September 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

screen pop with freeswitch？
by Saugort Dario Garcia Tovar 09 Sep '11

09 Sep '11

Hi, I am playing with freeswitch scripting with lua. I want call to number, ex 1000, a script lua will process the call ( as an IVR), then transfer the call to an extension (or transfer the call to a queue, to get a free agent). Questions: How I add a pair data (key/value. ex ID/1234) to the call in the lua script? How can check the data attached is avaliable to the end extension? What you use to make screen pop call data when the call ring in the extension? Thanks

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What is Event MEDIA_BUG_STOP
by Glen Ganderton 09 Sep '11

09 Sep '11

Hey Guys, I am trying to get event information from the event DETECTED_SPEECH however when I telnet into the event console and type "event plain ALL" I see a lot of events but no DETECTED_SPEECH event. My speech is detected though, this is a sample of the put from the fs_cli: --------------------------------------------------------------------------------------------------- 2011-09-08 14:27:50.512150 [DEBUG] mod_pocketsphinx.c:383 Recognized: YES, Confidence: 100 EXECUTE sofia/sipinterface_1/2001(a)10.3.5.77 detect_speech(stop) 2011-09-08 14:27:57.578391 [INFO] mod_pocketsphinx.c:221 Port Closed. However in the event system there is no DETECTED_SPEECH and I see another event called MEDIA_BUG_STOP and MEDIA_BUG_STOP these look they they have something To do with speech but this isn't what im after doesn't give me all the information. Can anybody explain what my issue may be? Thanks.

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FreeSwitch Windows ISO Hard Coded Install Path...
by Gerry Hull 08 Sep '11

08 Sep '11

Is there a reason why the ISO has a hard coded install path? Is it possible to get this changed, as it used to be, so that you can select the drive/directory where the install should be? Regards, Gerry

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FreeSwitch failover/HA
by Mateusz Bartczak 08 Sep '11

08 Sep '11

Hi I would like to implement following scenario: 1. Central SIP proxy with 1 IP address that will redirect all incoming and outgoing SIP traffic to two SIP application servers (FreeSwitch based). Proxy will know online/offline status of each box and route calls only to active one. I need it as a central point because IP address is authorized with my providers. Also providers route incoming calls to that IP address. My providers can't automatically reroute traffic to other server, it can be done manually but it's not fast to do. 2. Mini cluster of two FreeSwitch boxes, box 2 will be exact copy of box 1. If box 1 fails, central server should automatically route all traffic to box 2. Do you have any suggestions how to implement this scenario? I think that it should be easy to do, but have no idea where to start. Best Regards, Mateusz

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g729 Pass-through
by Joe Flemmings 08 Sep '11

08 Sep '11

Is there a way of having just calls using g711 do media passthrough and NOT all calls Because the following parameters in internal.xml apply to all calls <param name="inbound-bypass-media" value="true"/> <param name="inbound-proxy-media" value="true"/> Thanks Joe

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small problem with the context of an FXO
by Julio Saldivar 08 Sep '11

08 Sep '11

i have a problem with the context of an FXO, i editing the freetdm.conf.xml with the following configuration: <analog_spans> <span name="FXO"/> <param name="dialplan" value="XML"/> <param name="context" value="fxo"/> <param name="polarity-delay" value="600" /> <param name="answer-polarity-reverse" value="true" /> <param name="hangup-polarity-reverse" value="true" /> </analog_spans> after i rebooting freeswitch and i running teh command "ftdm list": freeswitch@internal> ftdm list +OK span: 1 (FXO) type: analog physical_status: ok signaling_status: UP chan_count: 1 dialplan: XML context: default dial_regex: fail_dial_regex: hold_music: analog_options: none and does not change the context -- Si alguna vez mi voz deja de escucharse piensen que el bosque hablará por mí con su lenguaje de raíces.

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how about config astpp for freeswitch？
by tangdu 08 Sep '11

08 Sep '11

hello， i installed astpp-2315 for freeswitch。now i can visit "http://xxx.xxx.xxx.xxx/astpp-admin/astpp-admin.cgi " and login useing default passwd。 Promt “Successful Login! Realtime Database Unavailable! ” it's because of i couldn't found the file look like realtime.sql ，haven't creat Realtime Database 。 i found word of menu still is asterisk ，not freeswitch 。when i click any menu，system come back login interface。 why？who can help me？ thanks -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/how-about-config-astpp-for-fr… Sent from the freeswitch-users mailing list archive at Nabble.com.

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FreeSWITCH, Nuance, ASR: IVR Integration
by Glen Ganderton 08 Sep '11

08 Sep '11

Hi Guys, What i am trying to do is configure an IVR using freeswitch and Nuance 5.1.5 to perform speech recognition. I have installed the latest release of freeswitch on my CentOS system, I have also enabled and configured the unimrcp module. I could find lots of information to configure the unimrcp module but there is no documentation on how to create IVR menu's using this module with Nuance. I have used the demo pizza IVR with the pocketsphinx module and I keep getting refered to this but it's no help as I want to use Nuance for ASR. Any help on how to configure this would be great, specifically: * How to I call the IVR script from the dialplan * What language/s can I code the IVR in and can you please provide a basic sample of how this is done with Nuance. Thank you in advance --Glen

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STUN usage with Freeswitch
by Claudio Álvarez 08 Sep '11

08 Sep '11

Hi all, I currently operate a publicly accessible stun server, which one of my FreeSWITCH instances operating on Amazon EC2 queries for proper SIP and RTP configuration. When I query the stun server from FS-EC2, I get the following: freeswitch@internal> stun 74.xx.xx.xx 184.xx.xx.xx:35087 Can anyone clarify what does the 35087 number port stand for? Thanks. Regards, -- Claudio Alvarez claudio(a)clickfono.com

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One-way audio issue with attended transfer
by John Platts 07 Sep '11

07 Sep '11

We have been experiencing a one-way audio issue with attended transfer. The problem is that after an call is transferred through an attended transfer, the caller cannot be heard, but the caller can hear the voice of the person that the call is transferred to. Steps to reproduce the problem: 1. Dial into FreeSWITCH from an external phone. 2. Answer the call. One-way audio is not experienced here. 3. Press the Transfer button on the IP phone to initiate an attended transfer. 4. Dial the extension that the call is to be transferred to. 5. Allow the extension that the call is to be transferred to pick up the call. One-way audio is not experienced here. 6. Press the Transfer button to complete the transfer. The caller does hear audio from the person that the call is transferred to, but the person that the call is transferred to cannot hear the caller. The one way audio issue is occurring here. The attended transfer was performed from an extension on one SPA303 phone to an extension on another SPA303 phone. FreeSWITCH actually detects that the transfer is an attended transfer. Here are attachments that I have uploaded to pastebin: conf/dialplan/features.xml file: http://pastebin.freeswitch.org/17291 FreeSWITCH log, including the attended transfer: http://pastebin.freeswitch.org/17286 How do we get this one-way audio issue fixed?

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