Re: [Freeswitch-users] switch_core_io.c:118 sofia/internal/XXXXX has no read codec.
Hello Jason, Sorry for the delay in answering - I saw your reply only now as it got burried with some other stuff... Anyway, I attach bellow the relevant sip trace. Phone 80678 (132.64.4.137) is calling 80679 (132.64.4.135) which answers. When 80679 presses the Hold or Transfer button the call is disconnected. Thanks! __Yehavi: 2009/9/8 Jason White ja...@jasonjgw.net Yehavi Bourvine yehavi.bourv...@gmail.com wrote: I have a problem when trying to put a call on hold: I get the above message and the call is disconnected. Any idea where to look for the source of the problem? My next step in your situation would be to obtain a Sip trace and post relevant details from it to the list. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org === HERE IS THE INITIAL INVITE = recv 1407 bytes from udp/[132.64.4.137]:2048 at 06:31:26.925580: INVITE sip:80...@pbx-dev.cc.huji.ac.il;user=phone SIP/2.0 Via: SIP/2.0/UDP 132.64.4.137:2048;branch=z9hG4bK-j6ogg8cwv5nm;rport From: Test Yehavi SNOM sip:80...@pbx-dev.cc.huji.ac.il;tag=j1fjgvgf7e To: sip:80...@pbx-dev.cc.huji.ac.il;user=phone Call-ID: 3c2696db6dfb-z5x2h00d9zcw CSeq: 2 INVITE Max-Forwards: 70 Contact: sip:80...@132.64.4.137:2048;reg-id=1 P-Key-Flags: keys=3 User-Agent: snom320/7.3.14 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change, Remote-Aprty-ID Session-Expires: 3600;refresher=uas Min-SE: 90 Proxy-Authorization: Digest username=80678,realm=pbx-dev.cc.huji.ac.il,nonce=27cd67de-dfbf-4a05-8e19-edfc00d159b5,uri=sip:80...@pbx-dev.cc.huji.ac.il;user=phone,qop=auth,nc=0001,cnonce=044d5d78,response=a29e4873f5e72ebbd5e526cc45e1de0d,algorithm=MD5 Content-Type: application/sdp Content-Length: 388 v=0 o=root 1073374100 1073374100 IN IP4 132.64.4.137 s=call c=IN IP4 132.64.4.137 t=0 0 m=audio 60606 RTP/AVP 8 0 9 99 3 18 4 101 a=direction:both a=rtpmap:8 pcma/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:9 g722/8000 a=rtpmap:99 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv send 342 bytes to udp/[132.64.4.137]:2048 at 06:31:26.937621: SIP/2.0 100 Trying Via: SIP/2.0/UDP 132.64.4.137:2048;branch=z9hG4bK-j6ogg8cwv5nm;rport=2048 From: Test Yehavi SNOM sip:80...@pbx-dev.cc.huji.ac.il;tag=j1fjgvgf7e To: sip:80...@pbx-dev.cc.huji.ac.il;user=phone Call-ID: 3c2696db6dfb-z5x2h00d9zcw CSeq: 2 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.4-exported Content-Length: 0 ** invite **80678 80679 [36m2009-09-15 09:31:27.214557 [NOTICE] switch_channel.c:602 New Channel sofia/internal/80...@pbx-dev.cc.huji.ac.il [98e4ed2c-fb37-4a19-9fa7-268bb04413f8] [32m2009-09-15 09:31:27.275406 [INFO] mod_dialplan_xml.c:315 Processing Test Yehavi SNOM-80679 in context huji -- send 1233 bytes to udp/[132.64.4.135]:5060 at 06:31:29.313289: INVITE sip:80...@132.64.4.135 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.164;rport;branch=z9hG4bKFrN8j6K4Ncjea Max-Forwards: 69 From: n8 l8 sip:80...@132.64.9.164;tag=9aSUyZB0m7y8N To: sip:80...@132.64.4.135 Call-ID: 3e2a8eb9-1c64-122d-4aa2-0002b35fc481 CSeq: 120379808 INVITE Contact: sip:mod_so...@132.64.9.164:5060 User-Agent: FreeSWITCH-mod_sofia/1.0.4-exported Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 376 P-Key-Flags: keys=3 Remote-Party-ID: n8 l8 sip:80...@132.64.9.164;party=calling;screen=yes;privacy=off v=0 o=root 1073374100 1073374100 IN IP4 132.64.4.137 s=call c=IN IP4 132.64.4.137 t=0 0 m=audio 60606 RTP/AVP 8 0 9 99 3 18 4 101 a=rtpmap:8 pcma/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:9 g722/8000 a=rtpmap:99 g726-32/8000
Re: [Freeswitch-users] CALL FOR VOLUNTEERS: Assisting With FreeSWITCH Subprojects
Hi Michael / FS enthusiast, In my opinion, it will be an undeniably perfect chaos if everyone wants to do everything. One way I can suggest out of this is the following: - sets of persons that does all the stuff on the FS core. I'm not sure if this one will work out but I think these are the only persons who has commit access to the source code and will also be willing to accept modifications, enhancements, etc. - individuals can ask if they can be the maintainer/tester for a particular module. Bug fixing, sending modifications and enhancements will still be subject to the approval of anthm. - an array of persons that will take ownership on what FS can do and provide realiable working examples for it. Currently, it takes amount of pain for a newbie on how to configure SIP because the examples are too confusing and sometimes there is not certainty if this will work on what release of FS or not. - any individual that has time and interest can start owning the porting of FS to other operating system. Like in my case, I am working on making FS ported to FreeBSD ports but I don't know if somebody did it or if he is currently doing it, as to what stage he is in? - there should be a release engineering team. In my opinion, a stable and current release will be much more sane. As with any other successful open source project, we won't be starting to say I want to contribute this and start working on that. We should indicate that you take ownership of this and any comments should be forwarded right unto you. Again, this is just my opinion. You can take some of it or leave it using sudo rm -rf blah/* . There and back again, Demuel I. Bendano a.k.a engrxyz Hi Michael, You can count with me for anything else, like documentation, coding/scripting, or any other FreeSWITCH related stuff. Regards, Diego 2009/9/14 João Mesquita jmesqu...@freeswitch.org You can assign two things to me. 1. libesl code documentation (partially done and Doxygened - needs cleaning) 2. Bug marshal. I am setting up the proper lab environment here to be able to test most stuff. Count me in for any questions I can answer and I am _always_ on IRC jmesquita On Mon, Sep 14, 2009 at 1:33 PM, Michael Collins m...@freeswitch.orgwrote: Hello FreeSWITCHers! We are looking for people who are in a position to help out with various subprojects that will help FreeSWITCH to keep growing. We need people to help out in these basic areas: Bug marshals (people who watch JIRA and test bug reports, patches, etc.) Documentation maintainers (people who update the wiki when new stuff comes out, also those familiar with mediawiki administration) Documentation authors (people who write new docs, how-to's, tutorials, examples, etc.) Package maintainers (people who manage Debian debs, RPMs, etc.) Additionally, we are always looking for more folks to assist with answering questions on IRC and the mailing list. It is definitely nice to have people who've gone through the pains of switching to FreeSWITCH (or learning it from scratch) who can assist the steady stream of new users. If you want to help and aren't sure where to go from here then please at least do the following: #1 - Join #freeswitch on irc.freenode.net and hang out as much as possible #2 - Check the recent changes link on wiki.freeswitch.org each day #3 - Join the Friday public conference call and listen in These three things, in addition to the mailing list, will keep you well in tune with the FreeSWITCH community and what's happening. Next, make a note of the parts of FS that you use frequently, know a lot about, or are particularly passionate about. Those are the items we'd love to have you help us with. For example: if you use mod_xml_curl frequently and have been through the set up process then you're a prime candidate to help answer questions, refine the mod_xml_curl wiki documentation, write up a tutorial, contribute a working example of a web server database schema, etc. If you are good with a scripting language then we could definitely use help with rounding out the docs for your favorite language. We could also use code samples, so ask for a contrib folder if you have things you would like to share. Or how about this: you read something on the wiki, it doesn't quite work when you try, so you tinker until you figure it out. Now you're in a position to update the wiki for everyone else's benefit, too. As you can see, you don't have to be a FreeSWITCH expert before you can help the project. What we really need are people who care about the project and want to see it flourish. If you are such a person then please contact me off list. Tell me what you're good at or where you would like to help. Many thanks for all of your support! -Michael ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org
[Freeswitch-users] stange segfaults with log
Hello we have a strange problem with 14144 revision. It seems switch_log_printf got NULL pointer as data. It happens a few times. however in the previous frame session seems to be good for us. http://pastebin.freeswitch.org/10357 Could you please tell me what is the problem? Did we make some mistakes with building? Missed make clean, or someting? Thanks in advance, Tamas ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Limit_Hash
Hi, I've moved this discussion to users as it seems my query is moving in that direction :) So, upon looking at limit_hash, it appears to do what I need to do. My question then becomes, how do I set a hash for an originated call? It seems that limit_hash is an application rather than a channel variable, and so far I've been doing most things without touching the dialplan. So, say I want to originate 9 calls, 3 from 3 customers. I would like to mark the calls with my_customer_group_1 through 3, and then use the limit_hash_usage command to verify the count of channels in each group. I therefore have a few questions: 1. Can I mark a call in the originate statement? 2. How do I use the limit_hash_usage command? The wiki states: You can verify the usage of any resource with the limit_hash_usage api call. limit_hash_usage realm id Is realm the same as a SIP realm? Is id the hash that I have used to mark the call with? Just making sure :) -- Cheers, Matt Riddell Director ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] switch_core_io.c:118 sofia/internal/XXXXX has no read codec.
Do you have Late Negotiation on? Also is this the only FreeSWITCH log output you have in this transfer? /b On Sep 15, 2009, at 1:55 AM, Yehavi Bourvine wrote: Hello Jason, Sorry for the delay in answering - I saw your reply only now as it got burried with some other stuff... Anyway, I attach bellow the relevant sip trace. Phone 80678 (132.64.4.137) is calling 80679 (132.64.4.135) which answers. When 80679 presses the Hold or Transfer button the call is disconnected. Thanks! __Yehavi: ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] switch_core_io.c:118 sofia/internal/XXXXX has no read codec.
No, I have late negotiation commented out. This is the only log from the beginning of the session until it disconnects. Shall I turn on more debugging (if available)? Thanks, __Yehavi: 2009/9/15 Brian West br...@freeswitch.org Do you have Late Negotiation on? Also is this the only FreeSWITCH log output you have in this transfer? On Sep 15, 2009, at 1:55 AM, Yehavi Bourvine wrote: Hello Jason, Sorry for the delay in answering - I saw your reply only now as it got burried with some other stuff... Anyway, I attach bellow the relevant sip trace. Phone 80678 (132.64.4.137) is calling 80679 (132.64.4.135) which answers. When 80679 presses the Hold or Transfer button the call is disconnected. Thanks! __Yehavi: ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@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] barge in implementation with mod_socket and eavesdrop
Hello! I'm trying to implement barge in functionality (see http://www.yourdictionary.com/telecom/barge-in) with eavesdrop but still with no success. The situation is: - Person A calls to the extension: extension name=some_ext condition field=destination_number expression=^900.$ action application=answer/ action application=park/ /condition /extension - I bridge him with person B with help of mod_socket: SendMsg some_uuid call-command: execute execute-app-name: bridge execute-app-arg: person B address in form: user/... - A and B talks - Person C decides to barge in the call A--B (to become a third participator in the call) a) I send (mod_socket): SendMsg C_uuid call-command: execute execute-app-name: eavesdrop execute-app-arg: A_uuid or B_uuid, result is the same b) Then, as the spec says ( http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop) I sent DTMF 3 with api uuid_send_dtmf C_uuid 3 but it doesn't work. I mean that A can hear B and vice verse, but both A and B can't hear C. C also doesn't hear neither A nor B. If I press 3 on the C's softphone (latest X-Lite) then, really, C becomes a full-capabilities participator of the call. Instead of uuid_send_dtmf I tried: 1) sendevent DTMF Unique-ID: C_uuid DTMF-Digit: 3 DTMF-Duration: 2000 2) first make queue_dtmf for the C_uuid, and then eavesdrop 3) SendMsg C_uuid call-command: execute execute-app-name: gentones execute-app-arg: 3 4) SendMsg C_uuid call-command: execute execute-app-name: send_dtmf execute-app-arg: 3 And none of these methods leads to the barged in call. Anyone knows how to press 3 programmatically on behalf of the given channel with mod_socket?! Artem ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] I got ERLang to fire a configuration request
I still need to stuff the Freeswitch PID into global storage somewhere
so the process that's handling the configuration requests can send the
reply without crashing (it's just getting a node id, not a Pid), but I
seem to be on my way to configuring Freeswitch via ERLang.
freeswitch_bind.erl has the calls added to register the ERLang
callbacks. The callback function itself is in the aptly named
freeswitch_callback.erl.
-module(freeswitch_bind).
-behaviour(gen_server).
-record(st, {fsnode, pbxpid, configpid, dirpid, dialpid}).
-export([start/3, terminate/2, code_change/3, init/1,
handle_call/3, handle_cast/2, handle_info/2]).
%%
%% gen_server methods
start(Node, Section, Pid) -
gen_server:start(?MODULE, [Node, Section, Pid], []).
init([Node, Section, Pid]) -
io:format( freeswitch_bind:init( [Node=~w, Section=~w, Pid=~w])~n,
[Node, Section, Pid] ),
{api, Node} ! {bind, Section},
receive
ok -
{ok, ConfigurationPid } = freeswitch:start_fetch_handler( Node,
configuration, freeswitch_callback, fetch_handler ),
{ok, DirectoryPid } = freeswitch:start_fetch_handler( Node,
directory, freeswitch_callback, fetch_handler ),
{ok, DialplanPid } = freeswitch:start_fetch_handler( Node,
dialplan, freeswitch_callback, fetch_handler ),
{ok, #st{fsnode=Node, pbxpid=Pid, configpid=ConfigurationPid,
dirpid=DirectoryPid, dialpid=DialplanPid}};
{error, Reason} -
{stop, {error, {freeswitch_error, Reason}}}
after 5000 -
{stop, {error, freeswitch_timeout}}
end.
terminate(_Reason, _State) -
ok.
code_change(_OldVsn, State, _Extra) -
{ok, State}.
%%
%% If the request isn't recognized, just log it and do nothing.
%%
handle_call(Request, _From, State) -
io:format(freeswitch_bind:handle_call( ~w, _From, State) unrecognized
request~n,
[Request]),
{reply, {error, unrecognized_request}, State}.
handle_cast(Message, State) -
error_logger:error_msg(~p received unrecognized cast ~p~n,
[self(), Message]),
{noreply, State}.
handle_info({fetch, Section, Tag, Key, Value, FetchID, Params},
#st{fsnode=Node, pbxpid=Pid}=State) -
{ok, XML} = gen_server:call(Pid, {fetch, Section, Tag, Key, Value, Params}),
{api, Node} ! {fetch_reply, FetchID, XML},
receive
ok -
{noreply, State};
{error, Reason} -
{stop, {error, Reason}, State}
end.
%% Author: mark
%% Created: Sep 15, 2009
%% Description: TODO: Add description to freeswitch_callback
-module(freeswitch_callback).
-behaviour(gen_server).
-record(st, {fsnode, pbxpid}).
-export([start/3, terminate/2, code_change/3, init/1,
handle_call/3, handle_cast/2, handle_info/2, fetch_handler/1]).
start(Node, Section, Pid) -
gen_server:start(?MODULE, [Node, Section, Pid], []).
init([Node, Section, Pid]) -
io:format( freeswitch_callback:init( [Node=~w, Section=~w,
Pid=~w])~n, [Node, Section, Pid] ),
{ok, #st{fsnode=Node, pbxpid=Pid}}.
terminate(_Reason, _State) -
ok.
code_change(_OldVsn, State, _Extra) -
{ok, State}.
%%
%% Callback for freeswitch:start_fetch_handler() called in
freeswitch_bind:init()
%%
fetch_handler( FreeswitchNode ) -
receive
{ nodedown, Node } -
io:format( freeswitch_callback:fetch_handler() Node ~w
is down~n, [Node] ),
ok;
{ fetch, Section, Tag, Key, Value, FetchId, Params } -
io:format( freeswitch_callback:fetch_handler()
Invoking xml_fetch()~n ),
{ok, Xml} = xml_fetch( {fetch, Section, Tag, Key, Value,
Params} ),
io:format( freeswitch_callback:fetch_handler() Sending
reply to FreeswitchNode ~w: ~s~n, [FreeswitchNode, Xml] ),
FreeswitchNode ! { fetch_reply, FetchId, Xml },
io:format( freeswitch_callback:fetch_handler() Reply
sent~n ),
ok
end,
{ ok } = fetch_handler( FreeswitchNode ),
{ ok }.
%%
%% Configuration handler replies that the requested document section, tag,
and key are not
%% found.
%%
xml_fetch({fetch, configuration, Tag, Key, Value, Params}) -
io:format( freeswitch_callback:handle_call( {fetch, configuration,
Tag=~s, Key=~s, Value=~s, Params=~w} )~n,
[Tag, Key, Value, Params]),
Xml =
document type=\freeswitch/xml\
section name=\result\
result status=\not found\ /
/section
/document,
{ok, Xml };
%%
%% Directory handler replies that the requested document section, tag, and
key are not
%% found.
%%
xml_fetch({fetch, directory, Tag, Key, Value, Params}) -
io:format( freeswitch_callback:xml_fetch( {fetch, directory, Tag=~s,
Key=~s, Value=~s,
Re: [Freeswitch-users] CALL FOR VOLUNTEERS: Assisting With FreeSWITCH Subprojects
Hey Demuel, et al, I agree that without coordination, there will be duplication of effort, etc. How does one find the line to get in to help with, in my case, documentation? I don't want to, with the best of intentions, simply create confusion. I'm new, so if I've simply not yet RTFM that answers my question above, some kind person might nudge me in the right direction. Back to registering with my SIP provider... Regards, Mike G. On Tue, Sep 15, 2009 at 2:38 AM, dem...@thephinix.org wrote: Hi Michael / FS enthusiast, In my opinion, it will be an undeniably perfect chaos if everyone wants to do everything. One way I can suggest out of this is the following: - sets of persons that does all the stuff on the FS core. I'm not sure if this one will work out but I think these are the only persons who has commit access to the source code and will also be willing to accept modifications, enhancements, etc. - individuals can ask if they can be the maintainer/tester for a particular module. Bug fixing, sending modifications and enhancements will still be subject to the approval of anthm. - an array of persons that will take ownership on what FS can do and provide realiable working examples for it. Currently, it takes amount of pain for a newbie on how to configure SIP because the examples are too confusing and sometimes there is not certainty if this will work on what release of FS or not. - any individual that has time and interest can start owning the porting of FS to other operating system. Like in my case, I am working on making FS ported to FreeBSD ports but I don't know if somebody did it or if he is currently doing it, as to what stage he is in? - there should be a release engineering team. In my opinion, a stable and current release will be much more sane. As with any other successful open source project, we won't be starting to say I want to contribute this and start working on that. We should indicate that you take ownership of this and any comments should be forwarded right unto you. Again, this is just my opinion. You can take some of it or leave it using sudo rm -rf blah/* . There and back again, Demuel I. Bendano a.k.a engrxyz Hi Michael, You can count with me for anything else, like documentation, coding/scripting, or any other FreeSWITCH related stuff. Regards, Diego 2009/9/14 João Mesquita jmesqu...@freeswitch.org You can assign two things to me. 1. libesl code documentation (partially done and Doxygened - needs cleaning) 2. Bug marshal. I am setting up the proper lab environment here to be able to test most stuff. Count me in for any questions I can answer and I am _always_ on IRC jmesquita On Mon, Sep 14, 2009 at 1:33 PM, Michael Collins m...@freeswitch.org wrote: Hello FreeSWITCHers! We are looking for people who are in a position to help out with various subprojects that will help FreeSWITCH to keep growing. We need people to help out in these basic areas: Bug marshals (people who watch JIRA and test bug reports, patches, etc.) Documentation maintainers (people who update the wiki when new stuff comes out, also those familiar with mediawiki administration) Documentation authors (people who write new docs, how-to's, tutorials, examples, etc.) Package maintainers (people who manage Debian debs, RPMs, etc.) Additionally, we are always looking for more folks to assist with answering questions on IRC and the mailing list. It is definitely nice to have people who've gone through the pains of switching to FreeSWITCH (or learning it from scratch) who can assist the steady stream of new users. If you want to help and aren't sure where to go from here then please at least do the following: #1 - Join #freeswitch on irc.freenode.net and hang out as much as possible #2 - Check the recent changes link on wiki.freeswitch.org each day #3 - Join the Friday public conference call and listen in These three things, in addition to the mailing list, will keep you well in tune with the FreeSWITCH community and what's happening. Next, make a note of the parts of FS that you use frequently, know a lot about, or are particularly passionate about. Those are the items we'd love to have you help us with. For example: if you use mod_xml_curl frequently and have been through the set up process then you're a prime candidate to help answer questions, refine the mod_xml_curl wiki documentation, write up a tutorial, contribute a working example of a web server database schema, etc. If you are good with a scripting language then we could definitely use help with rounding out the docs for your favorite language. We could also use code samples, so ask for a contrib folder if you have things you would like to share. Or how about this: you read something on the wiki, it doesn't quite work when you try, so you tinker until you figure
Re: [Freeswitch-users] CALL FOR VOLUNTEERS: Assisting With FreeSWITCH Subprojects
Answered my own question. Will email Diego off-list. On Tue, Sep 15, 2009 at 9:50 AM, Michael Gende mge...@gendesign.com wrote: Hey Demuel, et al, I agree that without coordination, there will be duplication of effort, etc. How does one find the line to get in to help with, in my case, documentation? I don't want to, with the best of intentions, simply create confusion. I'm new, so if I've simply not yet RTFM that answers my question above, some kind person might nudge me in the right direction. Back to registering with my SIP provider... Regards, Mike G. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] CALL FOR VOLUNTEERS: Assisting With FreeSWITCH Subprojects
The friday meetings are where we all collaborate on these group efforts and discuss project direction, goals and areas where people can help out more. Did we ever find someone to officially take over the Debian packages? /b On Sep 15, 2009, at 9:56 AM, Michael Gende wrote: Answered my own question. Will email Diego off-list. On Tue, Sep 15, 2009 at 9:50 AM, Michael Gende mge...@gendesign.com wrote: Hey Demuel, et al, I agree that without coordination, there will be duplication of effort, etc. How does one find the line to get in to help with, in my case, documentation? I don't want to, with the best of intentions, simply create confusion. I'm new, so if I've simply not yet RTFM that answers my question above, some kind person might nudge me in the right direction. Back to registering with my SIP provider... Regards, Mike G. ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] problem: no audio for one of the person in conference
replace execute_extension with transfer
or use the application three_way
action application=three_way data=any uuid of a bridged call/
On Tue, Sep 15, 2009 at 8:43 AM, Artem Shiyanov shiya...@gmail.com wrote:
Hi there!
The situation is:
- Person A calls to the extension:
extension name=some_ext
condition field=destination_number expression=^900.$
action application=answer/
action application=park/
/condition
/extension
- I bridge him with person B with help of mod_socket:
SendMsg some_uuid
call-command: execute
execute-app-name: bridge
execute-app-arg: person B address in form: user/...
- A and B talks
- Person C decides to barge in the call A--B (to become a third
participator in the call)
a) I send (mod_socket):
api originate user/person C address park()
b) then I move A, B, C to the extension:
extension name=barge_in
action application=conference data=my_confn...@my_profile
+flags{mintwo}/
/extension
conference profile my_profile is:
profile name=my_profile
!-- Sample Rate--
param name=rate value=8000/
!-- Number of milliseconds per frame --
param name=interval value=20/
!-- Energy level required for audio to be sent to the other users --
param name=energy-level value=300/
!-- Name of the caller control group to use for this profile --
param name=caller-controls value=none/
param name=comfort-noise-level value=1400/
!-- enable comfort noise generation --
param name=comfort-noise value=true/
/profile
The moving itself is done by sending this for each (A,B,C) channel
SendMsg uuid
call-command: execute
execute-app-name: execute_extension
execute-app-arg: barge_in
- Result: A, B, C are in the same conference with name my_confname, A can
hear B and vice verse, but both A and B can't hear C. C also doesn't hear
neither A nor B.
I also tried the moving to conference with
api uuid_transfer A_uuid -both barge_in
api uuid_transfer C_uuid barge_in
but result is the same.
Maybe someone already faced with such issue?
Thanks,
Artem
___
FreeSWITCH-users mailing list
FreeSWITCH-users@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/
Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm
MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com
GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org
iax:gu...@conference.freeswitch.org/888
googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org
pstn:213-799-1400
___
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
Re: [Freeswitch-users] barge in implementation with mod_socket and eavesdrop
yes call the app as three_way like i said in the other thread. On Tue, Sep 15, 2009 at 9:22 AM, Artem Shiyanov shiya...@gmail.com wrote: Hello! I'm trying to implement barge in functionality (see http://www.yourdictionary.com/telecom/barge-in) with eavesdrop but still with no success. The situation is: - Person A calls to the extension: extension name=some_ext condition field=destination_number expression=^900.$ action application=answer/ action application=park/ /condition /extension - I bridge him with person B with help of mod_socket: SendMsg some_uuid call-command: execute execute-app-name: bridge execute-app-arg: person B address in form: user/... - A and B talks - Person C decides to barge in the call A--B (to become a third participator in the call) a) I send (mod_socket): SendMsg C_uuid call-command: execute execute-app-name: eavesdrop execute-app-arg: A_uuid or B_uuid, result is the same b) Then, as the spec says ( http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop) I sent DTMF 3 with api uuid_send_dtmf C_uuid 3 but it doesn't work. I mean that A can hear B and vice verse, but both A and B can't hear C. C also doesn't hear neither A nor B. If I press 3 on the C's softphone (latest X-Lite) then, really, C becomes a full-capabilities participator of the call. Instead of uuid_send_dtmf I tried: 1) sendevent DTMF Unique-ID: C_uuid DTMF-Digit: 3 DTMF-Duration: 2000 2) first make queue_dtmf for the C_uuid, and then eavesdrop 3) SendMsg C_uuid call-command: execute execute-app-name: gentones execute-app-arg: 3 4) SendMsg C_uuid call-command: execute execute-app-name: send_dtmf execute-app-arg: 3 And none of these methods leads to the barged in call. Anyone knows how to press 3 programmatically on behalf of the given channel with mod_socket?! Artem ___ FreeSWITCH-users mailing list FreeSWITCH-users@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/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] CALL FOR VOLUNTEERS: Assisting With FreeSWITCH Subprojects
Demuel, Thanks for the input. Yes, we want to avoid chaos. I will work to keep everyone organized. I would welcome a mass of people all trying to do different things because the challenge would simply be to keep them organized. Right now the challenge is in recruiting people to stay with the not-so-glorious aspects of the project, namely documentation, maintenance, and janitorial style subprojects. Please keep the comments and suggestions coming! Thanks, MC On Tue, Sep 15, 2009 at 12:38 AM, dem...@thephinix.org wrote: Hi Michael / FS enthusiast, In my opinion, it will be an undeniably perfect chaos if everyone wants to do everything. One way I can suggest out of this is the following: - sets of persons that does all the stuff on the FS core. I'm not sure if this one will work out but I think these are the only persons who has commit access to the source code and will also be willing to accept modifications, enhancements, etc. - individuals can ask if they can be the maintainer/tester for a particular module. Bug fixing, sending modifications and enhancements will still be subject to the approval of anthm. - an array of persons that will take ownership on what FS can do and provide realiable working examples for it. Currently, it takes amount of pain for a newbie on how to configure SIP because the examples are too confusing and sometimes there is not certainty if this will work on what release of FS or not. - any individual that has time and interest can start owning the porting of FS to other operating system. Like in my case, I am working on making FS ported to FreeBSD ports but I don't know if somebody did it or if he is currently doing it, as to what stage he is in? - there should be a release engineering team. In my opinion, a stable and current release will be much more sane. As with any other successful open source project, we won't be starting to say I want to contribute this and start working on that. We should indicate that you take ownership of this and any comments should be forwarded right unto you. Again, this is just my opinion. You can take some of it or leave it using sudo rm -rf blah/* . There and back again, Demuel I. Bendano a.k.a engrxyz Hi Michael, You can count with me for anything else, like documentation, coding/scripting, or any other FreeSWITCH related stuff. Regards, Diego 2009/9/14 João Mesquita jmesqu...@freeswitch.org You can assign two things to me. 1. libesl code documentation (partially done and Doxygened - needs cleaning) 2. Bug marshal. I am setting up the proper lab environment here to be able to test most stuff. Count me in for any questions I can answer and I am _always_ on IRC jmesquita On Mon, Sep 14, 2009 at 1:33 PM, Michael Collins m...@freeswitch.org wrote: Hello FreeSWITCHers! We are looking for people who are in a position to help out with various subprojects that will help FreeSWITCH to keep growing. We need people to help out in these basic areas: Bug marshals (people who watch JIRA and test bug reports, patches, etc.) Documentation maintainers (people who update the wiki when new stuff comes out, also those familiar with mediawiki administration) Documentation authors (people who write new docs, how-to's, tutorials, examples, etc.) Package maintainers (people who manage Debian debs, RPMs, etc.) Additionally, we are always looking for more folks to assist with answering questions on IRC and the mailing list. It is definitely nice to have people who've gone through the pains of switching to FreeSWITCH (or learning it from scratch) who can assist the steady stream of new users. If you want to help and aren't sure where to go from here then please at least do the following: #1 - Join #freeswitch on irc.freenode.net and hang out as much as possible #2 - Check the recent changes link on wiki.freeswitch.org each day #3 - Join the Friday public conference call and listen in These three things, in addition to the mailing list, will keep you well in tune with the FreeSWITCH community and what's happening. Next, make a note of the parts of FS that you use frequently, know a lot about, or are particularly passionate about. Those are the items we'd love to have you help us with. For example: if you use mod_xml_curl frequently and have been through the set up process then you're a prime candidate to help answer questions, refine the mod_xml_curl wiki documentation, write up a tutorial, contribute a working example of a web server database schema, etc. If you are good with a scripting language then we could definitely use help with rounding out the docs for your favorite language. We could also use code samples, so ask for a contrib folder if you have things you would like to share. Or how about this: you read something on the wiki, it doesn't quite work when you try, so you
Re: [Freeswitch-users] CALL FOR VOLUNTEERS: Assisting With FreeSWITCH Subprojects
ok so the first job we should fill is someone to manage what the other volunteers do. like community czar We can make that person or persons a manager on jira and create projects for the various duties. Everyone can get an account if they don't already have one and take advantage of some of the workflow features in jira. On Tue, Sep 15, 2009 at 9:58 AM, Brian West br...@freeswitch.org wrote: The friday meetings are where we all collaborate on these group efforts and discuss project direction, goals and areas where people can help out more. Did we ever find someone to officially take over the Debian packages? /b On Sep 15, 2009, at 9:56 AM, Michael Gende wrote: Answered my own question. Will email Diego off-list. On Tue, Sep 15, 2009 at 9:50 AM, Michael Gende mge...@gendesign.comwrote: Hey Demuel, et al, I agree that without coordination, there will be duplication of effort, etc. How does one find the line to get in to help with, in my case, documentation? I don't want to, with the best of intentions, simply create confusion. I'm new, so if I've simply not yet RTFM that answers my question above, some kind person might nudge me in the right direction. Back to registering with my SIP provider... Regards, Mike G. ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@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/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] fax detection
Hi, is there any way to route fax calls according to the call capability? I mean .. if the fax call supports T.38 i'd like to route it to a T.38 capable gateway. All other fax calls (meaning inband) should be handled by FS/SpanDSP. Of course, I know that every fax call starts as a voice call and upon fax tone detection additional capabilities are being negotiated(T.38 or G711). Can it be done in early media, before the call is even answered? So, here the goal is to have a T.38 capable GW handling T.38 calls while SpanDSP handling T.30... Any chance to do that with FS? ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] FS Presence Implementation
I would like to modify my SIP phone and my gateway to convey/exchange presence information. Could someone point me toward the FS presence documentation? I've seen bits and pieces. Also, I think presence can be communicated via more than one protocol. Thanks And Best Regards, Jerry ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FS Presence Implementation
I think you better start with the libjingle module for this one. I would like to modify my SIP phone and my gateway to convey/exchange presence information. Could someone point me toward the FS presence documentation? I've seen bits and pieces. Also, I think presence can be communicated via more than one protocol. Thanks And Best Regards, Jerry ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@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] sending custom events from event_socket, channel_variable_changed event?
Hello *,
while trying to figure out how to send custom events from mod_socket
with sendmsg (like a telnet connection or something) i only found
how to do that from within javascript (with e.fire() and stuff).
So... first of all, how's the correct syntax to do that from the
event_socket? the wiki states
---
sendmsg uuid
Send a message to the call of given uuid (call-command execute or
hangup), see examples
below.
---
Is there a secret syntax i missed how to send CUSTOM events?
I tried to work-around that by setting a channel-variable and then
calling the info-app. That way, a channel_execute event is fired and
shows up in my event_socket-app.
If there's a prettier way... please ignore the following up to the
snipp mark :)
so it looked to me that i can only send command messages and looking
at the sources i found that mod_socket will hard-wire the event-name
to SWITCH_EVENT_COMMAND (in read_packet() in
./mod/event_handlers/mod_event_socket/mod_event_socket.c). If i
understand correctly how e.g.. the mod_spidermonkey-bindings enqueue
events, the event-name is taken from the user and can therefor be
CUSTOM (looks like it's being looked up in the
switch_event_types_t-enum).
---
if (switch_name_event(ename, etype) !=
SWITCH_STATUS_SUCCESS) {
switch_log_printf(SWITCH_CHANNEL_LOG,
SWITCH_LOG_WARNING, Unknown event %s\n, ename);
*rval = BOOLEAN_TO_JSVAL(JS_FALSE);
return JS_TRUE;
}
if (etype == SWITCH_EVENT_CUSTOM) {
[...]
if (switch_event_create_subclass(event, etype, subclass_name) !=
SWITCH_STATUS_SUCCESS) {
---
If it's currently not possible, do you think this could be useful, too?
I thought a bit about the syntax and came up with this idea to not
break compatibility, but feel free to change that, you know it better
:)
if you want to send an event that's not COMMAND, use event-name:
CUSTOM as the first line after the sendmsg [uuid] and change the
event-name (if that's possible) of the event already created by
read_packet() or reconstruct the event or ...
snipp
Just another idea. that could become handy is if there would be an
event CHANNEL_VAR_CHANGED, fired/enqueued by setvar(),
setvar_multi() and friends. But as i write this i admit that one will
be overwhelmed by the setvar massacre for example bridge() would do to
set disposition-variables... Just a thought.
Cheers
Beni.
___
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
Re: [Freeswitch-users] ERLang configuration callbacks
On Mon, Sep 14, 2009 at 01:54:31PM -0600, Mark Sobkow wrote: I seem to be missing something in implementing the ERLang callbacks for Freeswitch. Our Freeswitch server is starting and getting registered with ERLang, we're invoking the bind for configuration, but I'm not seeing any of my callbacks fire. What am I missing? The most obvious thing is that you're trying to catch info messages using handle_call. The erlang module doesn't use the OTP protocol for messages so handle_call/cast won't ever fire for messages sent from the freeswitch module. Andrew ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FS Presence Implementation
Also, is presence conveyed as any string? Or is presence a predefined list of status? Best Regards, Jerry -Original Message- From: Jerry Richards [mailto:jerry.richa...@teotech.com] Sent: Tuesday, September 15, 2009 8:46 AM To: 'freeswitch-users@lists.freeswitch.org' Subject: FS Presence Implementation I would like to modify my SIP phone and my gateway to convey/exchange presence information. Could someone point me toward the FS presence documentation? I've seen bits and pieces. Also, I think presence can be communicated via more than one protocol. Thanks And Best Regards, Jerry ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] ERLang configuration callbacks
On Tue, Sep 15, 2009 at 02:29:14PM -0400, Andrew Thompson wrote: On Mon, Sep 14, 2009 at 01:54:31PM -0600, Mark Sobkow wrote: Oh wait, I see what you're doing, you catch all the fetch requests in the handle_info and then make a call to another process to get the XML. My question is why are those both in the same process? The handle_info part is fine, but then the pid you make a gen_server:call to *must* be a different process or you'll hit a timeout and the process will exit. However, I can't start 2 copies of that process (one to catch requests, one to return XML) because your module always does a bind! Regardless, I at least get fetch requests when I run your module, I can't make it return something without refactoring it, but it does receive the requests at least. Andrew ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FS Presence Implementation
the default config ships with presence enabled for SIP if you have a phone that supports it, all you have to do is enable it on the phone. On Tue, Sep 15, 2009 at 1:30 PM, Jerry Richards jerry.richa...@teotech.comwrote: Also, is presence conveyed as any string? Or is presence a predefined list of status? Best Regards, Jerry -Original Message- From: Jerry Richards [mailto:jerry.richa...@teotech.com] Sent: Tuesday, September 15, 2009 8:46 AM To: 'freeswitch-users@lists.freeswitch.org' Subject: FS Presence Implementation I would like to modify my SIP phone and my gateway to convey/exchange presence information. Could someone point me toward the FS presence documentation? I've seen bits and pieces. Also, I think presence can be communicated via more than one protocol. Thanks And Best Regards, Jerry ___ FreeSWITCH-users mailing list FreeSWITCH-users@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/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] sending custom events from event_socket, channel_variable_changed event?
I think we can't do it just because nobody needed it.
I added a patch to r14874 to allow you to add a unique-id header to the
sendevent command
that should allow you to address and event right to a particular session
rather than fire the event.
On Tue, Sep 15, 2009 at 12:08 PM, Benedikt Fraunhofer
fraunhofer.lists.freeswitch-...@traced.net wrote:
Hello *,
while trying to figure out how to send custom events from mod_socket
with sendmsg (like a telnet connection or something) i only found
how to do that from within javascript (with e.fire() and stuff).
So... first of all, how's the correct syntax to do that from the
event_socket? the wiki states
---
sendmsg uuid
Send a message to the call of given uuid (call-command execute or
hangup), see examples
below.
---
Is there a secret syntax i missed how to send CUSTOM events?
I tried to work-around that by setting a channel-variable and then
calling the info-app. That way, a channel_execute event is fired and
shows up in my event_socket-app.
If there's a prettier way... please ignore the following up to the
snipp mark :)
so it looked to me that i can only send command messages and looking
at the sources i found that mod_socket will hard-wire the event-name
to SWITCH_EVENT_COMMAND (in read_packet() in
./mod/event_handlers/mod_event_socket/mod_event_socket.c). If i
understand correctly how e.g.. the mod_spidermonkey-bindings enqueue
events, the event-name is taken from the user and can therefor be
CUSTOM (looks like it's being looked up in the
switch_event_types_t-enum).
---
if (switch_name_event(ename, etype) !=
SWITCH_STATUS_SUCCESS) {
switch_log_printf(SWITCH_CHANNEL_LOG,
SWITCH_LOG_WARNING, Unknown event %s\n, ename);
*rval = BOOLEAN_TO_JSVAL(JS_FALSE);
return JS_TRUE;
}
if (etype == SWITCH_EVENT_CUSTOM) {
[...]
if (switch_event_create_subclass(event, etype, subclass_name) !=
SWITCH_STATUS_SUCCESS) {
---
If it's currently not possible, do you think this could be useful, too?
I thought a bit about the syntax and came up with this idea to not
break compatibility, but feel free to change that, you know it better
:)
if you want to send an event that's not COMMAND, use event-name:
CUSTOM as the first line after the sendmsg [uuid] and change the
event-name (if that's possible) of the event already created by
read_packet() or reconstruct the event or ...
snipp
Just another idea. that could become handy is if there would be an
event CHANNEL_VAR_CHANGED, fired/enqueued by setvar(),
setvar_multi() and friends. But as i write this i admit that one will
be overwhelmed by the setvar massacre for example bridge() would do to
set disposition-variables... Just a thought.
Cheers
Beni.
___
FreeSWITCH-users mailing list
FreeSWITCH-users@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/
Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm
MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com
GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org
iax:gu...@conference.freeswitch.org/888
googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org
pstn:213-799-1400
___
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
Re: [Freeswitch-users] I got ERLang to fire a configuration request
Mark,
You might want to send this question to
freeswitch-...@lists.freeswitch.orgas it's a bit intense for the users
list. :)
-MC
On Tue, Sep 15, 2009 at 7:38 AM, Mark Sobkow
m.sob...@marketelsystems.comwrote:
I still need to stuff the Freeswitch PID into global storage somewhere so
the process that's handling the configuration requests can send the reply
without crashing (it's just getting a node id, not a Pid), but I seem to be
on my way to configuring Freeswitch via ERLang.
freeswitch_bind.erl has the calls added to register the ERLang callbacks.
The callback function itself is in the aptly named freeswitch_callback.erl.
-module(freeswitch_bind).
-behaviour(gen_server).
-record(st, {fsnode, pbxpid, configpid, dirpid, dialpid}).
-export([start/3, terminate/2, code_change/3, init/1,
handle_call/3, handle_cast/2, handle_info/2]).
%%
%% gen_server methods
start(Node, Section, Pid) -
gen_server:start(?MODULE, [Node, Section, Pid], []).
init([Node, Section, Pid]) -
io:format( freeswitch_bind:init( [Node=~w, Section=~w, Pid=~w])~n,
[Node, Section, Pid] ),
{api, Node} ! {bind, Section},
receive
ok -
{ok, ConfigurationPid } = freeswitch:start_fetch_handler(
Node, configuration, freeswitch_callback, fetch_handler ),
{ok, DirectoryPid } = freeswitch:start_fetch_handler( Node,
directory, freeswitch_callback, fetch_handler ),
{ok, DialplanPid } = freeswitch:start_fetch_handler( Node,
dialplan, freeswitch_callback, fetch_handler ),
{ok, #st{fsnode=Node, pbxpid=Pid, configpid=ConfigurationPid,
dirpid=DirectoryPid, dialpid=DialplanPid}};
{error, Reason} -
{stop, {error, {freeswitch_error, Reason}}}
after 5000 -
{stop, {error, freeswitch_timeout}}
end.
terminate(_Reason, _State) -
ok.
code_change(_OldVsn, State, _Extra) -
{ok, State}.
%%
%% If the request isn't recognized, just log it and do nothing.
%%
handle_call(Request, _From, State) -
io:format(freeswitch_bind:handle_call( ~w, _From, State) unrecognized
request~n,
[Request]),
{reply, {error, unrecognized_request}, State}.
handle_cast(Message, State) -
error_logger:error_msg(~p received unrecognized cast ~p~n,
[self(), Message]),
{noreply, State}.
handle_info({fetch, Section, Tag, Key, Value, FetchID, Params},
#st{fsnode=Node, pbxpid=Pid}=State) -
{ok, XML} = gen_server:call(Pid, {fetch, Section, Tag, Key, Value,
Params}),
{api, Node} ! {fetch_reply, FetchID, XML},
receive
ok -
{noreply, State};
{error, Reason} -
{stop, {error, Reason}, State}
end.
%% Author: mark
%% Created: Sep 15, 2009
%% Description: TODO: Add description to freeswitch_callback
-module(freeswitch_callback).
-behaviour(gen_server).
-record(st, {fsnode, pbxpid}).
-export([start/3, terminate/2, code_change/3, init/1,
handle_call/3, handle_cast/2, handle_info/2, fetch_handler/1]).
start(Node, Section, Pid) -
gen_server:start(?MODULE, [Node, Section, Pid], []).
init([Node, Section, Pid]) -
io:format( freeswitch_callback:init( [Node=~w, Section=~w,
Pid=~w])~n, [Node, Section, Pid] ),
{ok, #st{fsnode=Node, pbxpid=Pid}}.
terminate(_Reason, _State) -
ok.
code_change(_OldVsn, State, _Extra) -
{ok, State}.
%%
%% Callback for freeswitch:start_fetch_handler() called in
freeswitch_bind:init()
%%
fetch_handler( FreeswitchNode ) -
receive
{ nodedown, Node } -
io:format( freeswitch_callback:fetch_handler() Node
~w is down~n, [Node] ),
ok;
{ fetch, Section, Tag, Key, Value, FetchId, Params } -
io:format( freeswitch_callback:fetch_handler()
Invoking xml_fetch()~n ),
{ok, Xml} = xml_fetch( {fetch, Section, Tag, Key, Value,
Params} ),
io:format( freeswitch_callback:fetch_handler()
Sending reply to FreeswitchNode ~w: ~s~n, [FreeswitchNode, Xml] ),
FreeswitchNode ! { fetch_reply, FetchId, Xml },
io:format( freeswitch_callback:fetch_handler()
Reply sent~n ),
ok
end,
{ ok } = fetch_handler( FreeswitchNode ),
{ ok }.
%%
%% Configuration handler replies that the requested document section,
tag, and key are not
%% found.
%%
xml_fetch({fetch, configuration, Tag, Key, Value, Params}) -
io:format( freeswitch_callback:handle_call( {fetch, configuration,
Tag=~s, Key=~s, Value=~s, Params=~w} )~n,
[Tag, Key, Value, Params]),
Xml =
document type=\freeswitch/xml\
section name=\result\
result status=\not found\ /
/section
/document,
{ok, Xml };
%%
%%
Re: [Freeswitch-users] I got ERLang to fire a configuration request
On Tue, Sep 15, 2009 at 08:38:41AM -0600, Mark Sobkow wrote:
I still need to stuff the Freeswitch PID into global storage somewhere
so the process that's handling the configuration requests can send the
reply without crashing (it's just getting a node id, not a Pid), but I
seem to be on my way to configuring Freeswitch via ERLang.
Looking at your code, assuming I read it right, you should be able to
just replace:
FreeSWITCHNode ! SomeMsg.
with
{api, FreeSWITCHNode} ! SomeMsg.
The freeswitch module doesn't really have a pid, since it's not a real
erlang node, it's all faked and all messages to a pid or a registered
process on the C node go to the same place.
Andrew
___
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
Re: [Freeswitch-users] A FreeSWITCH Calling Card Application written in Ruby
Hello, Someone could tell me what happens to the project, it seems that is no longer available in github ? http://github.com/diego/freeswitch-card/ thanks, On Sun, Sep 6, 2009 at 10:57 PM, Diego Viola diego.vi...@gmail.com wrote: I also have plans to add a GUI later, maybe I will merge my code and turn it into a ramaze app, but it should be usable right now. Regards, Diego On Mon, Sep 7, 2009 at 1:36 AM, Diego Viola diego.vi...@gmail.com wrote: Hello, I'm currently working on a calling card application written in Ruby, just a hobby, I currently have it on a usable state and I thought I would post it here in case if there is someone interested. It uses mod_nibblebill as the billing/rate engine and FSR (FreeSWITCHeR) as the event socket library. Here is the url of the project: http://github.com/diego/freeswitch-card/tree/master You can find a clone of the project on my FreeSWITCH contrib directory too. http://fisheye.freeswitch.org/browse/FreeSWITCH/contrib/diegoviola/ Regards, Diego ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] sending custom events from event_socket, channel_variable_changed event?
Hi Anthony, 2009/9/15 Anthony Minessale anthony.miness...@gmail.com: I added a patch to r14874 to allow you to add a unique-id header to the sendevent command that should allow you to address and event right to a particular session rather than fire the event. thx! While riding the train back from work i thought i could've missed that what i wanted to achieve was possible with sendevent instead of sendmsg but your reply made me calm down :) I won't be able to give it a try until this Friday, but I'll report back if this is the new way to send CUSTOM events from mod_event to mod_event and pay the wiki tax, accordingly :) Thx again! Beni. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Sip Allow Options
Because you have 100rel disabled so PRACK will NOT show up in the allow list. /b On Sep 15, 2009, at 3:38 PM, DJB wrote: I wonder whether anyone can tell me why the latest trunk has no PRACK comparing to the 1.0.4. Here is the sip message: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14877 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Compare to: User-Agent: FreeSWITCH-mod_sofia/1.0.4-exported Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Thank you, Dorn B. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_radius_cdr: Multiple RADIUS messages for single call?
Thanks to those for the info and help on this issue. Ultimately ended up having to use alternative software for the radius piece (not related to any shortfalls by Freeswitch). Vlad From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Friday, September 11, 2009 11:31 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_radius_cdr: Multiple RADIUS messages for single call? set the variable process_cdr=false on that a_leg first thing in your dialplan On Fri, Sep 11, 2009 at 2:26 AM, Anatoliy Kounitskiy anato...@kounitskiy.com wrote: It's normal to have to two records for a call - Start and Stop message. From what i see - you have one start and stop for each leg of the call. Regards, AK email lists wrote: Hello, Using the mod_radius_cdr module w/ Freeswitch, I am seeing duplicate RADIUS messages being generated for individual calls (sample messages for one call below). Looking at the Acct-Unique-Session-Id and Acct-Session-Id fields, it would appear that perhaps each call leg results in a pair of start/stop RADIUS messages; is this the expected behavior? If so, is there a way to disable RADIUS messaging for what I presume is the ingress or A leg of the call? Any leads would be appreciated. Thanks in advance. Vladimir Thu Sep 10 10:37:25 2009 Acct-Status-Type = Start Acct-Session-Id = b0f387c0-35bd-e32c-971a-d79d026a8004 User-Name = 8135793256 Freeswitch-Src = 8135793256 Freeswitch-CLID = sipp Freeswitch-Dst = 14043297226 Freeswitch-Dialplan = XML Framed-IP-Address = 50.46.50.55 Freeswitch-Context = public Freeswitch-Source = mod_sofia Freeswitch-Callstartdate = 2009-09-10T10:22:00.259042-0700 NAS-Port = 0 Acct-Delay-Time = 0 NAS-IP-Address = 1.1.1.1 Acct-Unique-Session-Id = 097c8472ff7bcec7 Timestamp = 1252604245 Request-Authenticator = Verified Thu Sep 10 10:37:25 2009 Acct-Status-Type = Start Acct-Session-Id = 584f4573-46f7-655f-91d7-84cd59c9ec12 User-Name = 8135793256 Freeswitch-Src = 8135793256 Freeswitch-CLID = sipp Freeswitch-Dst = 14043297...@x.x.x.x Freeswitch-Dialplan = XML Framed-IP-Address = 50.46.50.55 Freeswitch-Context = public Freeswitch-Source = mod_sofia Freeswitch-Callstartdate = 2009-09-10T10:22:00.279044-0700 NAS-Port = 0 Acct-Delay-Time = 0 NAS-IP-Address = 1.1.1.1 Acct-Unique-Session-Id = 53f729e173e8c0a9 Timestamp = 1252604245 Request-Authenticator = Verified Thu Sep 10 10:37:57 2009 Acct-Status-Type = Stop Acct-Session-Id = b0f387c0-35bd-e32c-971a-d79d026a8004 Freeswitch-Hangupcause = Normal-Unspecified User-Name = 8135793256 Freeswitch-Src = 8135793256 Freeswitch-CLID = sipp Freeswitch-Dst = 14043297226 Freeswitch-Dialplan = XML Framed-IP-Address = 50.46.50.55 Freeswitch-Context = public Freeswitch-Source = mod_sofia Freeswitch-Lastapp = bridge Freeswitch-Billusec = 32029926 Freeswitch-Callstartdate = 2009-09-10T10:22:00.259042-0700 Freeswitch-Callanswerdate = 2009-09-10T10:22:00.319197-0700 Freeswitch-Callenddate = 2009-09-10T10:22:32.349123-0700 Acct-Session-Time = 32 NAS-Port = 0 Acct-Delay-Time = 0 NAS-IP-Address = 1.1.1.1 Acct-Unique-Session-Id = 097c8472ff7bcec7 Timestamp = 1252604277 Request-Authenticator = Verified Thu Sep 10 10:38:02 2009 Acct-Status-Type = Stop Acct-Session-Id = 584f4573-46f7-655f-91d7-84cd59c9ec12 Freeswitch-Hangupcause = Normal-Clearing User-Name = 8135793256 Freeswitch-Src = 8135793256 Freeswitch-CLID = sipp Freeswitch-Dst = 14043297...@x.x.x.x Freeswitch-Dialplan = XML Framed-IP-Address = 50.46.50.55 Freeswitch-Context = public Freeswitch-Source = mod_sofia Freeswitch-Billusec = 32049973 Freeswitch-Callstartdate = 2009-09-10T10:22:00.279044-0700 Freeswitch-Callanswerdate = 2009-09-10T10:22:00.289136-0700 Freeswitch-Callenddate = 2009-09-10T10:22:32.339109-0700 Acct-Session-Time = 32 NAS-Port = 0 Acct-Delay-Time = 0 NAS-IP-Address = 1.1.1.1 Acct-Unique-Session-Id = 53f729e173e8c0a9 Timestamp = 1252604282 Request-Authenticator = Verified** -- -- ___ FreeSWITCH-users mailing list
Re: [Freeswitch-users] stange segfaults with log
On Tue, Sep 15, 2009 at 3:29 AM, Tamas Cseke cstomi.levl...@gmail.comwrote: Hello we have a strange problem with 14144 revision. It seems switch_log_printf got NULL pointer as data. It happens a few times. however in the previous frame session seems to be good for us. http://pastebin.freeswitch.org/10357 Could you please tell me what is the problem? Did we make some mistakes with building? Missed make clean, or someting? Well, for one thing you are more than 800 revs behind current SVN. I strongly recommend you make current and let the system get properly updated. -MC Thanks in advance, Tamas ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@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] 502 Bad Gateway: Destination out of order error
Hello All, Wondering if anyone has experienced this issue before. I've attached a snip of the log file where the error occurs and could use some leads on this. What's interesting is that the call appears to complete as normal, and a radius stop message even gets generated, though the duration is ~1 second. snip h323-disconnect-time = h323-disconnect-time=14:35:01.000 UTC Tue Sep 15 2009 h323-connect-time = h323-connect-time=14:34:59.000 UTC Tue Sep 15 2009 /snip While there are a lot of pieces involved, the call scenario is pretty basic (no transfers, no holds, etc.), just a few redirects that Freeswitch appears to be able to handle without issue. Attached is a dumb'd down call ladder. I tried different rates at which I generate the calls, but it didn't seem to correlate to the amount of errors I am seeing. Sending a total of 100 calls, with a call duration of 10 seconds: @10 calls per second = 14 502 errors. @5 calls per second = 4 502 errors. @4 calls per second = NO ERRORS (1st run) @4 calls per second = 39 502 errors. Please let me know if any additional information is needed. Thanks in advance for all help. Vladimir fs.log Description: Binary data attachment: ladder.png___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] 502 Bad Gateway: Destination out of order error
It's probably from this 2009-09-15 14:34:55.838365 [ERR] sofia_glue.c:2503 AUDIO RTP REPORTS ERROR: [Socket Error!] 2009-09-15 14:34:55.838365 [NOTICE] sofia_glue.c:2504 Hangup sofia/external/ 4...@192.168.0.150 [CS_CONSUME_MEDIA][DESTINATION_OUT_OF_ORDER ] 2009-09-15 14:34:55.838365 [ERR] sofia.c:3796 RTP Error! your machine failed to produce a socket when requested. My blind guess, you are on a 32 bit machine and you do not have the ulimits set for enough file descriptors etc.. ulimit -c unlimited ulimit -d unlimited ulimit -f unlimited ulimit -i unlimited ulimit -n 99 ulimit -q unlimited ulimit -u unlimited ulimit -v unlimited ulimit -x unlimited ulimit -s 244 ulimit -l unlimited On Tue, Sep 15, 2009 at 5:22 PM, email lists email.list.subscri...@gmail.com wrote: Hello All, Wondering if anyone has experienced this issue before. I've attached a snip of the log file where the error occurs and could use some leads on this. What's interesting is that the call appears to complete as normal, and a radius stop message even gets generated, though the duration is ~1 second. snip h323-disconnect-time = h323-disconnect-time=14:35:01.000 UTC Tue Sep 15 2009 h323-connect-time = h323-connect-time=14:34:59.000 UTC Tue Sep 15 2009 /snip While there are a lot of pieces involved, the call scenario is pretty basic (no transfers, no holds, etc.), just a few redirects that Freeswitch appears to be able to handle without issue. Attached is a dumb'd down call ladder. I tried different rates at which I generate the calls, but it didn’t seem to correlate to the amount of errors I am seeing. Sending a total of 100 calls, with a call duration of 10 seconds: @10 calls per second = 14 502 errors. @5 calls per second = 4 502 errors. @4 calls per second = NO ERRORS (1st run) @4 calls per second = 39 502 errors. Please let me know if any additional information is needed. Thanks in advance for all help. Vladimir ___ FreeSWITCH-users mailing list FreeSWITCH-users@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/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF CSeq: 505 INFO vs RTP DTMF
After digging into this issue, it might the case that the implementation of out-bound DTMF of the client i am using does not properly increments CSeq per DTMF. For those interested, i am currently integrating OpenBTS with Freeswitch! :) -aep -- Stopping junk mailers is good for the environment Hi, I am using the function session.collectInput and session.streamFile to collect a number of DTMF digits. If the DTMF digits are sent in the RTP, i can collect several digits until timeout. No problem there! If the DTMFs are received as a sequence of SIP INFO packages, collectInput only receives the first one. Any ideas? -- Stopping junk mailers is good for the environment ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] CALL FOR VOLUNTEERS: Assisting With FreeSWITCH Subprojects
Hi MC, Months ago we had tried the multi-language plugin on MediaWiki, I know you are still planning to do this but I just want how far it goes. Count me in when you are short of hand. On Sep 15, 2009, at 11:25 PM, Michael Collins wrote: Demuel, Thanks for the input. Yes, we want to avoid chaos. I will work to keep everyone organized. I would welcome a mass of people all trying to do different things because the challenge would simply be to keep them organized. Right now the challenge is in recruiting people to stay with the not-so-glorious aspects of the project, namely documentation, maintenance, and janitorial style subprojects. Please keep the comments and suggestions coming! Thanks, MC On Tue, Sep 15, 2009 at 12:38 AM, dem...@thephinix.org wrote: Hi Michael / FS enthusiast, In my opinion, it will be an undeniably perfect chaos if everyone wants to do everything. One way I can suggest out of this is the following: - sets of persons that does all the stuff on the FS core. I'm not sure if this one will work out but I think these are the only persons who has commit access to the source code and will also be willing to accept modifications, enhancements, etc. - individuals can ask if they can be the maintainer/tester for a particular module. Bug fixing, sending modifications and enhancements will still be subject to the approval of anthm. - an array of persons that will take ownership on what FS can do and provide realiable working examples for it. Currently, it takes amount of pain for a newbie on how to configure SIP because the examples are too confusing and sometimes there is not certainty if this will work on what release of FS or not. - any individual that has time and interest can start owning the porting of FS to other operating system. Like in my case, I am working on making FS ported to FreeBSD ports but I don't know if somebody did it or if he is currently doing it, as to what stage he is in? - there should be a release engineering team. In my opinion, a stable and current release will be much more sane. As with any other successful open source project, we won't be starting to say I want to contribute this and start working on that. We should indicate that you take ownership of this and any comments should be forwarded right unto you. Again, this is just my opinion. You can take some of it or leave it using sudo rm -rf blah/* . There and back again, Demuel I. Bendano a.k.a engrxyz Hi Michael, You can count with me for anything else, like documentation, coding/scripting, or any other FreeSWITCH related stuff. Regards, Diego 2009/9/14 João Mesquita jmesqu...@freeswitch.org You can assign two things to me. 1. libesl code documentation (partially done and Doxygened - needs cleaning) 2. Bug marshal. I am setting up the proper lab environment here to be able to test most stuff. Count me in for any questions I can answer and I am _always_ on IRC jmesquita On Mon, Sep 14, 2009 at 1:33 PM, Michael Collins m...@freeswitch.org wrote: Hello FreeSWITCHers! We are looking for people who are in a position to help out with various subprojects that will help FreeSWITCH to keep growing. We need people to help out in these basic areas: Bug marshals (people who watch JIRA and test bug reports, patches, etc.) Documentation maintainers (people who update the wiki when new stuff comes out, also those familiar with mediawiki administration) Documentation authors (people who write new docs, how-to's, tutorials, examples, etc.) Package maintainers (people who manage Debian debs, RPMs, etc.) Additionally, we are always looking for more folks to assist with answering questions on IRC and the mailing list. It is definitely nice to have people who've gone through the pains of switching to FreeSWITCH (or learning it from scratch) who can assist the steady stream of new users. If you want to help and aren't sure where to go from here then please at least do the following: #1 - Join #freeswitch on irc.freenode.net and hang out as much as possible #2 - Check the recent changes link on wiki.freeswitch.org each day #3 - Join the Friday public conference call and listen in These three things, in addition to the mailing list, will keep you well in tune with the FreeSWITCH community and what's happening. Next, make a note of the parts of FS that you use frequently, know a lot about, or are particularly passionate about. Those are the items we'd love to have you help us with. For example: if you use mod_xml_curl frequently and have been through the set up process then you're a prime candidate to help answer questions, refine the mod_xml_curl wiki documentation, write up a tutorial, contribute a working example of a web server database schema, etc. If you are
Re: [Freeswitch-users] CALL FOR VOLUNTEERS: Assisting With FreeSWITCH Subprojects
On Tue, Sep 15, 2009 at 4:37 PM, Seven Du dujinf...@gmail.com wrote: Hi MC, Months ago we had tried the multi-language plugin on MediaWiki, I know you are still planning to do this but I just want how far it goes. Count me in when you are short of hand. :) Thanks, we are still tinkering. I haven't found anyone who is familiar with the multilang extension of mediawiki, so I'm learning it myself. I'll keep you updated on my progress. -MC ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF CSeq: 505 INFO vs RTP DTMF
On Tue, Sep 15, 2009 at 3:37 PM, Alberto Escudero aep.li...@it46.se wrote: After digging into this issue, it might the case that the implementation of out-bound DTMF of the client i am using does not properly increments CSeq per DTMF. For those interested, i am currently integrating OpenBTS with Freeswitch! :) -aep We are very interested in seeing how this pans out. Please keep us posted on your progress and definitely come back when you have questions. -MC ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FS 1.0.4 erl configure error
hi folks, anyone encountered this problem? tks. /nandy On Mon, Sep 14, 2009 at 2:20 PM, Nandy Dagondon nandy1...@gmail.com wrote: meftah, i disabled mod_erlang_event in modules.conf. unixodbc is installed already. still ... the same error message. tks for your input. /nandy On Sun, Sep 13, 2009 at 11:56 PM, Meftah Tayeb tayeb.mef...@gmail.comwrote: hello, i think you enabled mod_erlang_event in the modules.conf install unixodbc if is not installed thanks Nandy Dagondon a écrit : hi, i want to enable odbc support which is required in mod_lcr feature. however, i encounter ./configure problem after installing Erlang R13B01. this is the portion of the error messages: ... checking for erl... /usr/local/bin/erl checking erlang version... 5.7.2 checking erlang libdir... /usr/local/lib/erlang/lib/erl_interface-3.6.2/lib checking erlang incdir... /usr/local/lib/erlang/lib/erl_interface-3.6.2/include checking ei.h usability... yes checking ei.h presence... no configure: WARNING: ei.h: accepted by the compiler, rejected by the preprocessor! configure: WARNING: ei.h: proceeding with the compiler's result checking for ei.h... yes checking for ei_encode_version in -lei... yes checking for ei_link_unlink in -lei... no configure: Your erlang seems OK, do not forget to enable mod_erlang_event in modules.conf configure: creating ./config.status config.status: creating src/include/switch_version.h.in .infig.status: error: cannot find input file: Makefile END i set ERL_TOP environment variable to the source directory. has anyone encountered this problem? can anyone give me a hint what's wrong. i'm compiling FS 1.0.4. thank you, /nandy -- ___ FreeSWITCH-users mailing listfreeswitch-us...@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org __ Information from ESET NOD32 Antivirus, version of virus signature database 4421 (20090913) __ The message was checked by ESET NOD32 Antivirus. http://www.eset.com __ Information from ESET NOD32 Antivirus, version of virus signature database 4421 (20090913) __ The message was checked by ESET NOD32 Antivirus. http://www.eset.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FS 1.0.4 erl configure error
On Wed, Sep 16, 2009 at 10:04:14AM +0800, Nandy Dagondon wrote: hi folks, anyone encountered this problem? tks. I don't think this has anything to do with erlang or the freeswitch erlang module, it's simply that that module's config checks are run shortly before the real failure occurs. Andrew ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FS 1.0.4 erl configure error
the ./configure script aborts after the last error message. any hint where to look for the problem? tks. /nandy On Wed, Sep 16, 2009 at 10:53 AM, Andrew Thompson and...@hijacked.uswrote: On Wed, Sep 16, 2009 at 10:04:14AM +0800, Nandy Dagondon wrote: hi folks, anyone encountered this problem? tks. I don't think this has anything to do with erlang or the freeswitch erlang module, it's simply that that module's config checks are run shortly before the real failure occurs. Andrew ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FS 1.0.4 erl configure error
is the Erlang source needed in the FS source directory? /nandy On Wed, Sep 16, 2009 at 11:21 AM, Nandy Dagondon nandy1...@gmail.comwrote: the ./configure script aborts after the last error message. any hint where to look for the problem? tks. /nandy On Wed, Sep 16, 2009 at 10:53 AM, Andrew Thompson and...@hijacked.uswrote: On Wed, Sep 16, 2009 at 10:04:14AM +0800, Nandy Dagondon wrote: hi folks, anyone encountered this problem? tks. I don't think this has anything to do with erlang or the freeswitch erlang module, it's simply that that module's config checks are run shortly before the real failure occurs. Andrew ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FS 1.0.4 erl configure error
it's working now. the problem? it's the configure script itself. some ^M characters somehow crept into the line containing ac_config_files. tks for the tip Andrew! /nandy On Wed, Sep 16, 2009 at 11:26 AM, Nandy Dagondon nandy1...@gmail.comwrote: is the Erlang source needed in the FS source directory? /nandy On Wed, Sep 16, 2009 at 11:21 AM, Nandy Dagondon nandy1...@gmail.comwrote: the ./configure script aborts after the last error message. any hint where to look for the problem? tks. /nandy On Wed, Sep 16, 2009 at 10:53 AM, Andrew Thompson and...@hijacked.uswrote: On Wed, Sep 16, 2009 at 10:04:14AM +0800, Nandy Dagondon wrote: hi folks, anyone encountered this problem? tks. I don't think this has anything to do with erlang or the freeswitch erlang module, it's simply that that module's config checks are run shortly before the real failure occurs. Andrew ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FS 1.0.4 erl configure error
something is messed up in your build environment, it has nothing to do with erlang. Is this with a fresh svn checkout or tarball? Mike On Sep 13, 2009, at 10:27 AM, Nandy Dagondon wrote: hi, i want to enable odbc support which is required in mod_lcr feature. however, i encounter ./configure problem after installing Erlang R13B01. this is the portion of the error messages: ... checking for erl... /usr/local/bin/erl checking erlang version... 5.7.2 checking erlang libdir... /usr/local/lib/erlang/lib/ erl_interface-3.6.2/lib checking erlang incdir... /usr/local/lib/erlang/lib/ erl_interface-3.6.2/include checking ei.h usability... yes checking ei.h presence... no configure: WARNING: ei.h: accepted by the compiler, rejected by the preprocessor! configure: WARNING: ei.h: proceeding with the compiler's result checking for ei.h... yes checking for ei_encode_version in -lei... yes checking for ei_link_unlink in -lei... no configure: Your erlang seems OK, do not forget to enable mod_erlang_event in modules.conf configure: creating ./config.status config.status: creating src/include/switch_version.h.in .infig.status: error: cannot find input file: Makefile END i set ERL_TOP environment variable to the source directory. has anyone encountered this problem? can anyone give me a hint what's wrong. i'm compiling FS 1.0.4. thank you, /nandy ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FS 1.0.4 erl configure error
mike, got it from tarball. - nandy On Wed, Sep 16, 2009 at 11:51 AM, Michael Jerris m...@jerris.com wrote: something is messed up in your build environment, it has nothing to do with erlang. Is this with a fresh svn checkout or tarball? Mike On Sep 13, 2009, at 10:27 AM, Nandy Dagondon wrote: hi, i want to enable odbc support which is required in mod_lcr feature. however, i encounter ./configure problem after installing Erlang R13B01. this is the portion of the error messages: ... checking for erl... /usr/local/bin/erl checking erlang version... 5.7.2 checking erlang libdir... /usr/local/lib/erlang/lib/erl_interface-3.6.2/lib checking erlang incdir... /usr/local/lib/erlang/lib/erl_interface-3.6.2/include checking ei.h usability... yes checking ei.h presence... no configure: WARNING: ei.h: accepted by the compiler, rejected by the preprocessor! configure: WARNING: ei.h: proceeding with the compiler's result checking for ei.h... yes checking for ei_encode_version in -lei... yes checking for ei_link_unlink in -lei... no configure: Your erlang seems OK, do not forget to enable mod_erlang_event in modules.conf configure: creating ./config.status config.status: creating src/include/switch_version.h.in .infig.status: error: cannot find input file: Makefile END i set ERL_TOP environment variable to the source directory. has anyone encountered this problem? can anyone give me a hint what's wrong. i'm compiling FS 1.0.4. thank you, /nandy ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_radius_cdr: Multiple RADIUS messages for single call?
Can you tell why Freeswitch + mod_radius_cdr was not good for you? Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of email lists Sent: 2009 m. rugsėjo 16 d. 00:53 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_radius_cdr: Multiple RADIUS messages for single call? Thanks to those for the info and help on this issue. Ultimately ended up having to use alternative software for the radius piece (not related to any shortfalls by Freeswitch). Vlad From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Friday, September 11, 2009 11:31 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_radius_cdr: Multiple RADIUS messages for single call? set the variable process_cdr=false on that a_leg first thing in your dialplan On Fri, Sep 11, 2009 at 2:26 AM, Anatoliy Kounitskiy anato...@kounitskiy.com wrote: It's normal to have to two records for a call - Start and Stop message. From what i see - you have one start and stop for each leg of the call. Regards, AK email lists wrote: Hello, Using the mod_radius_cdr module w/ Freeswitch, I am seeing duplicate RADIUS messages being generated for individual calls (sample messages for one call below). Looking at the Acct-Unique-Session-Id and Acct-Session-Id fields, it would appear that perhaps each call leg results in a pair of start/stop RADIUS messages; is this the expected behavior? If so, is there a way to disable RADIUS messaging for what I presume is the ingress or A leg of the call? Any leads would be appreciated. Thanks in advance. Vladimir Thu Sep 10 10:37:25 2009 Acct-Status-Type = Start Acct-Session-Id = b0f387c0-35bd-e32c-971a-d79d026a8004 User-Name = 8135793256 Freeswitch-Src = 8135793256 Freeswitch-CLID = sipp Freeswitch-Dst = 14043297226 Freeswitch-Dialplan = XML Framed-IP-Address = 50.46.50.55 Freeswitch-Context = public Freeswitch-Source = mod_sofia Freeswitch-Callstartdate = 2009-09-10T10:22:00.259042-0700 NAS-Port = 0 Acct-Delay-Time = 0 NAS-IP-Address = 1.1.1.1 Acct-Unique-Session-Id = 097c8472ff7bcec7 Timestamp = 1252604245 Request-Authenticator = Verified Thu Sep 10 10:37:25 2009 Acct-Status-Type = Start Acct-Session-Id = 584f4573-46f7-655f-91d7-84cd59c9ec12 User-Name = 8135793256 Freeswitch-Src = 8135793256 Freeswitch-CLID = sipp Freeswitch-Dst = 14043297...@x.x.x.x Freeswitch-Dialplan = XML Framed-IP-Address = 50.46.50.55 Freeswitch-Context = public Freeswitch-Source = mod_sofia Freeswitch-Callstartdate = 2009-09-10T10:22:00.279044-0700 NAS-Port = 0 Acct-Delay-Time = 0 NAS-IP-Address = 1.1.1.1 Acct-Unique-Session-Id = 53f729e173e8c0a9 Timestamp = 1252604245 Request-Authenticator = Verified Thu Sep 10 10:37:57 2009 Acct-Status-Type = Stop Acct-Session-Id = b0f387c0-35bd-e32c-971a-d79d026a8004 Freeswitch-Hangupcause = Normal-Unspecified User-Name = 8135793256 Freeswitch-Src = 8135793256 Freeswitch-CLID = sipp Freeswitch-Dst = 14043297226 Freeswitch-Dialplan = XML Framed-IP-Address = 50.46.50.55 Freeswitch-Context = public Freeswitch-Source = mod_sofia Freeswitch-Lastapp = bridge Freeswitch-Billusec = 32029926 Freeswitch-Callstartdate = 2009-09-10T10:22:00.259042-0700 Freeswitch-Callanswerdate = 2009-09-10T10:22:00.319197-0700 Freeswitch-Callenddate = 2009-09-10T10:22:32.349123-0700 Acct-Session-Time = 32 NAS-Port = 0 Acct-Delay-Time = 0 NAS-IP-Address = 1.1.1.1 Acct-Unique-Session-Id = 097c8472ff7bcec7 Timestamp = 1252604277 Request-Authenticator = Verified Thu Sep 10 10:38:02 2009 Acct-Status-Type = Stop Acct-Session-Id = 584f4573-46f7-655f-91d7-84cd59c9ec12 Freeswitch-Hangupcause = Normal-Clearing User-Name = 8135793256 Freeswitch-Src = 8135793256 Freeswitch-CLID = sipp Freeswitch-Dst = 14043297...@x.x.x.x Freeswitch-Dialplan = XML Framed-IP-Address = 50.46.50.55 Freeswitch-Context = public Freeswitch-Source = mod_sofia Freeswitch-Billusec = 32049973 Freeswitch-Callstartdate = 2009-09-10T10:22:00.279044-0700 Freeswitch-Callanswerdate = 2009-09-10T10:22:00.289136-0700
