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

[Freeswitch-dev] Mod_commands, originate, and existing sessions
by Kurtis Heimerl 23 Jul '11

23 Jul '11

Hello freeswitch-dev! I've got my chat dialplan working, and set about creating a simple "callback" dialplan. Basically, a chat comes in at a number (e.g., 5000) and the dialplan immediately originates a call between that user and that phone number. This didn't work, as the originate command (in mod_commands) does not allow it to operate when an existing session is active. I've modified it to allow this, and it works great. Callbacks happen, a new session is created, and I see no crashing or major issues. Line ~3400 in mod_command.c //kurtis //if (session || zstr(cmd)) { if (zstr(cmd)) { stream->write_function(stream, "-USAGE %s\n", ORIGINATE_SYNTAX); return SWITCH_STATUS_SUCCESS; } I'd like to have as few distinct blocks of code from vanilla FS as possible, so I was thinking of pushing this change upstream. Would any of the core developers tell me why this particular solution is a bad idea, and why originate disallowed operation in an existing session? If it's benign, I'll submit a patch. If not, I'll try to figure out another way to do this. Thanks!

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[Freeswitch-dev] Execution of channel functions and thread safety
by Peter Olsson 23 Jul '11

23 Jul '11

Hello all developers! I'm in the middle of the development of my own FreeSWITCH module, written in C. Since I've been following the project for quite some time now, it was pretty easy to get things going :) And most of the Core API is straight forward to use - thanks for that all FS developers! I have a minor question of some of the channel functions in FreeSWITCH, and if it's safe to call them from different threads (outside the channel's own thread). For instance; I have a dialplan app registered, and when the call arrives there I do some minor parsing of the call, and after that I park the call (using switch_ivr_park()). Now to my question, I understand that it's probably not safe to call switch_ivr_play_file() from another thread at this time, since the park app itself will then try to handle RTP at the same time as switch_ivr_play_file() (especially if park app sends silence frames) - so instead I queue this to the "playback" app on the channel, similar to how it's done in mod_event_socket. But, what if I call switch_channel_answer(), switch_channel_pre_answer() etc, is that safe to call from another thread, or should I queue these messages to the channel instead - so they are executed within the channel's own thread? My first thought was that these were supposed to be executed by the channel's thread only, but after looking at some of the uuid_-commands in mod_commands I'm not so sure anymore. Since there are quite a few commands messing with the channel's states etc, and that would be executed from another thread indeed (for instance hold/unhold, kill etc). Right now I'm calling switch_channel_answer() from another thread, and it works great, however, I want to be sure I do it right, so this won't cause any problems for me later on in the project. Regards, Peter Olsson PS! See you at ClueCon!

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[Freeswitch-dev] ClueCon 2011 - Sharing rooms
by Michael Collins 20 Jul '11

20 Jul '11

Hello all! It seems that the Sofitel is getting booked up rapidly and it's becoming difficult to find rooms. If you are looking for someone with whom you may share a room, or you have a room that you would like to share, please email me off list. Thanks, Michael

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[Freeswitch-dev] FreeSWITCH Conference Call Today
by Michael Collins 20 Jul '11

20 Jul '11

Hello all! Today's agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2011_07_20 We have a special guest: Alfred from the baresip project is coming in to talk about his cool new SIP client! Come join us today at 1PM eastern/10AM pacific/1700 UTC. -Michael

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[Freeswitch-dev] Urgent Feature Request - RTCP report blocks
by jay binks 19 Jul '11

19 Jul '11

I NEED to get Freeswitch generating RTCP Report Blocks ( with all associated data - packet_loss, jitter etc ) Im happy to consider a bounty on this requirement as I need this done as quickly as humanly possible. in preference to only filling my request, a complete RTCP implementation would be great however currently I simply need RTCP packet generation with Packet Loss Jitter and RTT information. we need this is to fulfill an interop test we are trying to get done, so as you can imagine time is of the essence. please contact me if you wish to work on this, to be eligible for bounty payment the code will need to be accepted into the FS Git tree and pass my testing. ( however I would like to test asap, even before inclusion in git ) I would like your guidance, as to what bounty you think this would require. please contact me for bounty approval. this email does not constitute acceptance of any bounty / work. Sincerely Jay Binks

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[Freeswitch-dev] Associating a Dialplan with a Session
by Kurtis Heimerl 19 Jul '11

19 Jul '11

Hello FS-Dev! I'm trying to figure out how to get associate a dialplan with a newly created Session. I create the new session when I receive an chat message. Here's the current state of the code: http://pastebin.freeswitch.org/16844 There are (undoubtedly) many things wrong with this code... but I'm focused on getting it to route the SMS through the dialplan right now. It fails in the following way: ... 2011-07-18 15:05:49.315430 [DEBUG] mod_smqueue.c:39 laaa CHANNEL ROUTING 2011-07-18 15:05:49.315430 [DEBUG] switch_core_state_machine.c:77 laaa Standard ROUTING 2011-07-18 15:05:49.315430 [INFO] switch_core_state_machine.c:142 No Route, Aborting 2011-07-18 15:05:49.315430 [DEBUG] switch_channel.c:2641 (laaa) Callstate Change RINGING -> HANGUP 2011-07-18 15:05:49.315430 [NOTICE] switch_core_state_machine.c:143 Hangup laaa [CS_ROUTING] [NO_ROUTE_DESTINATION] 2011-07-18 15:05:49.315430 [DEBUG] switch_channel.c:2657 Send signal laaa [KILL] ... The "No Route, Aborting" debug message SEEMS to indicate either that the extension isn't found, or that the dialplan isn't found. I think I've confirmed it's the dialplan by passing NULL instead of "globals.dialplan" in my own code and getting the same result. This is also confirmed by the line: switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "KURTIS %p\n", (void*) switch_loadable_module_get_dialplan_interface("default")); printing "(nil)" instead of returning a valid pointer. I assume there must be some other naming mechanism for dialplans that I'm not using correctly. I pass "default" right now (as I saw in mod_dingaling.c) but that's not working. Any suggestions would be appreciated!

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[Freeswitch-dev] Regarding the patch for Multiple grammar ASR support
by Yungwei Chen 18 Jul '11

18 Jul '11

Hi, I am interested in taking advantages of the patch (http://jira.freeswitch.org/browse/FS-2906) for Multiple grammar ASR support from a javascript program. But I am not sure exactly when to call the following function from a javascript program: ... this.addMultipleGrammars = function () { // hardcoded urls are just for my initial tests. this.session.execute("detect_speech", "grammar " + "http://192.168.16.9/pizza_arso.grxml"); this.session.execute("detect_speech", "grammaron " + "http://192.168.16.9/pizza_arso.grxml"); } ... I tried the following in scripts/js_modules/SpeechTools.jm, but this didn't work because the code snippet at the bottom prevented me from having multiple active grammars. ..... while(this.asr.session.ready() && this.collected_index < this.req) { var x; this.needConfirm = false; if (!rv) { this.asr.addMultipleGrammars(); rv = this.react(this.top_sound, this.top_sound); } ... I had to comment out the following piece of code in recog_asr_load_grammar function of /opt/freeswitch-snapshot/src/mod/asr_tts/mod_unimrcp/mod_unimrcp.c to get my expected results. This fix may not be needed. Perhaps you can point me to the right direction. Thanks. ... start_recognize = (char *) switch_core_hash_find(schannel->params, "start-recognize"); if (zstr(start_recognize) || strcasecmp(start_recognize, "false")) { // this disables all grammars /* switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "calling recog_channel_disable_all_grammars()\n"); if (recog_channel_disable_all_grammars(schannel) != SWITCH_STATUS_SUCCESS) { status = SWITCH_STATUS_FALSE; goto done; } */ switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "calling recog_channel_enable_grammar()\n"); if (recog_channel_enable_grammar(schannel, name) != SWITCH_STATUS_SUCCESS) { status = SWITCH_STATUS_FALSE; goto done; } switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "calling recog_channel_start()\n"); status = recog_channel_start(schannel); }

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Re: [Freeswitch-dev] Freeswitch and Google PUSH (C2DM)
by Nacho 18 Jul '11

18 Jul '11

Hi again. I'm having a little problem with the Java ESL. I can do some basic things like answer the call, subscribe to events, put music in the caller... Thought I have som questions about API commands. The flow right now in my program is this one: - I begin listenning at port 8084 for ESL events - If I receive an event with the name "channel_data" I begin the iteration - I answer the call with *sendMsg, addcallcommand("execute") and addExecuteAppName("answer");* - I subscribe to all events for debuging, I just use *EslMessage response = sendSyncSingleLineCommand(channel, "event text all");* - I put a tone on the caller with *sendMsg, addcallcommand("execute") and addExecuteAppName("playback");* - Now here is a tricky part, I send a petition with HttpURLConnection to an APP server, this server is the one that controls the C2DM stuff, he answers me when he sends the push and I continue the flow, but still have to wait for the legB to connect to the freeswitch server - I have to control when the legB gets connected to freeswitch, this is something i have to work on, i don't want to be making continuous requests to freeswitch to avoid overloading, so right now I assume legB is connected by default. - It's time to originate the call, so I send *sendMsg, addcallcommand("execute"), addExecuteAppName("set") and addExecuteAppArg("{ignore_early_media='true'}sofia/internal/1002%10.166.108.139"); * . *I'm using a default destiny for testing* - I have just tested that this also works *EslMessage response = sendSyncSingleLineCommand(channel, "api originate {ignore_early_media='true'}sofia/internal/1002%10.166.108.139"); * - And finally I bridge the call with SendMsg, addCallCommand("execute"), addExecuteAppName("bridge"), addExecuteAppArg("sofia/internal/1002%10.166.108.139") and addEventLock(); This works well, but right now my major issue is knowing when the legB gets connected to freeswitch so I can begin originating the call, but I have some questions that I would like to know the answer too: - The execute messages are for APP so they work like in the dialplan <action> field, I saw somewhere that to see the complete list one should look in the dialplan.xml, but I haven't found anything about a complete list :S - Things can be accomplished with API calls too, but I just dont get why I have to use "api" before "originate" to make a call, but not needing it for the events, it's a little confusing to use API commands like this, is there any simpler way to do it? - This makes my last question, is there any way to make freeswitch tell me when someone registers in the server or do I need to be continuosly asking freeswith with an API command if it is connected Thinks are going well, I have to begin working with the android client next to be able to process the C2DM messages, right now im using a default voip client. Sorry for the long post and thanks for your attention. Regards Jose P.S: Again, thanks to davidv for making possible the use of a Java ESL client :) 2011/7/11 Nacho <leucaruth(a)gmail.com> > Cool, thanks for the tip David, I'm using your library and have it working. > > The wiki should advice about the SWIGed library and it's problems with Java > and outbound connections. > > In a few days I will have news about the my progress. > > Regards, > > Jose > > 2011/7/11 david varnes <david.varnes(a)gmail.com> > >> Jose, >> >> If you want to use Java and ESL there are other alternatives >> than the SWIGed library. >> >> http://wiki.freeswitch.org/wiki/Java_ESL >> >> I maintain the one here http://wiki.freeswitch.org/wiki/Java_ESL_Client >> >> I have a few pending changes that I want to commit for a new >> release, but the existing release is stable. >> >> hope this helps >> davidv >> >> On 11 July 2011 00:12, Nacho <leucaruth(a)gmail.com> wrote: >> > I just noticed short before you answered :). >> > I have been reading it and I found it very interesting, but I have found >> a >> > problem in Java. >> > The constructor for an Outbound connection requires the File Descriptor >> (an >> > Int parameter) of the listening socket to create the ESL object, but as >> far >> > as I know, Java doesn't allow to get the int value of a File Descriptor >> from >> > a Socket. This would be veary easy to achieve in C, but I have no found >> any >> > means to get this to work with Java in Linux (Windows is out of the >> > question). >> > I tried to see the source code of the ESL client to see how it uses the >> int >> > value, but as I suspected is C code wrapped with SWIG. I have thought >> about >> > other ways to work with it, like using the FileDescriptor class in java >> and >> > Reflection, but I have not found any way to get the desired int value, >> so >> > i'm stuck about what to do right now. >> > Any ideas? if the constructor requires the int value of the File >> Descriptor >> > means that there is a way to get it, doesn't it? >> > Regards, >> > Jose >> > >> > 2011/7/7 Kristian Kielhofner <kris(a)kriskinc.com> >> >> >> >> http://wiki.freeswitch.org/wiki/Event_Socket_Library >> >> >> >> On Thu, Jul 7, 2011 at 7:35 AM, Nacho <leucaruth(a)gmail.com> wrote: >> >> > Hi again >> >> > Im doing as Kristian suggested, at first is an easier solution that >> >> > making a >> >> > module. >> >> > I created a plan that redirects calls to a local port where I will do >> >> > the >> >> > testings with the server that will work with that call. >> >> > The call is parked and the connection is stablished as I can see with >> >> > netcat, so I was thinking about making a program with C or Java that >> >> > will >> >> > become the custom server. The problem is that I don't find info about >> >> > the >> >> > structure of the packages I'll receive from Freeswitch, how to deal >> with >> >> > them or even how can I send orders to freeswitch. >> >> > Any suggestions about where I could find some info about it would be >> >> > kindly >> >> > appreciated. >> >> > Regards, >> >> > Jose. >> >> > 2011/7/6 Robert-iPhone <rhuddleston(a)gmail.com> >> >> >> >> >> >> I have interest too! Considering mwi / voicemail push for apple >> >> >> >> >> >> Sent from my iPhone >> >> >> >> >> >> On Jul 5, 2011, at 5:49 PM, Anthony Minessale >> >> >> <anthony.minessale(a)gmail.com> wrote: >> >> >> >> >> >> > it can be left out of FS code but still part of contrib if an >> >> >> > external >> >> >> > solution is developed. >> >> >> > >> >> >> > >> >> >> > On Tue, Jul 5, 2011 at 1:20 PM, Kristian Kielhofner >> >> >> > <kris(a)kriskinc.com> >> >> >> > wrote: >> >> >> >> Jose, >> >> >> >> >> >> >> >> Whether it's C2DM or APNS (Apple) this functionality is best >> left >> >> >> >> out of FreeSWITCH. FreeSWITCH has plenty of other existing means >> >> >> >> (XML_CURL, socket, etc) to drive dynamic call functionality and >> >> >> >> interact with other technologies. >> >> >> >> >> >> >> >> How do you ask? An imcoming call to FreeSWITCH could hit static >> >> >> >> dialplan and execute a socket connection to some other custom >> >> >> >> server. >> >> >> >> This server could tell FreeSWITCH to do something (play media, >> >> >> >> ringback, messages, whatever) while it sends a PUSH (via C2DM or >> >> >> >> APNS, >> >> >> >> for example). >> >> >> >> >> >> >> >> One could implement such a socket program easily. It can be >> >> >> >> extended and maintained separately as new push technologies >> become >> >> >> >> available. >> >> >> >> >> >> >> >> On Mon, Jul 4, 2011 at 7:48 AM, Nacho <leucaruth(a)gmail.com> >> wrote: >> >> >> >>> Hello all. >> >> >> >>> My name is Jose. This is my first post here. I joined the list >> >> >> >>> because >> >> >> >>> I was >> >> >> >>> wondering about developing a feature for Freeswitch and I would >> >> >> >>> like >> >> >> >>> to know >> >> >> >>> your opinion. >> >> >> >>> The idea is about making a non-persistent client for cell phones >> >> >> >>> that >> >> >> >>> works >> >> >> >>> with PUSH technology (C2DM and Android for example). These >> clients >> >> >> >>> wouldn't >> >> >> >>> be connected to the Freeswitch server at first, but if there is >> an >> >> >> >>> incoming >> >> >> >>> call, the Freeswitch server would send a PUSH message to these >> >> >> >>> clients, the >> >> >> >>> client would process it and, if accepted by the user, the client >> >> >> >>> would >> >> >> >>> awake, connect to Freeswitch and then receive the the call >> invite >> >> >> >>> message >> >> >> >>> and accept it automatically. >> >> >> >>> I don't know if this idea is plausible due to the real time >> >> >> >>> restrictions we >> >> >> >>> have to face in phone calls. If the delay introduced by C2DM >> >> >> >>> delivery >> >> >> >>> is >> >> >> >>> high, the waiting time for the caller is probably something that >> he >> >> >> >>> isn't >> >> >> >>> willing to accept in order to get his call answered. >> >> >> >>> Another question is about the battery life time it would save >> >> >> >>> because >> >> >> >>> the >> >> >> >>> application woulnd't need to be answering the "still alive" ACKS >> >> >> >>> from >> >> >> >>> Freeswitch. >> >> >> >>> I'm new in freeswitch developing and still learning about it, >> but >> >> >> >>> I'm >> >> >> >>> really >> >> >> >>> interested in this, so any help would be kindly appreciated. >> >> >> >>> -- >> >> >> >>> Aquellos que hablan son esclavos de sus palabras y los que >> callan >> >> >> >>> dueños de >> >> >> >>> su silencio. >> >> >> >>> >> >> >> >>> _______________________________________________ >> >> >> >>> 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 >> >> >> >>> >> >> >> >>> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> -- >> >> >> >> Kristian Kielhofner >> >> >> >> >> >> >> >> _______________________________________________ >> >> >> >> 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 >> >> > >> >> > >> >> > >> >> > -- >> >> > Aquellos que hablan son esclavos de sus palabras y los que callan >> dueños >> >> > de >> >> > su silencio. >> >> > >> >> > _______________________________________________ >> >> > 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 >> >> > >> >> > >> >> >> >> >> >> >> >> -- >> >> Kristian Kielhofner >> >> >> >> _______________________________________________ >> >> 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 >> > >> > >> > >> > -- >> > Aquellos que hablan son esclavos de sus palabras y los que callan dueños >> de >> > su silencio. >> > >> > _______________________________________________ >> > 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 >> > >> > >> >> >> >> -- >> david varnes >> >> e: david.varnes(a)gmail.com >> p: +61 404 925 633 >> > > > > -- > Aquellos que hablan son esclavos de sus palabras y los que callan dueños de > su silencio. > -- Aquellos que hablan son esclavos de sus palabras y los que callan dueños de su silencio.

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[Freeswitch-dev] FS-2875 SIP REGISTER Fetch (without Contact header)
by Leon de Rooij 15 Jul '11

15 Jul '11

Hi all, Can someone please take a look at http://jira.freeswitch.org/browse/FS-2875 ? Especially the last few comments and last patch which is here: http://jira.freeswitch.org/secure/attachment/14120/sofia_reg_enable_registe… With this it is possible to fetch bindings by sending a sip register without contact headers. A regular register also responds with all bindings that are in the db now (see traces below). Thanks, Leon --- Traces (I left out headers that are not important for this): * UAC does REGISTER fetch (without contact) when there are no bindings in the db yet: REGISTER sip:172.16.44.8 SIP/2.0 Via: SIP/2.0/UDP 172.16.42.150:5060;rport;branch=z9hG4bKD33480AB9580F76B From: <sip:1000@172.16.44.8>;tag=4041070207 To: <sip:1000@172.16.44.8> Call-ID: 967655ECA0FEA994(a)192.168.178.1 CSeq: 2 REGISTER SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.42.150:5060;rport=5060;branch=z9hG4bKD33480AB9580F76B From: <sip:1000@172.16.44.8>;tag=4041070207 To: <sip:1000@172.16.44.8>;tag=48QmcmccUc50B Call-ID: 967655ECA0FEA994(a)192.168.178.1 CSeq: 2 REGISTER Date: Fri, 15 Jul 2011 10:39:07 GMT * UAC does REGISTER (with contact) when there are no bindings in the db yet: REGISTER sip:172.16.44.8 SIP/2.0 Via: SIP/2.0/UDP 172.16.42.150:5060;rport;branch=z9hG4bKB8AA233F4F985F7C From: <sip:1000@172.16.44.8>;tag=4041070207 To: <sip:1000@172.16.44.8> Call-ID: 967655ECA0FEA994(a)192.168.178.1 CSeq: 3 REGISTER Contact: <sip:1000@172.16.42.150;uniq=87E52E228E505C8B8957ADA870829> Expires: 1800 SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.42.150:5060;rport=5060;branch=z9hG4bKB8AA233F4F985F7C From: <sip:1000@172.16.44.8>;tag=4041070207 To: <sip:1000@172.16.44.8>;tag=5HHDeFXFrNUKQ Call-ID: 967655ECA0FEA994(a)192.168.178.1 CSeq: 3 REGISTER Contact: <sip:1000@172.16.42.150;uniq=87E52E228E505C8B8957ADA870829>;expires=1800 Date: Fri, 15 Jul 2011 10:39:07 GMT * UAC does REGISTER (with contact) when there is already one binding in the db: REGISTER sip:172.16.44.8 SIP/2.0 Via: SIP/2.0/UDP 172.16.42.126:10818;branch=z9hG4bK-d8754z-a7869a477c0f317c-1---d8754z-;rport Contact: <sip:1000@172.16.42.126:10818;rinstance=f2aff35495d275bc> To: "1000"<sip:1000@172.16.44.8> From: "1000"<sip:1000@172.16.44.8>;tag=cf168522 Call-ID: MTAyZDJkMTUyN2ViYTE1NmFkNjFiNDc1ZDkwNWQwYTc. CSeq: 2 REGISTER Expires: 3600 SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.42.126:10818;branch=z9hG4bK-d8754z-a7869a477c0f317c-1---d8754z-;rport=10818 From: "1000"<sip:1000@172.16.44.8>;tag=cf168522 To: "1000" <sip:1000@172.16.44.8>;tag=ZrQtFUm2rpjtB Call-ID: MTAyZDJkMTUyN2ViYTE1NmFkNjFiNDc1ZDkwNWQwYTc. CSeq: 2 REGISTER Contact: <sip:1000@172.16.42.150;uniq=87E52E228E505C8B8957ADA870829>;expires=1787 Contact: <sip:1000@172.16.42.126:10818;rinstance=f2aff35495d275bc>;expires=3600 Date: Fri, 15 Jul 2011 11:08:10 GMT

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[Freeswitch-dev] FreeSWITCH Conference Call Today
by Michael Collins 13 Jul '11

13 Jul '11

Hello all, Today's agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2011_07_13 We have a few things to discuss as a community, plus a cool video to share. We are especially interested in your thoughts on the subject of FreeSWITCH "config sets" like what's here: http://svn.freeswitch.org/svn/configs/ Please be ready to share your config ideas with the group. Talk to you soon! -Michael

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