June 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

bind_meta_app no longer works on a-leg of outbound call
by elijah 22 Jun '11

22 Jun '11

My dialplan scheme to bind #2 no longer works on the a-leg of outbound calls after updating against git yesterday. Nothing is logged when a user presses #2. However, pressing un-mapped keys, such as #3 produces: 2011-06-22 12:46:04.598820 [WARNING] switch_ivr_async.c:2938 sofia/internal/ 11(a)live001.voice.telifi.com Ignoring meta digit '3' not mapped here's the relevant dialplan items: <condition field="destination_number" expression="^9(\d{10})$"> <action application="set" data="ringback=${us-ring}"/> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="bind_meta_key=#"/> <action application="bind_meta_app" data="2 a s execute_extension::dx2 XML callsdirect"/> <action application="bridge" data="sofia/gateway/onesource/$1"/> </condition> <extension name="bind_receiver_2"> <condition field="destination_number" expression="^dx2$"> <action application="answer"/> <action application="transfer" data="-both dynamic_conference XML callsdirect"/> </condition> </extension> <extension name="dynamic_conference"> <condition field="destination_number" expression="^dynamic_conference"/> <condition field="rdnis" expression="^(dx2)$"> <action application="answer"/> <action application="bind_digit_action" data="moderator,*1,exec:execute_extension,ASK_FOR_NUMBER__${telifi_call_id} XML callsdirect"/> <action application="bind_digit_action" data="moderator,*2,exec:execute_extension,CANCEL_LAST_CALL__${telifi_call_id} XML callsdirect"/> <action application="digit_action_set_realm" data="moderator"/> <action application="conference" data="${telifi_call_id}@simple"/> <anti-action application="log" data="INFO is moderator: 'no'"/> <anti-action application="answer"/> <anti-action application="conference" data="${telifi_call_id}@simple "/> </condition> </extension>

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Need help detecting when phone call is answered in lua script (it is starting before the call is picked up)
by Wes 22 Jun '11

22 Jun '11

Hello, I'm having trouble detecting when a phone call (originated from freeswitch) is answered by the called party using a lua script. I have freeswitch connected to a corporate phone system at my office via SIP. When I call my deskphone at work (914145551212), the script waits until I answer to start playing the welcome message, and other actions. But when I call my cellphone (912625551212), the script starts before I pick up the cellphone, as can be seen in the logs I've pasted below. Even though the lua script isn't properly detecting when the cell phone has been answered, freeswitch *does* seem to know, as shown by the line: 2011-06-22 09:21:28.334618 [NOTICE] sofia.c:5594 Channel [sofia/external/912625551212] has been answered which gets logged when I answer the cell, even though the script is already running. This fact gives me hope that there will be a way to fix this script. Any suggestions are appreciated! thanks. using: FreeSWITCH Version 1.0.head (git-8decee3 2011-06-20 13-21-20 -0500) Script http://pastebin.freeswitch.org/16564 logs: http://pastebin.freeswitch.org/16563

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FS and ASR engine
by Hector Geraldino 22 Jun '11

22 Jun '11

Hi everyone, I want to check with you guys to see if anyone has experience integrating FreeSwitch with an ASR engine, and using the engine as a merely transcriber of the conversation. I've been playing for the past two weeks or so with Nuance Speech Server/recognizer and pocketsphinx. Nuance is by far a better solution, but due to the lack of freely available documentation and my short expertise in this subject, I haven't been to achieve my goal. The communication between FS and the ASR engine works great using MCRP, my concern is with the ASR engine itself. I want to allow the user to speak freely, and get a transcription of what the user said. I don't want or need to understand what the meaning of the utterances are (definitely the engine doesn't need to do that), also I don't need/want to write any complex grammar or SLM to get an interpretation of the spoken phrases, I just want the plain text of what has been said. No decisions will be taken based on what the user said, this information will just be passed to a 3rd application. I don't know if this can be achieved or not without developing grammars (not suitable for open-ended dialogs) or training statistical language models. What I do recall is using Dragon Speak in MS word for dictation, without the need of doing some trtraining or developing grammars. That's exactly what I'm pursuing: a simple plain text transcription of the spoken words. Have anyone of you deal with something like this by any chance? Thanks for your help. I apologize if this is not the right place to ask this type of questions. Thanks again, Hector

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FreeSWITCH Conference Call Today
by Michael Collins 22 Jun '11

22 Jun '11

Hey folks, Our conf call agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2011_06_22 Please add your items. It turns out our speaker for today had to reschedule at the last minute so we're going to do a community scrum. Bring your topics for conversation! Thanks, MC

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mod_rtmp client API
by Madovsky 22 Jun '11

22 Jun '11

Is there any client side API to use mod_rtmp ? Thanks

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Looking for Cost Effective IVR Solution for Concurrent Calls
by ankIT WALiA 22 Jun '11

22 Jun '11

Dear all, I am working to develop an IVR solution for an NGO in Tamil Nadu, India using Freeswitch open source telephony interface. I am looking for any cost effective solution for hardware where concurrent inbound/outbound calls can be made by the IVR system. Some research, I did but I am skeptical, need some more guidance from experts. Number of concurrent calls for initial stage inbound 5-10 but based on response from the community, we will plan for scaling up. Outbound Scheduled calls are of lesser importance right now. I checked online that PRI line provides me 30 lines for voice which can be used concurrently. But, the rent for the line is very high which cant be feasible for the NGO. Min PRI line cost is Rs 4.5 K per month. Please suggest alternative solutions which I can use for the same. 1. What about VOIP and using their local access number? 2. What about SIP based IP phone with extensions? 3. What about trunk lines, DID etc? 4. What about GSM Modems, GSMOpen? I am very new telecom technology. Please help. Thanks Ankit -- Life is like a rose its upto u feel it as its fragrance or thorns

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Microsoft Speech Server
by Jan Berger 22 Jun '11

22 Jun '11

Hi, Have anyone connected to Microsoft Speech Server from FreeSWITCH? I have briefly tested it on an Athlon II X2 215 2.7 Ghz. Windows 7 Home (not with FS). It works, but the CPU bounce quite high on a single channel - so I am guessing 5 simultaneous voice streams before the CPU's are 100%. I am not a big fan of running speech on Windows, but I wanted to try it to see if I can get it working. Jan

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Re: [Freeswitch-users] Looking for Cost Effective IVR Solution for Concurrent Calls
by hemant kumar 22 Jun '11

22 Jun '11

You should be able to get pri with min rent of 1750rs from BSNL as i had got one installed myself, after the whatever is the usage. On Wednesday, June 22, 2011, sharad <sharad(a)coraltele.com> wrote: > > > > > > > > Yes, there are following ways > - > > 1. Analog trunk line > (FXO) - taken from BSNL. Connect those lines to FXO GW / PBX & route the > calls to freeswitch. If you are expecting your incoming calls within India, this > may be cheaper. > > 2. PRI - You already > know. > > 3. Through VoIP - If > you are expecting international calls, this medium may be cheaper. But you need > to discuss this with your VoIP provider in advance that you r expecting incoming > calls also. > > Regards > Sharad > > > > ----- Original Message ----- > From: > ankIT > WALiA <ankitwalia4u(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > > Sent: Wednesday, June 22, 2011 3:18 > PM > Subject: Re: [Freeswitch-users] Looking > for Cost Effective IVR SolutionforConcurrent Calls > > I want to know If there is any other way to receive concurrent > incomings calls apart from using PRI. > > I think Reliance is giving in > 4.5K. I saw here > http://mobile.ebest.in/pri-tariff-plans/reliance-pri/3.html > > > On Wed, Jun 22, 2011 at 5:17 AM, sharad <sharad(a)coraltele.com> > wrote: > > > Hi Ankit > > I differ with you on the PRI pricing. > Now a days, in north India, BSNL, Reliance, Airtel, etc, are providing > PRI connectivity somewhere arround 10-12 K per month. More over this, they > will permit you to make the free calls of the same value. Means if you opt > for 10K / month plan, you will be able to make the out calls of wrth Rs 10K > approx. Yes, if you want only inbound, you will have to pay at least > 10K. So you can check with the local service provider in Tamilnadu > & nagotiate with the service provider for minimum rent. > > Regarding your rest queries, I > am not sure what do you want to ask..plz explore your queries. > > Regards > Sharad, Noida > 9891499202 > > > > > > > ----- Original Message ----- > From: ankIT WALiA > To: FreeSWITCH Users Help ; The Linux-Delhi > mailing list ; ilugc(a)ae.iitm.ac.in > > Sent: Wednesday, June 22, 2011 10:46 > AM > Subject: [Freeswitch-users] Looking > for Cost Effective IVR Solution forConcurrent Calls > > Dear all, > > > I am working to develop an IVR solution for an NGO in Tamil Nadu, > India using Freeswitch open source telephony interface. > > > I am looking for any cost effective solution for hardware where > concurrent inbound/outbound calls can be made by the IVR system. Some > research, I did but I am skeptical, need some more guidance from experts. > Number of concurrent calls for initial stage inbound 5-10 but based on > response from the community, we will plan for scaling up. Outbound > Scheduled calls are of lesser importance right now. > > > I checked online that PRI line provides me 30 lines for voice which > can be used concurrently. But, the rent for the line is very high which > cant be feasible for the NGO. > Min PRI line cost is Rs 4.5 K per month. > > > Please suggest alternative solutions which I can use for the > same. > > > 1. What about VOIP and using their local access number? > 2. What about SIP based IP phone with extensions? > 3. What about trunk lines, DID etc? > 4. What about GSM Modems, GSMOpen? > > > I am very new telecom technology. Please help. > > > Thanks > Ankit > > -- > Life is >

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FreeSWITCH Cookbook Recipe List
by Michael Collins 21 Jun '11

21 Jun '11

Hello all! Some have inquired recently about what's needed to complete the FS cookbook. I've uploaded the most recent list of recipes that we have identified. Some are already written but many are yet to be done. Check out this page: http://wiki.freeswitch.org/wiki/Cookbook#Information_for_prospective_authors Contact me if you have any questions about helping with any of these recipes. Thanks, Michael

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Intermittent Audio Problem - Incomming Calls
by Brian Campbell 21 Jun '11

21 Jun '11

I have run into a very strange issue with periodically losing Audio on Incomming calls I am receiving incomming calls to FreeSwitch via a DID from an ITSP. Once FreeSwitch receives the call, I am adding a header to pass the UUID (which I use later to inject some custom values into CDRs) and then bridge to an IVR application running on Microsoft Speech Server 2007 which is on a separate box, using the following dial plan <extension name="Inbound_FromSoTel"> <condition field="destination_number" expression="^1(\d+)$"> <action application="set" data="sip_h_X-FS_UUID=${uuid}" /> <action application="bridge" data="sofia/external/$1@173.14.17.213:5060;transport=tcp" /> </condition> </extension> Some calls are performing properly, but in some of the calls (about 30%) the IVR application doesn't seem to hear anything the caller is saying I have eliminated any Firewalls for testing, so I go straight from the ITSP to FreeSwitch and then to Speech Server. Thee is no NAT in play. There doesn't seem to be any rhyme or reason as to when a call is going to work, or when it isn't. I have also seen exactly the same behavior in a completely different environment with a different ITSP and a completely different IVR application. So I figure that the problem is somewhere between FreeSwitch and Speech Server, but I don't see anything wrong. Using WireShark I can see the SIP signalling creating/tearing down the call and I see RTP traffic flowing both ways between FreeSwitch and Speech Server The version of FreeSwitch that I am running is 'FreeSWITCH Version 1.0.head (git-38f06a3 2011-05-19 15-39-43 -0500)' I have pasted 2 FreeSwitch traces into pastebin. The first one is of a working call, the second one is of a call where the IVR application doesn't seem to hear anything the caller is saying Can anyone advise on what the issue might be. I am completely out of ideas. Thanks Brian Campbell

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