, Freeswitch's ACD is cool! I am having a strange problem here. If I use a dialplan extension to bring a caller to a FIFO ( that already have the agent added using "fifo_member add" command), then, the agent's phone will ring and call is successfully bridged when the agent picks up the phone. The problem I have is that if the caller is bought to the FIFO after pressing a specific key in an IVR, then, when the agent picks up the phone, both caller and agent are being hung up by freeswitch. In the console, it says that it is failed to bridge the caller to the agent. Why is that the case and how can I resolve that? Best Regards, Woody

3 5

bypass_media
by Madovsky 24 Mar '10

24 Mar '10

Hi, when I use bypass_media on a dialplan extension to transit audio/video I noticed chunky audio like there is not enough buffer or something like this. Any idea ? Thanks

3 2

how to export variables from b-leg to a-leg?
by rentmycoder rentmycoder 24 Mar '10

24 Mar '10

Hi guys, Freeswitch Rocks! Does anybody know how to export variables from the b-leg to a-leg... I start a bridge from lua on an outgoing gateway and on the answer of the b-leg I would like to set a variable on the a-leg... <action application="export" data="nolocal:execute_on_answer=lua cc.lua extensions_answer ${uuid}"/> <bridge ....> from cc.lua I try: session:execute("export", "foo=123"); but no success... any hint?

3 2

leg B hangup
by Madovsky 24 Mar '10

24 Mar '10

Is there any variable to set to help leg B hangup when Leg A (originator call) hangs up or is it a a normal behaviour ? Thanks Fanck

1 0

Compile issue with openzap
by Joseph L. Casale 23 Mar '10

23 Mar '10

I just fetched freeswitch-1.0.5-latest.tar.bz2 right now and enabled openzap in modules.conf and compiled fs, but the openzap bits never made it in? I have done it before on this system (centos 5.3 x64) w/o issue, any known issues atm? Thanks! jlc

3 9

mod_xml_curl question
by Javier Aristizábal 23 Mar '10

23 Mar '10

Hi folks. :) is it possible to serve up you gateway configs but keep your sip profile stuff static? if yes!, could you give me clues, plz? Many thanks /Javier

1 0

Time limit
by Juan Carlos Gonzalez Cardona 23 Mar '10

23 Mar '10

Hi everyone, is posible use call limite for calls, to some extensiones? In Asterisk is possible creating custom context with set time limit to specific trunk, but i see that Freeswitch is more stable code, i need to know if is possible create groups for limit calls and assign extensiones to that group, if is possible, how i do that, thank´s. -- Juan Carlos Gonzalez C.

2 1

Paging - AASTRA phones
by Aloysius Lloyd 23 Mar '10

23 Mar '10

Hi All, How to build a dial plan for paging for groups of AASTRA phones. Here is my environment setup. FreeSWITCH on a Public IP address. All the AASTRA phones behind the Linksys WRT 54GL(Tomato firmware) . I see the following sample dialplan come with FreeSWITCH default.xml <extension name="rtp_multicast_page"> <condition field="destination_number" expression="^*pagegroup* $|^7243$"> <action application="answer"/> <action application="esf_page_group"/> </condition> </extension> What is the *pagegroup *variable referring to? Thanks Lloyd

3 8

Problem with bridge to multiple targets and loose race.
by Jonas Gauffin 23 Mar '10

23 Mar '10

Hello I got the following dial plan: http://pastebin.freeswitch.org/12527 Why do the registered phone loose the race after a second? The other number hasn't even started to ring yet. 336a7d1b-3815-4a10-8503-506765035dfe 2010-03-23 20:06:14.560000 [NOTICE] switch_channel.c:642 New Channel sofia/internal/u1000028 [336a7d1b-3815-4a10-8503-506765035dfe] 336a7d1b-3815-4a10-8503-506765035dfe 2010-03-23 20:06:15.300000 [NOTICE] switch_ivr_originate.c:3060 Hangup sofia/internal/u1000028 [CS_CONSUME_MEDIA] [LOSE_RACE] Log: http://pastebin.freeswitch.org/12528 //Jonas

2 3

Simulate dtmf on a channel
by Fraser Redmond 23 Mar '10

23 Mar '10

>From the CLI/event-socket I want to simulate a dtmf press on an existing channel to make it perform an action as if the caller had pressed the key themselves (specifically they're in a javascript dialplan and listening to a session.streamFile() which is waiting for a dtmf input.) I've had a play with uuid_send_dtmf but all that seems to do is play the tone on the channel, even with it's undocumented third parameter for from|to|both. The tone is heard, but the action that would have been performed if the caller had pressed doesn't happen. Is there any way to do this? I'd really like to keep the call in the javascript dialplan, as breaking them out could get messy, with the other things that are going on. Cheers, Fraser

1 0