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

[Freeswitch-dev] FreeSWITCH Wait Times
by Matthew Margolis 18 Aug '11

18 Aug '11

Hey is there any possible way to have a wait time posted in the fifo queue? Something like "Thank you for waiting. Your current wait time is 10 minutes." Matthew Margolis Programmer PATLive T 850.766.3236 E matthew.margolis(a)patlive.com

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[Freeswitch-dev] Speaker Identification implementation
by Viktor Jamrich 18 Aug '11

18 Aug '11

That's it ! I was talking about speaker identification by comparing the speaker's voiceprint with pre-trained voiceprint saved in a voiceprint repository. Sure, I've heard about Nuance is planning to acquire Loquendo, another speech processing shark ;) I was thinking about creating a new abstract interface called e.g. switch_sid_interface (SID = speaker identification) which could be binded to e.g. a proprietary interface module or (the better solution) to mod_unimrcp since MRCPv2 protocol supports speaker identification/verification. Using this sid interface w/ mod_unimrcp FreeSWITCH would be able to use MRCP servers for speaker identification/verification. Currently Nuance and Loquendo has few MRCP servers supporting this service and of course our company is planning to deploy a MRCPv2 server soon. Example application is e.g. instead of looking for your customer ID or telling your personal ID data to the operator who needs to have your identity confirmed before he/she tells you some confidental info, the company (whether call center or whatever) would just use speaker identification service via MRCPv2 and from 3 to 5 seconds of voice he/she will know, whether the person is really the one he/she is claiming he/she is or not. What do you think about it ?? Have a nice day ;) Sincerely, Viktor Jamrich.

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[Freeswitch-dev] FreeSWITCH Conference Call Today
by Michael Collins 17 Aug '11

17 Aug '11

Hello all! We are going to have a conf call today and talk about a few things, like how awesome ClueCon was this year! http://wiki.freeswitch.org/wiki/FS_weekly_2011_08_17 Come join us and be part of the discussion. I think we might have a few first-timers involved also. Thanks, MC

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[Freeswitch-dev] Meetup Paris?
by Mathieu Rene 17 Aug '11

17 Aug '11

English version follows. En cette fin de ClueCon, je pars sur Paris demain soir (donc j'arrive samedi en matiné, bien décalé), et j'y resterai jusqu'à mercredi. Je propose d'organiser un petit meetup afin de joindre l'utile à l'agréable et de voir un peu le visage de la communauté freeswitch parisienne. Est-ce quelqu'un peut proposer un bar sympa ou on pourrait faire un truc? --- ClueCon is over and I'm leaving to Paris tomorrow (landing saturday morning, well jetlagged) and staying until Wednesday. I'm proposing to organize a little meetup in order to get to know the face of the parisian freeswitch community. Does anyone know a bar where we could meet up? Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene(a)avgs.ca

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[Freeswitch-dev] Speaker Identification Implementation
by Viktor Jamrich 15 Aug '11

15 Aug '11

Hi guys, our company wants to use speaker identification/verification (SID) in FreeSWITCH. However, only ASR and TTS are currently implemented so we decided to contribute by coding the SID interface as well as speaker identification/verification part in mod_mrcp. Is it possible to get our code commited and made available to public within a month ?? And what's the procedure in getting our code commited ?? I think there's a huge gap on the FS web in information for newbies wanting to help with coding... PS: I'm not sure whether this is the right place to ask such question but I've been to #asterisk-dev on IRC and nobody answered my question... Thanks for help.

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[Freeswitch-dev] how to get dnis into mod_callcenter?
by Royce Mitchell III 15 Aug '11

15 Aug '11

I would like a few features added to mod_callcenter. I have already implemented 3 of the 4 I want. The 4th one is what I need a bit of advice on. 1) I need the ability to query an agent's state when logging in. The problem is, if my software crashes, I'd like to be able to restart the software while the agent is still on the phone and be able to read the agent's current state. The only way I've figure out how to do this is via "callcenter_config agent list". The problem is that returns a list of all agents which could be inefficient if there are a lot of agents in the system. Therefore I have implemented the following in the attached patch: add an optional [{agent_name}] to the end of agent list, so the function call looks like this: "callcenter_config agent list [[agent_name]]". I'm using the double [[ ]] to indicate that [agent_name] would be optional and, if present, that you only want the information from that command for the indicated agent. Actual example: "callcenter_config agent list 1001(a)127.0.0.1" 2) There's a "set status", "get status", and "set state", but no "get state". I have implemented "callcenter_config agent get state <agent_name>". Actual example: "callcenter_config agent get state 1001(a)127.0.0.1" 3) There's no way to directly query an agent's uuid except via agent list, so I implemented "callcenter_config agent get uuid <agent_name>". Actual example: "callcenter_config agent get uuid 1001(a)127.0.0.1" Finally, this is where I'm stuck... I'm not sure how to implement this last feature... 4) I need to get the caller's DNIS via cti when a call is delivered. I am getting all the information I need with an event socket filter of "filter CC-Agent 1001(a)127.0.0.1" except for the DNIS. Right now I'm having to add another filter, "filter variable_cc_agent 1001(a)127.0.0.1" in order to get that information from the "Caller-RDNIS" field, but that's causing me to get a whole lot more events than I want/need. I want to add a new field called "CC-Member-DNIS" that will be caught with a "CC-Agent" filter, but I'm not sure how to get the DNIS information to create that header. -- There's a fine line between genius and insanity. I like to use it for dental floss.

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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] Great ClueCon!
by Ali R. 12 Aug '11

12 Aug '11

Hello All, Back to work again; Thanks for this great conference. It was my first time at ClueCon and I will definitely be there next year!!!!! Ali R.

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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] help on GDB with core dump
by Seven Du 05 Aug '11

05 Aug '11

Hi, I'm debugging core dump in GDB, although I reported a jura issue I'd like to do something more useful that waiting Can someone give me some highlight on the message? http://pastebin.freeswitch.org/16993 from the first two lines I don't believe pthread_rwlock_wrlock has a problem and remove_session_elem_from_listener_locked also looks normal, where should I look more? Thanks. #0 0x00000034bcc0a48a in pthread_rwlock_wrlock () from /lib64/libpthread.so.0 #1 0x00002aaaab9fba50 in remove_session_elem_from_listener_locked (listener=0x5cff788, session_element=0x1) at -- Seven Du About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn Sent with Sparrow (http://www.sparrowmailapp.com)

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