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Re: [Freeswitch-users] Adding an info digit to sip from
by Josh Forman 25 Feb '09

25 Feb '09

The problem here is that what you are showing me produces: From: "1004" <sip:1004@99.185.85.3>;tag=1SparjgraS69m To: <sip:1235@conference.freeswitch.org;this=rocks> when what I need to output would look like this: From: "1004" <sip:1004@99.185.85.3;this=rocks>;tag=1SparjgraS69m To: <sip:1235@conference.freeswitch.org> with the "this=rocks" in the FROM field, not the TO field. I know that you can change parts of the from field by setting effective_caller_id_name and effective_caller_id_number, but I don't know how I would add that bit of data to the end of the SIP URI inside the < > Is there a variable that I could set or perhaps some method similar to overwriting the To header shown at http://wiki.freeswitch.org/wiki/Sofia#Modifying_the_To :_header that can be used to accomplish this? Thanks Josh On Feb 25, 2009, at 2:35 PM, freeswitch-users-request(a)lists.freeswitch.org wrote: >> Message: 3 >> Date: Wed, 25 Feb 2009 09:29:26 -0600 >> From: Brian West <brian(a)freeswitch.org> >> Subject: Re: [Freeswitch-users] Adding an info digit to sip from >> header >> To: freeswitch-users(a)lists.freeswitch.org >> Message-ID: <0E1E9F62-64C5-4B70-9C67-C7B5728DB111(a)freeswitch.org> >> Content-Type: text/plain; charset="us-ascii" >> >> You can do something like this "sofia/blah/somenumber@someip: >> 5060;this=rocks" >> >> /b >> >> On Feb 25, 2009, at 9:22 AM, Josh Forman wrote: >> >>> I'm trying to edit the sip headers to make the from field look like >>> this: >>> >>> From: <sip:SOMENUMBER@SOMEIP:5060;isup-oli=27>;tag=gK0a00d6ea. >>> >>> I know that to read that data on an incoming sip message it is in $ >>> {sip_from_params}, but how can I add the ;isup-oli=27 part on an >>> outgoing message? >>> >>> Thanks >>> >>> Josh > > > > ------------------------------ > > Message: 3 > Date: Wed, 25 Feb 2009 10:53:39 -0600 > From: Brian West <brian(a)freeswitch.org> > Subject: Re: [Freeswitch-users] Adding an info digit to sip from > header > To: freeswitch-users(a)lists.freeswitch.org > Message-ID: <D58A3A53-8535-4EAF-BFE5-B86F621C1393(a)freeswitch.org> > Content-Type: text/plain; charset=US-ASCII; format=flowed > > It will actually add it to both places > > INVITE sip:1235@conference.freeswitch.org;this=rocks SIP/2.0 > Via: SIP/2.0/UDP 99.185.85.3;rport;branch=z9hG4bK0Kaa1322U42eK > Max-Forwards: 69 > From: "1004" <sip:1004@99.185.85.3>;tag=1SparjgraS69m > To: <sip:1235@conference.freeswitch.org;this=rocks> > > I verified it does indeed add it in both places. > > /b > > > > On Feb 25, 2009, at 10:34 AM, Josh Forman wrote: > >> Which variable would I need to set via the dialplan to do this >> though? Your example looks like it would be the dialstring for the >> bridge application but if that works it would probably be added to >> the >> To header instead of the From, right? I can't be sure since nothing >> I've tried has had any affect. >> Between looking at the wiki and random experimenting I haven't found >> anything that works thus far. >

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Re: [Freeswitch-users] problem speaking with cepstral voice in xml ivr menu
by Alexander de Greiff 25 Feb '09

25 Feb '09

michael, i googled for "asterisk alternative" and voila... the trigger was that every now and then i renew servers in the infrastructure and the one with asterisk was overdue. i wasn't really unhappy with asterisk, but these things bothered me (maybe i am not up to date): - dialplan gets messy - no conferences without hardware (rented remote server!) - ivr with cepstral voices: sometimes get hickups so far i like the fs approach very much. stable sip channels, no hickups with voices. kind regards alex ----- Ursprüngliche Mail ----- Von: "Michael Collins" <msc(a)freeswitch.org> An: freeswitch-users(a)lists.freeswitch.org Gesendet: Mittwoch, 25. Februar 2009 19:31:24 GMT +01:00 Amsterdam/Berlin/Bern/Rom/Stockholm/Wien Betreff: Re: [Freeswitch-users] problem speaking with cepstral voice in xml ivr menu > so far i am a happy camper with freeswitch. this is a different snack bracket than asterisk... If you don't mind telling us, where did you hear about FS and what made you decide to try it? Are you unhappy with Asterisk or are you simply looking for something a bit different? Just curious. Thanks, MC _______________________________________________ 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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problem speaking with cepstral voice in xml ivr menu
by Alexander de Greiff 25 Feb '09

25 Feb '09

hi all, here is my second problem trying to migrate from * to fs: i can speak with cepstral voices from my dialplan, but when i implement an ivr menu with cepstral voices like this: <menu name="demo3" greet-long="say:text to speak" greet-short="say:main menu" invalid-sound="say:invalid entry" exit-sound="say:goodbye" timeout ="10000" max-failures="3" tts-engine="cepstral" tts-voice="allison" phrase_lang="en"> <entry action="menu-exit" digits="*"/> <entry action="menu-back" digits="5"/> </menu> i get the following errors: [ERR] mod_native_file.c:68 native_file_file_open() Error opening /usr/local/freeswitch/sounds/en/us/callie/say:text to speak.GSM can you point me in the right direction? thanks alex --- freeswitch 1.0.3 build 12166

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Question about BLF...
by Klaus Hochlehnert 25 Feb '09

25 Feb '09

Hi, I'm just playing around with FreeSWITCH and I have 2 questions about BLF (with SNOM phones): - When I played around with the sample dial plan I found out that BLF works better than Asterisk, but not 100% right: > When phone 1000 gets a call the BLF lamp on phone 1001 blinks and after phone 1000 takes the call the lamp on phone 1001 is on > But when phone 1000 gets a second call, takes it and hangs up the lamp on phone 1001 turns off even if the first call is still active > Is that a problem or did I do something wrong??? - Second question is how can I set up BLF if I want to have my dial plan completely in a perl script (no XML besides calling the perl script)? Thanks, Klaus

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Re: [Freeswitch-users] howto originate fs call from webapp (python)
by Alexander de Greiff 25 Feb '09

25 Feb '09

hi, oops, i must have been very tired when i wrote my first mail to the list... thanks for your replies. {ignore_early_media=true} really worked for me. i try very hard to "unlearn" asterisk. with asterisk i did not do much more with the python script, but i would like the pthon script to interact more with freeswitch like: - call number1 - say a welcome message with cepstral voice - call number2 - bridge other scenario: enter telephone number in webapp python script have fs to call number say "please enter the pin code from the website" validate dtmf code pass back to webapp: correct or not correct unfortunately just from reading the wiki i don't know how to do it in my python script. can you share your experience? thanks alex

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Re: [Freeswitch-users] Adding an info digit to sip from header
by Josh Forman 25 Feb '09

25 Feb '09

Which variable would I need to set via the dialplan to do this though? Your example looks like it would be the dialstring for the bridge application but if that works it would probably be added to the To header instead of the From, right? I can't be sure since nothing I've tried has had any affect. Between looking at the wiki and random experimenting I haven't found anything that works thus far. On Feb 25, 2009, at 11:01 AM, freeswitch-users-request(a)lists.freeswitch.org wrote: > Message: 3 > Date: Wed, 25 Feb 2009 09:29:26 -0600 > From: Brian West <brian(a)freeswitch.org> > Subject: Re: [Freeswitch-users] Adding an info digit to sip from > header > To: freeswitch-users(a)lists.freeswitch.org > Message-ID: <0E1E9F62-64C5-4B70-9C67-C7B5728DB111(a)freeswitch.org> > Content-Type: text/plain; charset="us-ascii" > > You can do something like this "sofia/blah/somenumber@someip: > 5060;this=rocks" > > /b > > On Feb 25, 2009, at 9:22 AM, Josh Forman wrote: > >> I'm trying to edit the sip headers to make the from field look like >> this: >> >> From: <sip:SOMENUMBER@SOMEIP:5060;isup-oli=27>;tag=gK0a00d6ea. >> >> I know that to read that data on an incoming sip message it is in $ >> {sip_from_params}, but how can I add the ;isup-oli=27 part on an >> outgoing message? >> >> Thanks >> >> Josh

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Re: [Freeswitch-users] Recording and outbound rtp
by freeswitch-users＠digitaldan.com 25 Feb '09

25 Feb '09

no, I'm matching the incoming sip call via the destination number in my public context and executing the javascript appliaction. This app directly answers the call and records it until the user hangs up. D- ----- Original Message ----- From: "Anthony Minessale" <anthony.minessale(a)gmail.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Tuesday, February 24, 2009 12:05:33 PM GMT -07:00 US/Canada Mountain Subject: Re: [Freeswitch-users] Recording and outbound rtp is it during a bridged call? On Tue, Feb 24, 2009 at 11:49 AM, Dan < freeswitch-users(a)digitaldan.com > wrote: Hi, I have a small javascript application that accepts a call, retrieves some dtmf digits and then records the call to an icecast server. This works great. The problem I'm having is that when the call is being recorded freeswitch is no longer sending rtp packets back to the originating caller, in my case a Cisco 5300 with a bunch of T1 voice circuits in it. This makes sense, since no voice data back is being generated. Unfortunately my Cisco gear has rtp inactivity timers set up to hang up a call after 3 minutes of no incoming rtp packets, this is a global setting that cannot be configured for a single dial peer. Does anyone have a suggestion to generate rtp packets every once in a while? I tried setting comfort noise which did not seem to send anything. I could try playing a empty/short wav file every minute or so but the javascript call session.record is blocking, would a traditional javascript timer and callback to play a wav file be my best bet or is there a better approach? I'm using FreeSWITCH Version 1.0.trunk (12108M) on debian etch. Thanks! Dan- _______________________________________________ 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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Adding an info digit to sip from header
by Josh Forman 25 Feb '09

25 Feb '09

I'm trying to edit the sip headers to make the from field look like this: From: <sip:SOMENUMBER@SOMEIP:5060;isup-oli=27>;tag=gK0a00d6ea. I know that to read that data on an incoming sip message it is in $ {sip_from_params}, but how can I add the ;isup-oli=27 part on an outgoing message? Thanks Josh

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Skypiax, same skype user, multiple channels
by Eric Chamberlain 25 Feb '09

25 Feb '09

I was reading through the Skypiax documentation and saw the comment that it's not possible to run multiple skype clients on the same linux machine, all using the same skype user account. It's possible to run multiple skype clients with the same skype user account, as long as the skype clients are not accessing the same Skype dbpath. We use runuser to run multiple skype clients. All the clients use the same skype user, but each instance uses a different home directory, each with its own .Skype folder. In such a configuration, will Skypiax support multiple channels using the same skype username? -- Eric Chamberlain, Founder RF.com - http://RF.com/

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Skypiax (Skype Network endpoint) for FreeSWITCH
by Giovanni Maruzzelli 25 Feb '09

25 Feb '09

Hello FreeSWITCHers, mod_skypiax is available for testing, feature requests, bug hunting. I would like to ask the help of you all to make Skypiax robust and feature full on FreeSWITCH, and particularly of Massimo Cetra (CtRiX on IRC), that has developed mod_airpe (another Skype endpoint). I've written a first documentation on Skypiax installation and usage at: http://wiki.freeswitch.org/wiki/Skypiax and there is a Jira module at: http://jira.freeswitch.org/browse/MODSKYPIAX So, please, test the software, edit the wiki page both for style and content, file bug reports and feature requests. FreeSWITCH is now the platform of first development for me, so the FreeSWITCH part of Skypiax is more tested (if any) and the code is more readable compared to the Asterisk part where lot of legacy from my other projects clutter the code. But Skypiax strive to be available as a Skype compatible endpoint for all the opensource telephony community, and in the near time the Asterisk part will be cleaned much more, and documented. As you will see, the code is made by skypiax_protocol.c (the interaction with Skype client), mod_skypiax.c (the interaction with FreeSWITCH), chan_skypiax.c (the interaction with Asterisk). Please consider me available for all infos, clarifications, discussions, etc. I would like to thanks all the peoples that helped me via mail and IRC (so bad to have different timezones, isn't?), the *very early adopters*, the testers, the patchers, and you all. Particularly Anthony Minessale, Michael Jerris, Brian West, Michael Collins, Ken Rice, Seven Du, Clif Cox, Hristo Trendev, Rehan Allah Wala, Jason Garland and Antonio Gallo. >From the wiki page (http://wiki.freeswitch.org/wiki/Skypiax) WHAT IS SKYPIAX This software (Skypiax) uses the Skype API but is not endorsed, certified or otherwise approved in any way by Skype. Skypiax is an endpoint (channel driver) that uses the Skype client as an interface to the Skype network, and allows incoming and outgoing Skype calls to/from FreeSWITCH (that can be bridged, originated, answered, etc. as in all other endpoints, e.g. sofia/SIP). Skypiax works in FreeSWITCH (FS) on both Linux and Windows, at both 8khz and 16khz (Skype client has 16khz audio I/O). Skypiax works on Asterisk too, at 8khz, on Linux and Windows (through CygWin). Think of Skypiax as similar to OpenZAP for analog lines. For each channel you need an interface (a Skype client). So, for example, two concurrent calls would need two channels, and therefor two Skype clients running on your FreeSWITCH server. If your Skype client(s) have Skype credits, then Skypiax works for SkypeOut calls as well. Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Thu, Jan 15, 2009 at 6:39 PM, Giovanni Maruzzelli <gmaruzz(a)celliax.org> wrote: > Hi FreeSWITCH developers > > I would like to propose to the community my plans, so we can discuss > and coordinate efforts. > > I developed a couple of channel drivers for Asterisk in the past > (works on both Linux and Windows), and I would like to port them to FS > and further enhance them. > > The two endpoints are: > - Skypiax, Skype compatible, makes and receives calls to/from Skype > network and Skypeout service, using the Skype client as interface. > - Celliax, GSM and SMS endpoint, makes and receives voice calls and > SMSs to/from the GSM/CDMA network, using second hand cellphones and/or > embedded professional devices as interfaces > > My aims are: > > a) port both endpoints from Asterisk to FreeSWITCH > b) have both endpoints continue to support at least Linux and Windows on FS > c) I would like better having most of the endpoints code working for > both FreeSWITCH and Asterisk, maintaining separated the code that > interface with the GSM and Skype network, from the code that interface > with the core. > > Skypiax, the skype compatible endpoint, is a fork of celliax, the GSM > endpoint, and they share the same skeleton and logic, so porting > celliax after having ported skypiax will be easier and faster :-). > > Current situation and next steps: > 1) skypiax (http://wiki.freeswitch.org/wiki/Skypiax) is now available > for testing and debugging, needs to be polished and cleaned > 2) starting mid next week (I'll be back in office), I want to > integrate into skypiax the code and ideas from mod_airpe of Massimo > (ctrix), that has developed an alternative Skype compatible module, > and coordinate any future development with him and any other > interested developer > 3) begin the porting of celliax to FS, aiming at a pre-beta release > for Linux and Windows during February. > 4) coordinate further development of celliax with any other developer > interested in GSM, SMSs, CDMA, IDEN, AT commands, FBUS commands, > embedded devices, audio sampling > > I am gmaruzz on #freeswitch and #freeswitch-dev, you can find more > info at www.celliax.org. > > Sincerely, > > Giovanni Maruzzelli > ========================================= > www.celliax.org > via Pierlombardo 9, 20135 Milano > Italy > gmaruzz at celliax dot org > Cell : +39-347-2665618 > Fax : +39-02-87390039 >

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