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FreeSWITCH Cookbook - New recipes needed
by Michael Collins 29 Jun '11

29 Jun '11

Hello all! I've replaced the old recipe list with a new one. The link is still here on the wiki along with other information for prospective authors: http://wiki.freeswitch.org/wiki/Cookbook#Information_for_prospective_authors We've made a few changes to the content. Specifically, we've removed two chapters: "Call Center Services" and "Carrier Class Services". We do not feel that these can properly be done in "recipe" format. So instead we've take a few of the recipes from those chapters and put them into the "misc." section on the FSCB recipe list. We invite your input on what to replace them with. What do you think? What recipes would you like to see? Send use your ideas. Thanks, Michael

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pause and restart while recording?
by Wes 29 Jun '11

29 Jun '11

I couldn't find much on this: Is it possible in lua script to let the user press a key to pause recording, and press a key again to continue the recording, just like a traditional recorder? Thanks!

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FreeSWITCH Conf Call - Plivo Discussion - Recordings Up
by Michael Collins 29 Jun '11

29 Jun '11

Hello all! I knew that today's call would be popular so I did a rush job and got the recordings up: http://wiki.freeswitch.org/wiki/Weekly_Conference_Call#Past_Calls Just grab one of the three file types from June 29. Enjoy! -MC

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SRTP problem with leg B inbound to FreeSwitch
by Jim Miller 29 Jun '11

29 Jun '11

I've gotten as far as I can with this problem and could use some help. Here's the problem: - Two soft phones can dial each other (ext 1001 and 1002) and voice works in both directions - Wireshark shows Leg A in/out from phone 1 (10.1.10.5) is SRTP and Leg B out from freeswitch (10.1.10.2) is SRTP. However Inbound from phone 2 (10.1.10.4) to the switch is just RTP. Any ideas? I've read what I could find on the freeswitch wiki and made the obvious changes to the dialplan  <action application="export" data="sip_secure_media=true"/> and then did this to the internal.xml sofia config  <param name="inbound-late-negotiation" value="true"/> Using Freeswitch version 1.0.head (git-2486306 2011-06-23 11-57-38 -0500) Blink is being used as the soft phone v0.24.1 released in May 2011. Linux and Mac versions. I've also tried sflphone for a linux host with no change in the behavior. If I bypass the media stream and send the SRTP between the soft clients directly all works fine. I've attached various stripped down dialplans, sip configs, and a wireshark packet dump. Thank you -Jim

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Proper prompt gain/level
by Bryan Smart 29 Jun '11

29 Jun '11

As part of creating prompts for my IVRs, I've tried to match the audio gain of my prompts with the gain of the stock English prompts. During this process, I noticed that the English Callie prompts are recorded extremely low (max gain around -16DB). I have the Cepstral Callie voice, and I must set the Cepstral volume to about 50% in order to match the gain of the English prompts. In conferences and other situations where prompts are played over conversations, the level of the prompts are obviously low. I can, of course, renormalize the gain of the prompts up to -10DB or more with a sound editor. However, I wonder if there is a better way to change the level of the prompts, or if there is a good reason for the prompts to be encoded at such a low level. I haven't considered all of the implications yet, but I'm fairly sure that encoding the prompts this quietly is not the best approach, even if it is desirable for the prompts to play quietly on a call. For each 6DB reduction in gain, there is a 50% reduction in perceived volume, and one less significant bit is used for storing the audio. In a 16-bit file, a maximum gain of -16 means that only the 14 least significant bits are actually used for encoding the audio. This results in a reduction in dynamic range, but the difference isn't really noticeable as long as the data remains 16-bit. The problem comes when the audio is converted to a different bit depth. For example, most quickie routines for converting 16-bit audio to 8-bit audio will simply chop off the 8 least significant bits. Therefore, when the prompts are converted to 8-bit audio for use by most of the narrow band codecs, the prompts are only using 6 bits of audio. If the volume of the channel is increased, then the 6 bits are promoted, and the dithering errors at the bottom become louder. In the worst case, since these prompts are only encoded with 14 bits of actual data, and converting to an 8-bit channel will only leave 6 bits of actual data, boosting the gain of the prompts on an 8-bit channel to full loudness would result in the noise floor (the level of the dithering crackle) being about -18DB. That's almost as loud as the prompts themselves sound at the moment. Anyway, regardless of observations, is there a reason why the prompts must be recorded so quietly? If I'd like them louder (without increasing the gain of the entire channel), is there a way other than running them through a tool to renormalize their max levels? Bryan

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FreeSWITCH Conf Call Today: Plivo, an OSS Twilio Alternative
by Michael Collins 29 Jun '11

29 Jun '11

Hello all! Today's conference call is busy! We have Venky from the Plivo project discussing the new software. We also have quite a few new FS items that have been added in the last month so we need to get some documentation done. :) Here's the agenda: http://wiki.freeswitch.org/wiki/FS_weekly_2011_06_29 There is a link in there to a PPT for us to look at during Venky's presentation. Talk to you soon! -Michael

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Copy custom SIP headers - {sip_copy_custom_headers=true} doesn't seem to be working
by Jonas Borjesson 29 Jun '11

29 Jun '11

Hi all, As many others, I need to copy customer X-headers from leg B back over to leg A and have found a few threads talking about using the sip_copy_custom_headers=true with the dial string but this does not seem to be working. I am doing exactly what is described in the thread below but like that guy, it does not work for me either and that thread never came to any conclusion. Does anyone know the status of this feature or what I may be doing wrong? Any help is greatly appreciated! Of course, I tried on the latest and greatest FS - 1.0.head-git-4962542 2011-06-27 10-15-03 -0500 The tread talking about this very feature: http://comments.gmane.org/gmane.comp.telephony.freeswitch.user/36287 And documentation: http://wiki.freeswitch.org/wiki/Variable_sip_copy_custom_headers Thanks, /Jonas

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mod_sndfile.c - path separator for windows - system error
by ankIT WALiA 29 Jun '11

29 Jun '11

While executing a lua script, I got this error. I am running FS on window 7. 2011-06-29 17:27:52.265291 [ERR] mod_sndfile.c:195 Error Opening File [C:\ankitw alia\FreeSwitch\freeswitch-1.0.6\Debug\sounds/en/us/callie\blah.wav] [System err or : The system cannot find the file specified. ] This seems to be because of path seperator used as "/" instead of "\" for windows in mod_sndfile.c file. Thanks Ankit -- Life is like a rose its upto u feel it as its fragrance or thorns

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ClueCon Registration by July 4th Means More Chances to Win
by Anthony Minessale 28 Jun '11

28 Jun '11

Don't forget that anyone who registers for ClueCon before July 4th gets 3 extra tickets to win the prizes! Now is your chance to interact with our fast-paced world-wide development community in person.... http://www.cluecon.com visit our community conference for tomorrow's meeting or anytime at http://conference.freeswitch.org and press the register for ClueCon button to register by web phone right from your browser! -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900

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Forwarding SIP calls between 2 VoIP carriers without using SIP TRUNKING
by mosbah abdelkader 28 Jun '11

28 Jun '11

Hi, One client have 2 SIP accounts: SIP1 and SIP2 with 2 different carriers: C1 and C2. He wants to forward all calls destined to SIP1 to SIP2. C2 is using FreeSWITCH. I don't want to register SIP1 as SIP trunk in C2 because if the number of SIP accounts is big, there will be a performance problem as each account will be registered as a SIP trunk. Can anyone help me by giving an idea and a config sample for FreeSWITCH. Thank you. -- Best Regards

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