July 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Trying to confirm answer directly on Originate command
by Gustavo Dartagnan Xavier 03 Jul '09

03 Jul '09

I'm trying to build a click to call app using the FreeSWITCH webapi. Maybe I'm trying the wrong way, but, I couldn't understand why it's playing the confirmation wave only on the second leg, and not only on the first one. Is this originate command correct? http://myFreeSWITCH:8080/webapi/originate?{forked_dial=false,ignore_early_m… Is therea an easy way to do it? Thanks in advance, Gustavo

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Database for Nibble Rates?
by Edmar Cruz 03 Jul '09

03 Jul '09

Is there any options make nibble rates to MySQL database? -- View this message in context: http://www.nabble.com/Database-for-Nibble-Rates--tp24320955p24320955.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Language Handling: call for assistance
by Michael Collins 03 Jul '09

03 Jul '09

Hello all! There's been some discussion lately on how to handle multiple languages, specifically with the *say* application. We would like some input from the community on how to handle multiple languages and sound files. Anthony notes that the say application needs to build the path to the sound files by using the ${sound_prefix} and ${lang} variables. Some have asked about countries or language variants, like European Portugese vs. Brazilian Portugese. These are good questions. >From the community we need input. If you have experience with multiple languages in a telephony environment then please give us your suggestions. How would you like to see the say application handle various languages and dialects? Please give us your helpful suggestions. Thanks, Michael

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BOUNTY - extended RTP stats - RFC3611
by Jay Binks 03 Jul '09

03 Jul '09

howdy all.. ive added some extra $$$ to this existing ( seemingly closed ) bounty http://wiki.freeswitch.org/wiki/Bounty#RFC_3611_-_RTP_Control_Protocol_Exte… http://jira.freeswitch.org/browse/BOUNTY-4 it would be sweet if a) a Jira admin would re-open the bounty ticket b) other users would get on board and post some $$$ towards this cause quality monitoring in your network must be important to more than a few of us, so lets show the core devs and offer some support to get some awesome quality QOS Stats into FS. Jay

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How to remove the IP from the SIP caller id number
by Mitchel Constantin 02 Jul '09

02 Jul '09

Hello, I'm working on configuring my FreeSWITCH and would like to set the caller id number like this in dialplan/default.xml: <action application="set" data="effective_caller_id_name=John Doe"/> <action application="set" data="effective_caller_id_number= johndoe(a)weavver.com"/> I wonder if this is a problem with eyeBeam.. When the call is received the CID is like this: John Doe johndoe@weavver.com@205.134.225.20 205.134.225.20 is the EXT IP of the switch I'd like to remove the 205.134.225.20 from the Caller ID if possible. From what I understand the CID without the IP would still point to the correct server. TIA, -- Mitchel Constantin Weavver, Inc.

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seeking for a freeswitch expert
by Tzury Bar Yochay 02 Jul '09

02 Jul '09

Hi, We are looking to deploy a secured sip environment whereas all traffic, signalling and rtp encrypted etc. according to the rfc's. We are looking for an expert with deep understanding in both fields the freeSwitch and crypto world so to config and compile this for us. you may contact me at tzury.by(a)reguluslabs.com. kindly provide information about your experience with freeSwitch deployment. -- Tzury Bar Yochay Regulus Labs ltd. http://reguluslabs.com +972 52 5133399

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Freswitch closes UDP port after OPTIONS with TLS?
by Peter P GMX 02 Jul '09

02 Jul '09

Hello, I have the following problem: Every call stops after 30 seconds when TLS is enabled. SIP/RTP and SIP/SRTP works but not TLS/SRTP. The phones are behind NAT. So I expect, that every 30 seconds an Options request is sent. Wiresharking the traffic I can see * that there are ongoing UDP packets. * Then a TSLv1 packet ist sent from FS to the Phone. * This is acknowleged by the phone * Next the phone send another UDP packet to the same FS port as before * Then the Phone receives an ICMP request that the FS port is closed. Anybody has a clue about this? Best regards Peter

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SIP re-invite / bypass_media
by Phillip Jones 02 Jul '09

02 Jul '09

Hi there, I was wondering whether it is possible to have FreeSwitch go into bypass_media mode on demand? For instance, leg a bridges to leg b - leg b is invited to accept the call by pressing 1. I want to go to bypass_media (do a SIP reinvite to reroute the media) after the one is pressed. Currently I am issuing the following from my js script that prompts for the 1: session.apiExecute("uuid_media",session.uuid); Not working however. Any help to get me going would be appreciated. Thanks Phillip Jones.

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Jingle (Mod_Dingaling Dialplan Extention
by Meftah Tayeb 02 Jul '09

02 Jul '09

hello, i asked about mod_dingaling usage befor finally, i configured my XMPP acount with my FS, connected to it and is ready / Online now i need to call / recev call from / to GTalk (Jingle) cool anyone give me a sample Dialplan Extention? thanks __________ Information from ESET NOD32 Antivirus, version of virus signature database 4209 (20090702) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com

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Connecting to FS through text or SMS?
by Edmar Cruz 02 Jul '09

02 Jul '09

Hi everyone, Can you give me some proper instructions what I will do to enable texting or SMS through freeswitch using a GSM gateway? Thanks for your cooperation, Edmar -- View this message in context: http://www.nabble.com/Connecting-to-FS-through-text-or-SMS--tp24304261p2430… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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