August 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

FREE webinar video about Auto-Dialer Business Model (Telemarketing)
by Kolmisoft Marketing 30 Aug '11

30 Aug '11

Hello, We would like to share our webinar about Auto-Dialer Business Model (Telemarketing). It is educational video which we made for our clients and now we are sharing it with you. http://www.kolmisoft.com/how-to-start-a-VoIP-business/webinars/ Enjoy. NOTE: This is not attempt to sell you anything. No product or service is sold/marketed in the video. Regards, Kolmisoft Team www.kolmisoft.com

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Start Conference with first user
by Mitja Thomas 30 Aug '11

30 Aug '11

Hello List, I try to start a conference in mod_conference when the first member attempts to join. This may sound weird, but the conference is set to wait for the moderator anyhow and I want to use the enter-sound to play something like "There is currently no Moderator in this room. Please hold the line". If the moderator joins, the enter-sound setting will be set to none from dialplan. I thought I just have to NOT set the mintwo member flag, but that didnt do the trick. When the first member (no moderator) join there is no enter-sound, just the moh file. When the second member joins (no moderator) both get the enter-sound. I thought there was a bug with the mintwo flag and I was close to submit to JIRA because of that, but looking further into the code Im not sure if the mintwo flag is even supposed to behave the way I want to. It just kills the conference when less then two users are in it. So what I want to know, am I expecting the wrong behaviour from the mintwo member flag? If so, is there a different approach to tell the conference users right at the beginning if there is a moderator present or not. Regards, Mitja

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Error compling jsxml.c
by D Foulkes 30 Aug '11

30 Aug '11

Hi, I am still fairly new to freeswitch... Having problems with a setup on a Ubuntu 10.10, so I decided to do a rebuild from scratch. Following recommended instructions on the wiki installation guide (using git clone) I get the following error when running the compile: make && make install ... libtool: compile: gcc -DOSSP -DXP_UNIX -DEXPORT_JS_API -DJS_HAS_FILE_OBJECT=1 -DJS_HAS_XML_SUPPORT=1 -DJS_THREADSAFE=1 -Insprpub/dist/include/nspr -Insprpub/pr/include -DHAVE_CONFIG_H -Isrc -O2 -DNDEBUG -pipe -c src/jsxml.c -fPIC -DPIC -o src/.libs/jsxml.o src/jsxml.c: In function ‘PutProperty’: src/jsxml.c:4571: error: expected expression before ‘>’ token src/jsxml.c:4571: error: stray ‘\37’ in program src/jsxml.c:4571: error: stray ‘`’ in program src/jsxml.c:4572: error: stray ‘\37’ in program src/jsxml.c:4572: error: stray ‘\37’ in program src/jsxml.c:4572: error: stray ‘\37’ in program src/jsxml.c:4572: error: stray ‘\37’ in program src/jsxml.c:4572: error: stray ‘\37’ in program src/jsxml.c:4573: error: stray ‘\37’ in program src/jsxml.c:4573: error: stray ‘\37’ in program src/jsxml.c:4587: error: ‘JSXML’ has no member named ‘xml’ src/jsxml.c:4587: error: stray ‘`’ in program src/jsxml.c:4587: error: expected ‘;’ before ‘value’ src/jsxml.c:4588: error: expected expression before ‘goto’ src/jsxml.c:4591: error: expected expression before ‘/’ token src/jsxml.c:4592: error: expected statement before ‘)’ token src/jsxml.c:4592: warning: missing terminating ' character src/jsxml.c:4592: error: missing terminating ' character src/jsxml.c:4593: error: ‘JSXMLQName’ has no member named ‘localOame’ src/jsxml.c:4593: error: expected expression before ‘)’ token make[5]: *** [src/jsxml.lo] Error 1 make[4]: *** [/usr/local/src/freeswitch/libs/js/libjs.la] Error 2 make[3]: *** [mod_spidermonkey-all] Error 1 make[2]: *** [all-recursive] Error 1 make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 Any help would be appreciated. Thanks, Daniel

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Question -- DTMF from RFC2833 to SIP INFO
by Dan Lan 29 Aug '11

29 Aug '11

Hi, I am new to the Freeswitch community, just compiled my first FS window version one month ago. I have registered two SNOM 300 IP-Phone with extension 1001, 1002 into FS. I also setup an external gateway as SIP trunk to service provider. (For certain reason, the service provider only accept INFO as the DTMF signal.) I add the following line into the external gateway profile <param name="dtmf-type" value="info"/> I also add the following line into the dail-plan <action application = "set" data="dtmf_type=info"/> When I dail the phone number from extension 1001, the channel bridge through the SIP trunk with both-way audio no problem. However, when I enter the DTMF, the signal did not send to the provider SIP trunk with SIP INFO. from the log file, I can see the DTMF was recorgnized by FreeSwitch from extension 1001 (it was sending RFC2933), and I can see FS try to send signal to the external trunk. (but no SIP INFO or RFC2833 in Wireshark capture) [DEBUG] switch_rtp.c:3303 RTP RECV DTMF 1:1280 2011-08-29 15:44:10.081713 [DEBUG] switch_ivr_bridge.c:391 Send signal sofia/external/2131234567@xxx.xxx.xxx.xxx:5060 [BREAK] 2011-08-29 15:44:11.033081 [DEBUG] switch_rtp.c:3303 RTP RECV DTMF 2:1120 2011-08-29 15:44:11.033081 [DEBUG] switch_ivr_bridge.c:391 Send signal sofia/external/2131234567@xxx.xxx.xxx.xxx:5060 [BREAK] 2011-08-29 Did I do something wrong here? Is FS capable of convert RFC2833 signal from one leg and send SIP INFO to another leg?

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Dialed number identification service
by Aakash 29 Aug '11

29 Aug '11

Hi, Is this possible - Dialed number identification service in freeswitch. For eg: I have configured my two inbound number to the same extension(eg 1000).When calling party dials any of the two inbound number should able to show the DNIS to the routed extension(eg 1000). Thanks Aakash -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Dialed-number-identification-… Sent from the freeswitch-users mailing list archive at Nabble.com.

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freeswitch php
by Sam 29 Aug '11

29 Aug '11

Does anyone know how to retrieve channel variables (ie. uuid, etc.) using the php example that was shown in the wiki below? #!/usr/bin/php -q <?php // set a couple of things so we dont kill the system ob_implicit_flush(true); set_time_limit(30); // Open stdin so we can read the data in $in = fopen("php://stdin", "r"); // Connect echo "connect\n\n"; // Answer echo "sendmsg\n"; echo "call-command: execute\n"; echo "execute-app-name: answer\n\n"; // Play a prompt echo "sendmsg\n"; echo "call-command: execute\n"; echo "execute-app-name: playback\n"; echo "execute-app-arg: /usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav\n\n"; // Wait sleep(5); // Hangup echo "sendmsg\n"; echo "call-command: hangup\n\n"; fclose($in); ?>

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Strange problem with Freeswitch and external extension
by Charles Bujold 29 Aug '11

29 Aug '11

I have a remote extension which registers with the Freeswitch server. The user can record his greeting in the Freeswitch IVR without a problem. Yet if the user calls me I cannot hear him, but he can hear me. I thought at first that it was a NAT issue so we placed the remote telephone directly on the internet and the same problem remains. The Freeswitch server is behind a NAT/router and I can see the packets from the phone coming into the router and being forwarded to the Freeswitch server. The router is not blocking the packets and is forwarding them to the correct server IP. I them placed wireshark on the Freeswitch server and I can see the packets leave the Freeswitch server and going to the phone But I cannot see the Freeswitch server receive the packets from the PFsense router, even though I can see in the router that the packets are being forwarded properly to the Freeswitch server IP. The Freeswitch server is running Ubuntu with no firewall, so in theory I should see the server accept or reject the packets. Yet I do not see any in wireshark coming from the phone, only those leaving from the Freeswitch server. What I do not understand is why, since he is registered, I cannot hear him, but yet he can interact with Freeswitch and record a greeting? Obviously packets are getting routed properly and Freeswitch recognizes them, some of the time. Can somebody suggest a possible explanation and solution to this problem? Thanks cjb

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CallWithUs
by Robert Huddleston 29 Aug '11

29 Aug '11

I already reviewed the wiki - does anyone have experience with CallWithUs and using IP Auth / not registration based?

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Looking up variable from config
by Rob Hutton 29 Aug '11

29 Aug '11

Is there any way of looking up a variable from the XML config from the dialplan? For instance, I know what number is dialed from ${destination_number} and I know that the directory user entry attribute of ID matches the number because I have it set up that way. Can I lookup the value email-addr parameter from the user and, say, set a variable to the value? Something like an XPATH lookup?

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mod_conference - 2-digit DTMF and events
by Michael Kemp 29 Aug '11

29 Aug '11

I have set up a custom caller-controls group in the my conference.conf.xml, as follows <caller-controls> <group name="mick"> <control action="event" digits="#1"/> <control action="event" digits="#2"/> <control action="event" digits="#3"/> <control action="mute" digits="#6"/> </group> </caller-controls> I have a lua script running to detect these events and perform actions according to which digits it receives. However, unless I press both the 2nd digit very quickly after the first one the event is not raised Is there any way of tuning the inter digit delay for caller-control groups so that multi-digit DTMF sequences are handled correctly?

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