July 2011 - freeswitch-dev@lists.freeswitch.org

[Freeswitch-dev] mod_portaudio multiple endpoints support
by Moises Silva 25 Oct '11

25 Oct '11

Hello, I've been doing some changes to mod_portaudio sponsored by Comrex Corporation. The changes involve adding support for using multiple audio devices simultaneously, including devices with multiple channels (note that is not exactly the same as stereo). The changes are available at git:// github.com/moises-silva/mod_portaudio-endpoints.git Complete documentation on the new functionality in portaudio can be found in the sample configuration in the form of comments (conf/autoload_configs/portaudio.conf.xml). The portaudio changes are totally backwards compatible, existing configurations should keep working as expected. In addition to the new portaudio functionality there are a few other changes that were necessary to use the new functionality. All the changes described below: 1. mod_portaudio.c and pablio.c modifications to allow usage of multiple streams, channels and devices. 2. uuid_outgoing_answer API added to mod_commands.c, this API allows to answer outgoing channels. This may only be useful for portaudio endpoints since most, if not all, endpoints receive the answer notification using some sort of signaling message (200 OK for example). 3. Changes to switch_generate_sln_silence() in switch_resample.c to allow specifying -1 as a special divisor value meaning to zero-out the buffer completely. This option was required by Comrex too since the equipment they have seems to be amplifying the random low-level noise that switch_generate_sln_silence() currently generates by default and they don't like the "hissing noise" on the background while the call is answered and there is no ringback tone. I'd appreciate anyone who can review and comment. I just merged a few moments ago with latest git therefore reviewing should be easy with git diff adding the github repo as a remote. I'd like to get approval from the core FreeSWITCH team to merge this into mainstream. Needless to say I'll be available in the case any bugs are introduced in the portaudio code, virtually impossible, but just in case :-) Thanks, - Moy

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[Freeswitch-dev] bounty request - sofia to not route MESSAGE [was] Chat Dialplan
by david varnes 14 Aug '11

14 Aug '11

I raised a bounty jira http://jira.freeswitch.org/browse/FS-3408 which is a request to modify the native sofia treatment of incoming MESSAGE, and generate some events to allow alternate handling. I also added it to the bottom of http://wiki.freeswitch.org/wiki/Bounty I was just reading back through the list and saw this discussion, it seems pretty much what Kurtis was suggesting as well ? Given the rest of the previous thread, is this request (FS-3408) likely to get any traction ? Is there anything else I should/could do to get a yes/no answer. The reason I ask is that I have a client that would like to use this kind of feature, and will try and do the modification themselves if the change does not fit with FS dev direction. I think it would be much better if the change was available in FS though. What amount of work would it be ? thanks for any feedback :-) davidv On 7 July 2011 10:50, Kurtis Heimerl <kheimerl(a)cs.berkeley.edu> wrote: > Let me repeat myself: the changes I am proposing will be backwards > compatible. Nothing will be broken. > > The dialplan abstraction is tangential to this whole discussion. I can > build a mod foo which runs the dialplan, and have all messages routed > to it via the + mechanism. This will probably be my initial version of > the system. I understand the work you put into this, and that your > decisions were well-reasoned for the use cases you envisioned (e.g., > conference calls), but the abstraction isn't working for me. This > system is just a piece, interoperating with a variety of other > systems. Attaching a specific module tag, used by just one piece of > one program, to a username in use throughout the entire system is not > an elegant solution. This means I have to put a proxy in front of FS, > adding this + tag to every message coming in. > > I'm here and open to any solutions you might have to make the existing > solution work for me. Fundamentally, I can't just shove a +tag onto > every incoming message in any elegant way, so far as I know. I think > that means this entire line of reasoning is just broken for me. Don't > get me wrong here, I would be really really happy if I didn't have to > touch sofia to solve this, it's really hard to grok. It's also a lot > less work for me. > > So far, I've proposed one alternate method for doing this. I want > sofia to generate events and not route messages itself. Then I can > listen to these perform whatever actions I need to. This may not work, > and I'm open to implementing any other architectural changes you think > would enable my use case. You're clearly the guy who knows what to do, > and your advice is invaluable. > > On Wed, Jul 6, 2011 at 5:23 PM, Anthony Minessale > <anthony.minessale(a)gmail.com> wrote: >> you control the user names on your system the prefix+ string is just >> part of the username people would talk to. >> >> for instance ext+ namespace is has auto presence built in so if you >> add ext+1000 to your list it will auto advertise that it's online. >> >> conf+ goes to the conference. >> >> conf+888(a)server.com will go to FS on server.com and route the chat >> messages to the mod_conference. >> >> so basically you are proposing to try to modify sofia and break the >> other stuff because you do not like the prefix string on the user >> names but its designed to allow you to cross connect protocols, eg >> mod_sofia has sip+ registered and mod_dingaling has jingle+ so you can >> use jabber to pass sip+user(a)domain.com to your buddy list and have it >> report accurate presence. >> >> Basically you make a new module called mod_foo and register the foo namespace. >> Then on your clients you subscribe to foo+1000(a)server.com for instance. >> Chat messages to that foo namespace will arrive in your code and you >> have the option to reply to them completely agnostic of the protocol >> sip, jabber etc. >> >> You seem to be asking for advice but then ignoring it coming from the >> author of all of the above who spent many man hours solving this >> problem so it would be abstract. so I'm lost for what else to tell >> you. >> >> Maybe we can confirm that you are simply taken aback by the idea of >> starting all your usernames with foo+ and simply propose to unravel >> everything in pursuit of removing it? >> >> >> >> >> >> On Wed, Jul 6, 2011 at 6:14 PM, Kurtis Heimerl <kheimerl(a)cs.berkeley.edu> wrote: >>> Nope, normal SIP MESSAGE events, no invites. >>> >>> I just need an easy place where an app can look up some variables in >>> the directory, check presence information, and start a call based on >>> an incoming chat event. I also need a system that interops with >>> existing SIP chat services. I don't think the +routing does these for >>> me. >>> >>> On Wed, Jul 6, 2011 at 8:46 AM, Anthony Minessale >>> <anthony.minessale(a)gmail.com> wrote: >>>> Are we talking about that ridiculous new chat concept in SIP where it >>>> uses INVITES and dialogs? >>>> >>>> >>>> On Wed, Jul 6, 2011 at 12:56 AM, Kurtis Heimerl >>>> <kheimerl(a)cs.berkeley.edu> wrote: >>>>> I think I understand it. Basically, a client can prepend a +code that >>>>> causes the message to be routed to a specific module. I don't think >>>>> that fulfills my requirements. This is fine for basic routing, but >>>>> it's not a very good programmable environment for chat applications, >>>>> unless I'm missing something, which is always possible. >>>>> >>>>> If I am, can you direct me to anything that explains it in a little >>>>> more depth than your email? >>>>> >>>>> On Tue, Jul 5, 2011 at 10:49 PM, Anthony Minessale >>>>> <anthony.minessale(a)gmail.com> wrote: >>>>>> Read my response and my mod conference example again. >>>>>> You can register a chat callback from any module bound to a particular >>>>>> namespace....... >>>>>> >>>>>> On Jul 6, 2011 12:35 AM, "Michael Collins" <msc(a)freeswitch.org> wrote: >>>>>>> On Tue, Jul 5, 2011 at 10:20 PM, Kurtis Heimerl >>>>>>> <kheimerl(a)cs.berkeley.edu>wrote: >>>>>>> >>>>>>>> I don't think so. I set the ESL to log all events (/event all) from >>>>>>>> the command line and received no events when sofia received SIMPLE >>>>>>>> messages. Is that a bug? I had assumed it's intentional. >>>>>>>> >>>>>>> I'm 99% sure that those messages never make it up into FS from Sofia. I'd >>>>>>> go >>>>>>> look into the code but Sofia is really scary and there's no amount of >>>>>>> Scooby >>>>>>> Snacks you could give me to convince me to wander into that creepy old >>>>>>> code. >>>>>>> :) >>>>>>> -MC >>>>>> >>>>>> _______________________________________________ >>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago >>>>>> http://www.cluecon.com 877-7-4ACLUE >>>>>> >>>>>> FreeSWITCH-dev mailing list >>>>>> FreeSWITCH-dev(a)lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> _______________________________________________ >>>>> Join us at ClueCon 2011, Aug 9-11, Chicago >>>>> http://www.cluecon.com 877-7-4ACLUE >>>>> >>>>> FreeSWITCH-dev mailing list >>>>> FreeSWITCH-dev(a)lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> -- >>>> 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 >>>> >>>> _______________________________________________ >>>> Join us at ClueCon 2011, Aug 9-11, Chicago >>>> http://www.cluecon.com 877-7-4ACLUE >>>> >>>> FreeSWITCH-dev mailing list >>>> FreeSWITCH-dev(a)lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >>>> http://www.freeswitch.org >>>> >>> >>> _______________________________________________ >>> Join us at ClueCon 2011, Aug 9-11, Chicago >>> http://www.cluecon.com 877-7-4ACLUE >>> >>> FreeSWITCH-dev mailing list >>> FreeSWITCH-dev(a)lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >>> http://www.freeswitch.org >>> >> >> -- >> Anthony Minessale II >> -- david varnes e: david.varnes(a)gmail.com

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[Freeswitch-dev] Sipx TLS connections make Freeswitch crash
by barisyanar 11 Aug '11

11 Aug '11

Hi all, I am using Freeswitch 1.0.7 with sipx release-4.4. Whenever I try to make an IVR call with TLS, Freeswitch crashes dumping the attached core file. The output bt command is: #0 __strchr_sse42 () at ../sysdeps/x86_64/multiarch/strchr.S:131 #1 0x00007f9c0b4373c3 in sofia_glue_get_url_from_contact (buf=0x0, to_dup=0 '\000') at sofia_glue.c:4943 #2 0x00007f9c0b438e1b in sofia_overcome_sip_uri_weakness ( session=0x7f9c04017108, uri=<value optimized out>, transport=SOFIA_TRANSPORT_TCP_TLS, uri_only=SWITCH_TRUE, params=0x0) at sofia_glue.c:1296 #3 0x00007f9c0b4182b0 in sofia_handle_sip_i_invite (nua=0x7f9c04008890, profile=0x148dad0, nh=0x7f9c00009ec0, sofia_private=<value optimized out>, sip=<value optimized out>, tags=<value optimized out>) at sofia.c:6793 #4 0x00007f9c0b433aca in sofia_event_callback (event=nua_i_invite, status=100, phrase=0x7f9c0000ab40 "Trying", nua=0x7f9c04008890, profile=0x148dad0, nh=0x7f9c00009ec0, sofia_private=0x0, sip=0x7f9c00006658, tags=0x7f9c0000ab30) at sofia.c:942 #5 0x00007f9c0b4a5f53 in nua_application_event (dummy=<value optimized out>, sumsg=<value optimized out>, ee=0x7f9c0000ab08) at nua_stack.c:393 #6 0x00007f9c0b503974 in su_base_port_execute_msgs (queue=0x0) at su_base_port.c:280 #7 0x00007f9c0b503ec1 in su_base_port_step (self=0x7f9c040010f0, tout=0) at su_base_port.c:473 #8 0x00007f9c0b42d4ea in sofia_profile_thread_run ( thread=<value optimized out>, obj=0x148dad0) at sofia.c:1623 #9 0x0000003efd2068e0 in start_thread (arg=0x7f9c18116710) ---Type <return> to continue, or q <return> to quit--- at pthread_create.c:297 #10 0x0000003efcee0c9d in clone () And sipx's freeswitch log prints the line: 2011-05-16 15:04:51.530426 [WARNING] switch_core.c:1120 Cannot locate domain deneme.deneme.karelarge.com, where deneme.deneme.karelarge.com is my fqdn. No problem occurs during TCP connection. Problem is possibly caused by sipx, but i'll be glad to hear FS community's ideas.

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[Freeswitch-dev] FINAL UPDATE: ClueCon Hotel
by Michael Collins 27 Jul '11

27 Jul '11

Thanks for your patience - we really wanted to make sure the word got out. The updated information about the hotel can be found on our website: http://cluecon.com/hotel Keep in mind that you MUST ask for "in-house reservations" in order to get to the right operator and get your special rate. Any questions please email me off list. Thanks! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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[Freeswitch-dev] ClueCon Hotel Update - Even *Better* Rates For The Talbott
by Michael Collins 27 Jul '11

27 Jul '11

Brian West is racking up the karma points today! He has finished talking to The Talbott and has secured a rate of $148 per night! Everyone thank Brian for his hard work on this - he's really outdone himself, especially when you consider that Tony has been on vacation for more than a week. If you're so inclined, you can visit Brian's wishlist here<http://www.amazon.com/gp/registry/wishlist/1BWDJUX5LYQE0>. :) Thanks. -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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[Freeswitch-dev] GOOD NEWS: Alternate ClueCon Hotel Arrangements
by Michael Collins 27 Jul '11

27 Jul '11

Everyone say thanks to Brian West for making new hotel arrangements! If you need a room for ClueCon then we've got a great deal for you: $169 per night and you still get the $699 ClueCon rate! The new hotel: The Talbott 20 E Delaware Place Chicago, IL 60611 (312) 944-4970 Go get your room before we sell this one out, too! Thanks for supporting ClueCon. -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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[Freeswitch-dev] ClueCon 2011 Update - Sofitel Full, Alternate Hotel Arrangements Being Made
by Michael Collins 27 Jul '11

27 Jul '11

Hello! Thank you all for supporting ClueCon 2011! We've maxed out the Sofitel, so we're looking at another hotel for our attendees. If you are coming to ClueCon and need a hotel room please contact Brian West ASAP at marketing(a)cluecon.com or 877-742-CLUE. We need to know how many rooms we can guarantee to another hotel in order to get discounted rates. Time is of the essence, so please contact us right away. Thanks! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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[Freeswitch-dev] FreeSWITCH Conference Call Today
by Michael Collins 27 Jul '11

27 Jul '11

Hey all, We are having a simple FreeSWITCH conf call today: http://wiki.freeswitch.org/wiki/FS_weekly_2011_07_27 I want to talk about mod_fifo specifically and call queuing in general. If you use mod_fifo at all please join us. There have been a lot of updates to mod_fifo that make it more "ACD-like" but the wiki page needs some serious attention. Please hop on and help us get the page updated. Thanks! -MC

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[Freeswitch-dev] Wait times
by Matthew Margolis 26 Jul '11

26 Jul '11

Is there any way to set a fifo queue to automatically tell callers on hold an approximate time they have left to wait every minute? Something like "Thank you for holding, your current wait time is 'X' minutes." If this feature doesn't exist, it would probably be very useful. Matthew Margolis Programmer PATLive T 850.766.3236 E matthew.margolis(a)patlive.com

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[Freeswitch-dev] Bug in http://fisheye.freeswitch.org/browse/freeswitch.git/clients/flex/freeswitch.html
by Dmitry Kravchenko 25 Jul '11

25 Jul '11

In order web telephone works correctly one should replace line 4 <script language="javascript" content-type="text/javascript" src="http://ajax.googleapis.com/ajax/libs/jquery/1.4/jquery.min.js"></script> for one referring local jquery-1.4.2.js<http://fisheye.freeswitch.org/browse/freeswitch.git/clients/flex/jquery-1.4…> which is not used otherwise. Page is not working with line above. I.e. it should be <script language="javascript" content-type="text/javascript" src="jquery-1.4.2.js <http://ajax.googleapis.com/ajax/libs/jquery/1.4/jquery.min.js>"></script> Regards, Dmitry.

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