- FreeSWITCH-users - FreeSWITCH Mailing Lists

DTMF Question/Issue
by Ira Tessler 08 May '13

08 May '13

I am having customer complaints that when they dial an outside conference bridge systems, (e.g. Go To Meeting, etc) or outside IVRs, they are having trouble with the remote system recognizing DTMF tones correctly (e.g. when entering a PIN for a conference bridge). I put in a way in our system to dial an outside number and bypass media for just that call. This seems to resolve the issue, but it is annoying for our customers. My question is how can I troubleshoot this issue? (I have looked at pcap's and from what I can see FS is setting the DTMF correctly) What DTMF settings in FS should I look at? Here is why I currently am using on my external sip profile: <param name="rfc2833-pt" value="101"/> <param name="dtmf-type" value="rfc2833"/> <param name="liberal-dtmf" value="true"/> <param name="rtp-digit-delay" value="60"/> <param name="manual-rtp-bugs" value="IGNORE_DTMF_DURATION"/> <param name="rtp-rewrite-timestamps" value="false"/> Thanks, Ira Tessler Lead Software Engineer ConnectMe (732) 490-9007 x2 ira(a)connectmevoice.com

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Re: [Freeswitch-users] Windows binary MSI installer
by Moishe Grunstein 08 May '13

08 May '13

The one I noticed I am missing was voicemail/vm-enter_new_pin.wav, however after comparing to the recording download I saw the download has 91 files in that folder and the MSI only has 81 files. I can see based on the callie folder size that there are many more missing, not sure if those are needed. Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support(a)nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS -----Original Message----- From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Jeff Lenk Sent: Wednesday, May 08, 2013 9:50 AM To: freeswitch-users(a)lists.freeswitch.org Subject: spam>spam>Re: [Freeswitch-users] Windows binary MSI installer What is missing? Is it a version problem? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Windows-binary-MSI-installer-… Sent from the freeswitch-users mailing list archive at Nabble.com. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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Windows binary MSI installer
by Moishe Grunstein 08 May '13

08 May '13

The weekly generated MSI files are great and up to date, and include most 8khz sounds. Is there any particular reason not all 8khz audio files are included? Is it do to the file size? Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support(a)nysolutions.com<mailto:support@nysolutions.com> Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [cid:image001.jpg@01C72F94.9EE45D60]<http://www.nysolutions.com/> Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS

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curl issue with dynamic variables
by rajat 08 May '13

08 May '13

Hi I am new to freeswitch and i am trying to integrate one php application with freeswitch. Purpose is freeswitch will send the user dialed (dtmf numbers) as input to this php application and in return it will send the response back. Dialplan which I have written for this is <extension name="inte"> <condition field="destination_number" expression="^9199$"> <action application="read" data="0 9 $${base_dir}/sounds/en/us/callie/conference/8000/input-string.wav res 60000 #"/> <action application="sleep" data="2000"/> <action application="export" data="new=${res}"/> <action application="sleep" data="2000"/> <action application="say" data="en name_spelled iterated ${new}"/> <action application="curl" data="http://192.168.10.27/index.php post id=${new}" inline="true"/> <action application="sleep" data="2000"/> <action application="speak" data="flite|slt|${curl_response_data}"/> <action application="info"/> <action application="speak" data="flite|slt|Thanks for your query"/> <action application="lua" data="helloworld.lua ${curl_response_data}"/> </condition> </extension> Now http://192.168.10.27/index.php this is a url on which i am sending my requests. Problem here is curl is not taking variable in the post. If instead of ${new}, i put some random static value like 256, then it is working perfectly fine but I want my users to input the values. Kindly suugest some way via which I can post variables in curl url -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/curl-issue-with-dynamic-varia… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Generate dynamic dialplan based on variables
by Grant Bagdasarian 08 May '13

08 May '13

Hello, When using the Originate command, is it possible to generate a dynamic sequence of actions to take place when the user answers the call? For example: Originate command is called with some arguments in this specific order; playback, transfer, playback. FreeSwitch should first play something, then transfer to some destination, and after returning from the transfer, play another message. The sequence of these actions is defined by the external application which is calling the Originate command. If this is possible, what is the best way to accomplish this? I have no experience with FreeSwitch, just doing some research. Regards, Grant

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To the New People...
by Ken Rice 08 May '13

08 May '13

Hey Guys, If you are new to the list, keep in mind that everyone joins the list moderated, and posting to the list the same email more then one time will just annoy the list moderators as they handle the moderated posts. Why do we moderate new users? Simple answer spam... Do you guys really want us to help the spammers? -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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Codec Negotiation Help
by Spencer Thomason 08 May '13

08 May '13

Hello all, I'm having a bit of trouble getting codec negotiation working the way I'd like to. I'm trying to avoid transcoding unless its absolutely necessary for the call to be setup. I use two profiles with late negotiation enabled, an internal facing registered user endpoints and an external facing the PSTN via various ITSPs. For calls to users I've been setting codec_string=${ep_codec_string} in my dial strings and normal bridges work fine but I'm now reworking the hunt groups using enterprise originate and that doesn't seem to work correctly (the callee's preferences are always used). So I'd like to do the the "right" way now :-) The basic scenario is this: On the internal profile I have: CODECS IN G722,PCMU,G729,GSM CODECS OUT G722,PCMU,G729,GSM On the external profile I have: CODECS IN G722,PCMU,G729 CODECS OUT G722,PCMU,G729 What I'm trying to accomplish is this: Call from PSTN, use caller's codecs as priority for the bridge, i.e. PSTN GW --- PCMU, G729 --> FreeSWITCH --- PCMU,G729,G722, GSM ---> Local Endpoint PSTN GW --- G729,PCMU --> FreeSWITCH --- G729,PCMU,G722, GSM ---> Local Endpoint PSTN GW --- G722,PCMU --> FreeSWITCH --- G722,PCMU,G729, GSM ---> Local Endpoint Call from local endpoint to PSTN, use callee's codecs as priority for the bridge, i.e. Local Endpoint Offers G722, PCMU, G729 GW Responds PCMU,G729 Local Endpoint --- G722,PCMU,G729 ---> FreeSWITCH --- G722,PCMU, G729 --> PSTN Desired negotiation: Both legs use PCMU Local Endpoint Offers G722, PCMU, G729 GW Responds G729,PCMU Local Endpoint --- G722,PCMU,G729 ---> FreeSWITCH --- G722,PCMU,G729 --> PSTN Desired negotiation: Both legs use G729 Local Endpoint Offers GSM GW Responds PCMU, G729 Local Endpoint --- GSM ---> FreeSWITCH --- G722,PCMU,G729 --> PSTN Desired negotiation: Leg A uses GSM, Leg B uses PCMU, FreeSWITCH transcodes. Any help is greatly appreciated! Thanks, Spencer

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How to modify the way freeswitch handle SIP OPTIONS message
by Iwan Budi Kusnanto 08 May '13

08 May '13

Hi, I can see from console log that freeswitch always reply with SIP 200 when receive SIP OPTIONS message. Can i modify this behaviour? How to modify it? Thanks -- Iwan Budi Kusnanto

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Do Freeswitch conference support silent suppression?
by nordin mohamed 07 May '13

07 May '13

Currently I'm trying to create conference server using Freeswitch and hook up with the ROIP device (Vocality). The ROIP device is act as the voip extension for the Freeswitch, In order for the radio link to work, the conference must support the VAD/CNG so that the silence is not transmitted by the ROIP device to its radios.My problem is the ROIP will always transmit something on the radio. Am I correct to conclude that the Freeswitch not supporting silent suppression while in the conference mode.? thank you -nordin mohamed

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Request for review - FS-5388
by Cal Leeming [Simplicity Media Ltd] 07 May '13

07 May '13

Hello! Could a core dev spend a minute or two reviewing the following ticket please; http://jira.freeswitch.org/browse/FS-5388 In return, I have triaged the following ticket; http://jira.freeswitch.org/browse/FS-5168 I spent 10 minutes looking for other tickets I could help with, but all of them were beyond my skill level :/ Cal

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