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Sip Allow Options
by DJB 15 Sep '09

15 Sep '09

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.

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I got ERLang to fire a configuration request
by Mark Sobkow 15 Sep '09

15 Sep '09

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, Params=~w} )~n", [Tag, Key, Value, Params]), Xml = "<document type=\"freeswitch/xml\"> <section name=\"result\"> <result status=\"not found\" /> </section> </document>", {ok, Xml }; %% %% Dialplan handler replies that the requested document section, tag, and key are not %% found. %% xml_fetch({fetch, dialplan, Tag, Key, Value, Params}) -> io:format( "freeswitch_callback:xml_fetch( {fetch, dialplan, 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 }; %% %% Default handler replies that the requested document section, tag, and key are not %% found. %% xml_fetch({fetch, Section, Tag, Key, Value, Params}) -> io:format( "freeswitch_callback:xml_fetch( {fetch, Section=~w, Tag=~s, Key=~s, Value=~s, Params=~w} )~n", [Section, Tag, Key, Value, Params]), Xml = "<document type=\"freeswitch/xml\"> <section name=\"result\"> <result status=\"not found\" /> </section> </document>", {ok, Xml }; %% %% If the request isn't recognized, just log it. %% xml_fetch( Request ) -> io:format( "freeswitch_callback:xml_fetch( Request=~w ) not recognized~n", [Request]), Xml = "<document type=\"freeswitch/xml\"> <section name=\"result\"> <result status=\"not found\" /> </section> </document>", {ok, Xml }. %% %% If the request isn't recognized, just log it and do nothing. %% handle_call(Request, _From, State) -> io:format("freeswitch_callback: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.

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ERLang configuration callbacks
by Mark Sobkow 15 Sep '09

15 Sep '09

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? Sample code follows: -module(freeswitch_bind). -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]). %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% %% 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, #st{fsnode=Node, pbxpid=Pid}}; {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}. %% %% Configuration handler replies that the requested document section, tag, and key are not %% found. %% handle_call({fetch, configuration, Tag, Key, Value, Params}, _From, State) -> io:format( "freeswitch_fetch:handle_call( {fetch, configuration, Tag=~w, Key=~w, Value=~w, Params=~w}, _From, State=~w)~n", [Tag, Key, Value, Params, State]), Xml = "<document type=\"freeswitch/xml\"> <section name=\"result\"> <result status=\"not found\" /> </section> </document>", { reply, {ok, Xml }, State }; %% %% Directory handler replies that the requested document section, tag, and key are not %% found. %% handle_call({fetch, directory, Tag, Key, Value, Params}, _From, State) -> io:format( "freeswitch_fetch:handle_call( {fetch, directory, Tag=~w, Key=~w, Value=~w, Params=~w}, _From, State=~w)~n", [Tag, Key, Value, Params, State]), Xml = "<document type=\"freeswitch/xml\"> <section name=\"result\"> <result status=\"not found\" /> </section> </document>", { reply, {ok, Xml }, State }; %% %% Dialplan handler replies that the requested document section, tag, and key are not %% found. %% handle_call({fetch, dialplan, Tag, Key, Value, Params}, _From, State) -> io:format( "freeswitch_fetch:handle_call( {fetch, dialplan, Tag=~w, Key=~w, Value=~w, Params=~w}, _From, State=~w)~n", [Tag, Key, Value, Params, State]), Xml = "<document type=\"freeswitch/xml\"> <section name=\"result\"> <result status=\"not found\" /> </section> </document>", { reply, {ok, Xml }, State }; %% %% Default handler replies that the requested document section, tag, and key are not %% found. %% handle_call({fetch, Section, Tag, Key, Value, Params}, _From, State) -> io:format( "freeswitch_fetch:handle_call( {fetch, Section=~w, Tag=~w, Key=~w, Value=~w, Params=~w}, _From, State=~w)~n", [Section, Tag, Key, Value, Params, State]), Xml = "<document type=\"freeswitch/xml\"> <section name=\"result\"> <result status=\"not found\" /> </section> </document>", { reply, {ok, Xml }, 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.

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FS Presence Implementation
by Jerry Richards 15 Sep '09

15 Sep '09

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

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FreeSWITCH Weekly Conference Call Agenda For Friday September 18
by Michael Collins 15 Sep '09

15 Sep '09

Hello, The weekly conference call agenda page is ready for everyone to contribute: http://wiki.freeswitch.org/wiki/FS_weekly_2009_09_18 Please add questions, comments, concerns, and ideas. As always, we are looking for volunteers to assist with various sub-projects. All are invited to listen in on the weekly call. It's a great way to get a feel for what's happening with the project and to communicate directly with the core developers. Thanks again for your support and for working so hard to make FreeSWITCH a successful project and for making the FreeSWITCH community such a great place to be. -Michael

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problem: no audio for one of the person in conference
by Artem Shiyanov 15 Sep '09

15 Sep '09

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_confname@my_profile +flags{mintwo}"/> </extension> conference profile "my_profile" is: <profile name="my_profile">  <param name="rate" value="8000"/>  <param name="interval" value="20"/>  <param name="energy-level" value="300"/>  <param name="caller-controls" value="none"/> <param name="comfort-noise-level" value="1400"/>  <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

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compilation error with the latest codes
by Jingwei Yang 15 Sep '09

15 Sep '09

Hi Folks, I've got a compilation error with the latest codes (r14842) Making all in packages Creating mod_sofia_la-mod_sofia.lo Compiling mod_sofia.c ... Creating mod_sofia_la-sofia.lo Compiling sofia.c ... sofia.c: In function â€˜sofia_handle_sip_r_inviteâ€™: sofia.c:3221: error: expected expression before â€˜<<â€™ token make[5]: *** [mod_sofia_la-sofia.lo] Error 1 make[4]: *** [all] Error 2 make[3]: *** [mod_sofia-all] Error 1 make[2]: *** [all-recursive] Error 1 Does anyone have ideas about this? Regards, -Jingwei

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DAHDI Dial 9 Receiving Setup Acknowledge
by Ryan Wagoner 14 Sep '09

14 Sep '09

I have a Toshiba PBX connected via a QSIG PRI to Asterisk. I can make calls from the Toshiba to Asterisk and internal calls from Asterisk to the Toshiba. What I can't do is make an call with an outside destination from Asterisk to the Toshiba. The Toshiba is looking for 9 to grab an outside line then it expects to see the 10 digits. In the FreePBX dial plan I use 9|. which sends 9 plus the 10 digit number. Using Wireshark to look at the QSIG commands coming from a Sangoma wanpipemon trace I see the following for an Asterisk to Toshiba internal call. Asterisk -> SETUP Toshiba -> CALL PROCESSING Toshiba -> CONNECT Asterisk -> CONNECT ACKNOWLEDGE However when trying to dial 9 + number I received the following Asterisk -> SETUP Toshiba -> SETUP ACKNOWLEDGE Looking at http://tools.ietf.org/html/rfc4497 I see the following On receipt of a QSIG SETUP message containing no Sending complete information element and a number in the Called party number information element that the gateway cannot determine to be complete, the gateway SHALL send back a QSIG SETUP ACKNOWLEDGE message, start QSIG timer T302, and await further number digits. Otherwise, the gateway SHALL wait for more digits to arrive in QSIG INFORMATION messages. Looking in the chan_dahdi.c code I see case PRI_EVENT_SETUP_ACK: chanpos = pri_find_principle(pri, e->setup_ack.channel); if (chanpos < 0) { ast_log(LOG_WARNING, "Received SETUP_ACKNOWLEDGE on unconfigured channel %d/%d span %d\n", PRI_SPAN(e->setup_ack.channel), PRI_CHANNEL(e->setup_ack.channel), pri->span); } else { chanpos = pri_fixup_principle(pri, chanpos, e->setup_ack.call); if (chanpos > -1) { ast_mutex_lock(&pri->pvts[chanpos]->lock); pri->pvts[chanpos]->setup_ack = 1; /* Send any queued digits */ for (x = 0;x < strlen(pri->pvts[chanpos]->dialdest); x++) { ast_debug(1, "Sending pending digit '%c'\n", pri->pvts[chanpos]->dialdest[x]); pri_information(pri->pri, pri->pvts[chanpos]->call, pri->pvts[chanpos]->dialdest[x]); } ast_mutex_unlock(&pri->pvts[chanpos]->lock); } else ast_log(LOG_WARNING, "Unable to move channel %d!\n", e->setup_ack.channel); } break; How do I get Asterisk to queue these digits so DAHDI can send them in response to the SETUP ACKNOWLEDGE message. What should be happening is Asterisk sends 9 via the SETUP message, waits for the SETUP ACKNOWLEDGE, then send the 10 digits number via a INFORMATION message. Ryan

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ATTENTION BEHAVIOR CHANGE of sip_invite_params variable.
by Brian West 14 Sep '09

14 Sep '09

I just committed revision 14849 to make sip_invite_params only apply to the RURI, If you wish to modify the To param son the invite you MUST use sip_invite_to_params moving forward. you have sip_invite_contact_params and sip_invite_from_params to work with also which were already there just making sure you know that now. Thanks, Brian West

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Re: [Freeswitch-users] Inbound Gateway Call Not Working
by Jerry Richards 14 Sep '09

14 Sep '09

Okay. I got the Grandstream Gateway's 1-stage dialing working with Freeswitch (Thank You, Michael Collins and Thank All You Developers for creating this really slick Softswitch/PBX). Here are the changes/additions I made to the XML files: conf/sip_profiles/exernal/grandstreamGXW4104.xml (added file): <include> <gateway name="192.168.72.186"> <param name="username" value="1000"/> <param name="password" value="1234"/> <param name="proxy" value="192.168.72.186"/> <param name="register" value="false"/> <param name="extension" value="1000"/> </gateway> </include> conf/dialplan/default.xml (added to existing file): <extension name="GrandstreamTest"> <condition field="destination_number" expression="^(9{0,1}\d{10})$"> <action application="bridge" data="sofia/gateway/192.168.72.186/$1(a)192.168.72.186"/> </condition> </extension> conf/dialplan/public.xml (added to existing file): <extension name="GrandstreamTest"> <condition field="destination_number" expression="^(5000)$"> <action application="transfer" data="$1 XML default"/> </condition> </extension> conf/autoload_configs/acl.conf.xml (added to existing file): <list name="lan" default="allow"> <node type="allow" cidr="192.168.72.186/32"/> ... </list> ... <list name="domains" default="deny"> <node type="allow" cidr="192.168.72.186/32"/> ... </list> Best Regards, Jerry -----Original Message----- From: Jerry Richards [mailto:jerry.richards@teotech.com] Sent: Friday, September 11, 2009 1:27 PM To: 'freeswitch-users(a)lists.freeswitch.org'; 'Michael Collins' Subject: RE: Inbound Gateway Call Not Working Thanks. I added the <node type="allow" cidr="x.x.x.x/32"/> to both the "lan" list and "domain" list in the acl.conf.xml file and it does not try to authenticate anymore. However, now it replies to the INVITE with a 480 TEMPORARILY UNAVAILABLE. Best Regards, Jerry -----Original Message----- From: Jerry Richards [mailto:jerry.richards@teotech.com] Sent: Friday, September 11, 2009 10:57 AM To: 'freeswitch-users(a)lists.freeswitch.org' Subject: RE: Inbound Gateway Call Not Working By the way, the FS DEBUG console is saying the following when an inbound call is made: Rejected by acl "domains". Falling back to Digest auth. Best Regards, Jerry -----Original Message----- From: Jerry Richards [mailto:jerry.richards@teotech.com] Sent: Friday, September 11, 2009 10:25 AM To: 'freeswitch-users(a)lists.freeswitch.org' Subject: Inbound Gateway Call Not Working I am trying to configure a Grandstream gateway to work with FS. I can make outbound calls without a problem. However, inbound calls are getting a 403 Forbidden from FS in response to the INVITE from the gateway. Now, the INVITE's from address is the caller's number (e.g. 1112223333), which ofcourse, is foreign to the FS. So the FS sends a 407 Proxy Authentication Required and the gateway uses username "Anonymous" and the uri "sip:4000@192.168.72.38" (4000 is the destination for all calls from the gateway). Is there an example configuration for this scenario? Thanks and Best Regards, Jerry

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