October 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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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mod_httapi problem when trying to play an audio hosted in S3 amazon cloud
by Leonardo N. S. Pereira 01 Feb '13

01 Feb '13

Hi. I'm having problem when trying to play an audio hosted in S3 Amazon cloud The MP3 URL is: / / /h//ttp://test-media-dev.s3.amazonaws.com/u-asevyeard22pl6sc2z4knzi%2Fblind_willie.mp3?Expires=1349381157&AWSAccessKeyId=AKIAJKQT6QPSI3BXJJ4A&Signature=bnJgjbnxqlgsdYIcKIbYUa9gqao%3D/ The result is: 2012-10-04 20:27:28.891298 [DEBUG] switch_ivr.c:605 sofia/external/+15752222222(a)67.231.8.195 Command Execute playback(http://test-media-dev.s3.amazonaws.com/u-asevyeard22pl6sc2z4knzi%2… EXECUTE sofia/external/+15752222222(a)67.231.8.195 playback(http://test-media-dev.s3.amazonaws.com/u-asevyeard22pl6sc2z4knzi%2… 2012-10-04 20:27:28.903180 [ERR] mod_httapi.c:2545 File at url [http://test-media-dev.s3.amazonaws.com/u-asevyeard22pl6sc2z4knzi%2Fblind_wi…] is unreachable! If I use the Browser with same URL, I'm able to listen to the sample audio. Any idea about what is wrong? Thanks in advance, Leonardo

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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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Re: [Freeswitch-users] Install ASTPP
by Joao Cesar / Consultor TI 05 Jan '13

05 Jan '13

Em 25/10/2012 10:46, Michael Jerris escreveu: > Try sending this email to the mailing list? > > On Oct 25, 2012, at 8:41 AM, Joao Cesar / Consultor TI <consultoriaticesar(a)gmail.com> wrote: > >> Anyone know how to install the ASTPP really, because the tutorial site, have not the slightest condition to do. >> Thank you in advance. >> >> -- >> Joao Cesar S Pereira >> Consultor TI >> ---------------------------------------- >> (31)3044-2576 >> (31)8626-6943 >> ---------------------------------------- >> Gentileza, se nao houver necessidade >> não imprima. A natureza agradece. >> ---------------------------------------- >> >> -- Joao Cesar S Pereira Consultor TI ---------------------------------------- (31)3044-2576 (31)8626-6943 ---------------------------------------- Gentileza, se nao houver necessidade não imprima. A natureza agradece. ----------------------------------------

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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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FreeSwitch Video Calls Bria iPhone Client
by Mike Burlingame 27 Dec '12

27 Dec '12

I have been playing around trying to get my iPad and iPhone using Bria to work with video it *seems* that FreeSwitch is passing the signaling to both devices and FreeSwitch is holding the media due to the NAT issue of the devices - I just wanted to know before I spent too much time on this (not a high priority) if anyone else has gotten FreeSwitch to work with Bria for iPhone or Android? Thanks

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Call disconnect upon re-invite - FS tears down call
by Phil Quesinberry 05 Dec '12

05 Dec '12

I'm experiencing an odd issue here where a direct media call (bypass media mode) between two Aastra endpoints is disconnected when one of the endpoints sends a re-invite 2 minutes into the call, presumably to refresh the session - RTP ports & codecs remain unchanged. FreeSWITCH passes the invite on to the other endpoint which replies with its own invite, again keeping the same call parameters, ports/codecs etc. FS then sends a 491 - Request Pending to both endpoints, acknowledges the second endpoint's invite and tears down the call. This seems wrong. is this expected behavior? I went ahead and filed a Jira on this just in case - it includes all trace info: http://jira.freeswitch.org/browse/FS-4754 Phil Quesinberry Q Systems Engineering, Inc. Electronic Controls and Embedded Systems Development (410) 969-8002 http://www.qsystemsengineering.com

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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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