January 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Recording ULAW files
by freeswitch-users＠digitaldan.com 21 Jan '09

21 Jan '09

Thanks, that's what I'm currently doing now. ----- Original Message ----- From: "Anthony Minessale" <anthony.minessale(a)gmail.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Wednesday, January 21, 2009 12:28:19 PM GMT -07:00 US/Canada Mountain Subject: Re: [Freeswitch-users] Recording ULAW files The default audio framework operates on raw audio. The mod_native_file triggers a special flag that tells the higher level api's for recording not to transcode the audio first. It would probably be easier for you to use the api_hangup_hook variable to trigger a sox command to wrap the files in a wav or use a batch process in cron to do so than to try to figure it out in the code. On Wed, Jan 21, 2009 at 1:05 PM, < freeswitch-users(a)digitaldan.com > wrote: Thanks for the quick reply. I'm new to this project so I'm not familiar with the inner workings just yet but at looking at mod_native_file.c it seems this is a thin wrapper around the switch's own file input and output routines? Would it be best to change this class or register a new file type, like .ul? If so, where would be a good starting point. ----- Original Message ----- From: "Anthony Minessale" < anthony.minessale(a)gmail.com > To: freeswitch-users(a)lists.freeswitch.org Sent: Wednesday, January 21, 2009 10:56:45 AM GMT -07:00 US/Canada Mountain Subject: Re: [Freeswitch-users] Recording ULAW files no, there is currently no way to do that. It would be feasible to add an option to mod_native_file to write wav headers around the raw data but it has not been attempted. On Wed, Jan 21, 2009 at 10:48 AM, < freeswitch-users(a)digitaldan.com > wrote: Hi, I'm recording files using the pcmu extension in order to save them in the g.711 ulaw format, which is what everything in my network uses. It appears that the recorded file is just raw data without a header. Is there any way to save this as a wav type with a header (keeping the ulaw format)? for example, running the unix command 'file' on the recording prints: /tmp/185065_f7bb8e0c-e641-11dd-800d-5ffe41c540dd.PCMU: data if I run it through sox which just outputs the same data (ulaw,8000,mono) but specifying the type as wav /usr/bin/sox -t .ul -r 8000 -c 1 -b -U file.pcmu -t wav -r 8000 -c 1 -b -U file.wav it produces a file that shows: /tmp/185065_f7bb8e0c-e641-11dd-800d-5ffe41c540dd.wav: RIFF (little-endian) data, WAVE audio, ITU G.711 mu-law, mono 8000 Hz if I try and save it within freeswitch using the wav extension, it trans-codes it to a pcm format RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz Is there any way to have freeswitch record the file as ulaw with the RIFF wav header? Thanks! _______________________________________________ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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 iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 _______________________________________________ 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 _______________________________________________ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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 iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 _______________________________________________ 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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Recording ULAW files
by freeswitch-users＠digitaldan.com 21 Jan '09

21 Jan '09

Hi, I'm recording files using the pcmu extension in order to save them in the g.711 ulaw format, which is what everything in my network uses. It appears that the recorded file is just raw data without a header. Is there any way to save this as a wav type with a header (keeping the ulaw format)? for example, running the unix command 'file' on the recording prints: /tmp/185065_f7bb8e0c-e641-11dd-800d-5ffe41c540dd.PCMU: data if I run it through sox which just outputs the same data (ulaw,8000,mono) but specifying the type as wav /usr/bin/sox -t .ul -r 8000 -c 1 -b -U file.pcmu -t wav -r 8000 -c 1 -b -U file.wav it produces a file that shows: /tmp/185065_f7bb8e0c-e641-11dd-800d-5ffe41c540dd.wav: RIFF (little-endian) data, WAVE audio, ITU G.711 mu-law, mono 8000 Hz if I try and save it within freeswitch using the wav extension, it trans-codes it to a pcm format RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz Is there any way to have freeswitch record the file as ulaw with the RIFF wav header? Thanks!

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Re: [Freeswitch-users] How to bridge without Answer? (Anthony Minessale)
by Kareem Hamdy 21 Jan '09

21 Jan '09

Hello everyone: I think what Anthony wants is (please excuse me if I'm wrong - but what I'm assuming is) a call to come in - let's say that its DID goes to person A. He wants to ring person A, let person A pick up, and then bridge the call. When working at an Asterisk VoIP vendor, I had a call in which a gentleman wanted just that. I think they paid for incoming calls or something. Anthony, please let us know if that's accurate. Thanks, Kareem -----Original Message----- From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of freeswitch-users-request(a)lists.freeswitch.org Sent: Wednesday, January 21, 2009 6:54 AM To: freeswitch-users(a)lists.freeswitch.org Subject: Freeswitch-users Digest, Vol 31, Issue 125 Send Freeswitch-users mailing list submissions to freeswitch-users(a)lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request(a)lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner(a)lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of Freeswitch-users digest..." Today's Topics: 1. Re: How to bridge without Answer? (Anthony Minessale) 2. Re: firing events from javascript - working example needed (Michael Collins) 3. Re: Problem with digium te220p (Krzysztof Zimnicki) 4. Re: Problem with digium te220p (Michael Collins) 5. Re: Hang up not received (Michael Collins) 6. ATA-answering machine question/recommendation (jonathan augenstine) ---------------------------------------------------------------------- Message: 1 Date: Wed, 21 Jan 2009 07:56:51 -0600 From: Anthony Minessale <anthony.minessale(a)gmail.com> Subject: Re: [Freeswitch-users] How to bridge without Answer? To: freeswitch-users(a)lists.freeswitch.org Message-ID: <191c3a030901210556n2d443179n17d8bbb9ed24b8ab(a)mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" You can't. Why would you need that? Are you trying to forward inbound calls from the pstn to an ivr without answering them? That could get you in trouble FYI. On Wed, Jan 21, 2009 at 7:40 AM, shehzad p <pmhshz(a)gmail.com> wrote: > > Hi all, > > When I dial a number from Originator Gateway, It will route to Freeswitch > Server and then FS will bridge the call to Terminator Gateway as below. > Terminator Answer the call (and runs playback, and look for DTMF). > > |Originator Gateway|---------------> |FreeSwitch |------------------> > |Terminator Gateway| > > I used bridge application to route call to Terminator. > But my requirement is that when Terminator answer the call (Respnd with > 200OK) , Freeswitch should NOT Answer call for A leg (Originater Gateway). > > How can be this done? > > Thanks in advance. > msp. > -- > View this message in context: > http://www.nabble.com/How-to-bridge-without-Answer--tp21583334p21583334.html > Sent from the Freeswitch-users mailing list archive at Nabble.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 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale(a)hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale(a)gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888(a)conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888(a)conference.freeswitch.org> pstn:213-799-1400

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Hang up not received
by Tomás 21 Jan '09

21 Jan '09

Hi all, I'm configuring my home PBX using FreeSwitch. I'm using a X101P card configured as FXO (conected to analog PSTN line) and I have several IP phones and softphones conected to FreeSwitch. I can call from an IP phone to other IP phone (the same with the softphones) and also from an IP phone (or softphone) to an external number thought PSTN. When I call from an external analog phone to FreeSwitch, I bridge the call to all internal IP phones and softphones and they ring, but the problem is that when I hang up the call in the external phone, all internal phones (IP phones and softphones) keeps ringing... I'm pretty sure the problem is that FreeSwitch don't receive the hang up, because I cann't see anything on the log. I've also created my own tones.conf for my country (Spain) but I'm not sure if it's ok (but I have the same problem with hang up) I've googled the list, and I've found several people with a similar problem but no solution... That's my pastebin with the most importants printouts and config files: http://pastebin.freeswitch.org/6822 Thank you very much in advance.

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Problem with digium te220p
by Krzysztof Zimnicki 21 Jan '09

21 Jan '09

>Can you join irc later today? I will be on as mercutioviz. I would >like to discuss this more. > >-MC >Sent from my iPhone Sorry, i can't join to irc. Can you put your questions here? I'll try to answer. Our CallCenter have strange situation, because now is working on Asterisk and we can only put this card in other machine after 22 pm. Thanks.

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FreeSWITCH Windows Service does not Start
by Ron Avriel 21 Jan '09

21 Jan '09

Hi, If I try starting FreeSWITCH Windows service it immediately fails with a messagebox: "The FreeSWITCH service on local computer started and then stopped. Some services stop automatically...etc." I noticed that the service is installed to log on as "NT AUTHORITY\NetworkService". If I change this to "Local System account" then FreeSWITCH starts and runs OK. This failure occurred on multiple Windows XP servers. Why does it fail and why does it not use the local system account like almost all services? Thanks, Ron

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New wiki page needs your attention!
by Michael Collins 21 Jan '09

21 Jan '09

Hello FreeSWITCHers! I'm putting the finishing touches on a wiki page that we hope will make it easier for users to request help and for the dev team and power users to digest and process those requests. What I need first and foremost is for everyone to please read this page: http://wiki.freeswitch.org/wiki/Reporting_Bugs Please give me feedback. Put yourself in the shoes of a relative newbie. Is the information easy to follow? On the flip side, if you wanted to help someone, ask yourself, if they follow the steps on this page will that suffice? Are there places that need improvement? Can you think of anything else that can be added? NOTE: I'm still working on the TDM/OpenZAP section as well as the sections on scripting, event socket, elements of a jira ticket, etc. If you have suggestions for content on those pages please email me off list or hop and and fill in some of the blanks. The core development team really appreciates all of your help. Now that FS is growing like mad we are at the point where it is imperative that we have reliable documentation for new ones so that the developers and other experts can focus on advancing the project even further. Let's all lend a hand by improving the documentation. Thanks again! -MC (mercutioviz)

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Bridge destination returned from a python script?
by Brian Deacon 21 Jan '09

21 Jan '09

Greetings my soon-to-be-BFF's :) After the last 6 weeks or so thrashing about in the land of Asterisk, I must say I'm quite impressed with what I'm learning about FreeSwitch. So the scenario I'm trying to deal with is bridging a call between two not-internal endpoints where the numbers to connect are determined by database-fed business logic. The recipe here is very close to what I need: http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#exec_in_answer_con… Except that I'm a language snob and would greatly prefer python to javascript. :) So my minor question is can this line: <action application="set" data="group_confirm_file=javascript test.js"/> Become this line: <action application="set" data="group_confirm_file=python foo.bar.baz"/> But javascript isn't a deal-breaker. I have two bridging scenarios that were too tricky for me in asterisk land. (And my apologies, I'm new to telephony, so I don't have the lingo down). Our FS server would be reachable by multiple 800 numbers. Based on the number that was dialed to reach our FS server, we would look up an external phone number + extension to be part 1 of a bridge (which we would use the group_confirm_file script to make sure we had gotten past the IVR). So for that scenario, how would I accomplish something like: <action application="bridge" data=" ${someVariableSetByScript}"/> Can my foo.bar.baz script make a session.setVariable call that I could drop into data=""? My other scenario is similar, but we need to externally signal the FS server to place a call to phone + extension, confirm it was connected, and then bridge it to a DID. Is XmlRpc the best (only?) way to poke FS into doing something? Thanks muchly! Brian

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Caller ID presentation
by Paul D. 21 Jan '09

21 Jan '09

Hi, I am fairly new to FS. Is there a way to switch off/on caller ID presentation dynamically, similar to "callingpres" in *? Also, probably a stupid question, but how do I search this mailing list archive? Thx.

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SBC functions in FS
by Jon Bruel 20 Jan '09

20 Jan '09

I'm looking into the first design phase of a setup like this: PSTN <-> FS <-> Various SIP PBXs or nothing <-> SBC <-> Internet <-> Customer Where I'm considering using the FS as the SBC and as an alternative to OpenSIPS (OpenSER), where the FS has the advantage of a design having only one type of switch, e.g. less education to the team supporting the system. If this should work, the FS should be able to route SIP messages such as NOTIFY, SUBSCRIBE etc. Is that possible, and what are the measures to implement such a routing function? /Jon

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