July 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Check if UA is still there?
by Peter Steinbach 03 May '13

03 May '13

Hello, is there a way to check if a UA is still available? The reason is, that sometimes a channel hangs and is not being terminated because the UA was disconnected due to a network failure. As the media is outside Freeswitch (bypass media), it is difficult for us to determine, if the call is still active. Meanwhile the B leg to PSTN is also not hungup and produces costs. Is there a way Freeswitch can detect this? Or is there a way Freeswitch may send a SIP message to the UA by an external process and wait for an OK which can be captured e.g. by event socket interface? An help is welcome. -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de

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MPL v2
by Puskás Zsolt 12 Apr '13

12 Apr '13

Hi. A new version of the MPL license just came out ( http://www.opensource.org/licenses/MPL-2.0 ). Is the new version better than the last one? I'm trying to find the differences but this kind of text is to hard for me :) Any plans for changing FreeSWITCH license to the new one ? Zsolt

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Playing of mms stream by means of FS
by Yuriy Nasida 05 Mar '13

05 Mar '13

Hi guys What the best way for the playing of mms stream like this mms://live.rfn.ru/rcult_64 by means of FS ?I try to use mod_shout but without success. Do I have to use mplayer or vlc for this ?I saw this in wiki http://wiki.freeswitch.org/wiki/Mod_vlc But... probably there is the more better way ? Please adviseThanks.

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ooH323 vs mod_h323
by Saeed Ahmed 17 Feb '13

17 Feb '13

Dear FSers, which one is better to use when there are higher number of concurrent calls? *ooH323 *or *mod_h323*. Thanks

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Callgroup keeps ringing after answer - changed from shared extensions
by Mario G 04 Feb '13

04 Feb '13

Been at this for 3 weeks now, hope someone can help.... This started after replacing shared extensions with unique users and updating from an early 2011 git. Ring group via "pointer" works fine as it always has. However, answer via callgroup variable intermittently (30-40%) leaves one or mores phone ringing even though the the call was answered at another extension. The darn thing is that I can't reproduce it myself to get a trace. Any help/suggestions greatly appreciated. Phones are SPA962 (UDP) and Bria on iPad (TCP, same problems) and they all stopped ringing when they shared IDs. Mario G default.xml ring-group has (abbreviated to save space)------------------: <group name="home"> <users> <user id="100" type="pointer"/> <user id="101" type="pointer"/> ..... more users </users> </group> <group name="work"> <users> <user id="200" type="pointer"/> <user id="201" type="pointer"/> </users> </group> </groups> Extension----------------------: <user id="101"> User: <params> <param name="password" value="$${default_password}"/> </params> <variables> <variable name="voicemail_domain" value="xxxxxx.com"/> <variable name="toll_allow" value="domestic,domestic10,international,local"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Front Desk"/> <variable name="effective_caller_id_number" value="101"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/>  <variable name="callgroup" value="home"/> </variables> </user> Bridge command-----------------: <extension name="xxphone"> <condition field="destination_number" expression="^2920$"> <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <action application="bind_meta_app" data="2 b s record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/>  <action application="execute_extension" data="2925 XML default"/>  <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="ringback=file_string://${xxsounds}hi-long.wav!${sayname}!tone_stream://${us-ring};loops=-1"/>  <action application="bridge" data="{originate_timeout=17}${group_call(home@${domain_name}+A)}"/>

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Routing inbound call to multiple extentions
by Todd Bailey 31 Jan '13

31 Jan '13

Hi All, At witt's end on this, how do I get a inbound call to go to multiple extension? I've tried groups but so far no avail I using a spa 3102 as a analog adapter that is configured to rout calls to extension 1000, I want FS to automatically forward this call to extensions 1001-1009. Here is an extract from default.xml (/usr/local/freeswitch/conf/directory) <group name="frontoffice"> <users>  <user id="1001" type="pointer"/> <user id="1002" type="pointer"/> <user id="1003" type="pointer"/> <user id="1004" type="pointer"/> <user id="1005" type="pointer"/> <user id="1006" type="pointer"/> <user id="1007" type="pointer"/> <user id="1008" type="pointer"/> <user id="1009" type="pointer"/> </users> </group>

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Webrtc
by Tamas Jalsovszky 28 Jan '13

28 Jan '13

Hello, Is there any plan to support webrtc in FS? Right I've found this post about asterisk: http://blogs.digium.com/2012/05/23/asterisk-11-webrtc/ T.

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mod_stress
by BF 22 Jan '13

22 Jan '13

Hello, I'd like to know what the "stress write" from mod_stress module does. I have hard time finding documentation or an example about mod_stress module. Anyone? Thank you. Eric

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Sofia fails to detect offline gateway
by Steven Ayre 03 Jan '13

03 Jan '13

Hi Guys, A couple of our gateways have been shut down for hardware upgrades. However, the FreeSWITCH servers didn't spot the gateways go offline so they're still trying to send calls to them. I see this in the logs: 2011-01-11 09:35:05.396610 [WARNING] sofia.c:3981 Ping failed gw4 with code 503 - count 1/0/10, state UP 2011-01-11 09:35:05.396610 [WARNING] sofia.c:3981 Ping failed gw3 with code 503 - count 1/0/10, state UP 2011-01-11 09:35:13.408674 [WARNING] sofia.c:3981 Ping failed gw4 with code 503 - count 1/0/10, state UP 2011-01-11 09:35:13.408674 [WARNING] sofia.c:3981 Ping failed gw3 with code 503 - count 1/0/10, state UP 2011-01-11 09:35:21.420884 [WARNING] sofia.c:3981 Ping failed gw4 with code 503 - count 1/0/10, state UP 2011-01-11 09:35:21.420884 [WARNING] sofia.c:3981 Ping failed gw3 with code 503 - count 1/0/10, state UP 2011-01-11 09:35:29.458114 [WARNING] sofia.c:3981 Ping failed gw4 with code 503 - count 1/0/10, state UP 2011-01-11 09:35:29.458114 [WARNING] sofia.c:3981 Ping failed gw3 with code 503 - count 1/0/10, state UP 2011-01-11 09:35:37.472424 [WARNING] sofia.c:3981 Ping failed gw4 with code 503 - count 1/0/10, state UP 2011-01-11 09:35:37.472424 [WARNING] sofia.c:3981 Ping failed gw3 with code 503 - count 1/0/10, state UP 2011-01-11 09:35:46.475934 [WARNING] sofia.c:3981 Ping failed gw4 with code 503 - count 1/0/10, state UP 2011-01-11 09:35:46.475934 [WARNING] sofia.c:3981 Ping failed gw3 with code 503 - count 1/0/10, state UP 2011-01-11 09:35:55.489967 [WARNING] sofia.c:3981 Ping failed gw4 with code 503 - count 1/0/10, state UP 2011-01-11 09:35:55.489967 [WARNING] sofia.c:3981 Ping failed gw3 with code 503 - count 1/0/10, state UP ... you get the idea. I've done some testing and it happens both when FS is running and the gateway goes offline (code 408 then) and when FS starts up with the gateway already offline (code 503 as above). Both IP addresses are entirely offline. I had expected these would go to state DOWN but they haven't, I just get the above repeating. The gateway definitions look like this: <gateway name="gw3"> <param name="realm" value="a.b.c.d"/> <param name="username" value="nouser"/> <param name="password" value="nopass"/> <param name="register" value="false"/> <param name="caller-id-in-from" value="true"/> <param name="ping" value="5"/> <param name="ping-max" value="10"/> <param name="ping-min" value="1"/> <param name="retry-seconds" value="5"/> <param name="expire-seconds" value="60"/> </gateway> Is there a mistake in my configuration? -Steve

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XMPP Presence?
by Gavin Henry 03 Dec '12

03 Dec '12

Hi all, We're running our own XMPP chat server and would like to tie this in to FreeSWITCH. Has anyone tested presence via XMPP? The wiki is very light on this and would be happy to update. Thanks.

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