- FreeSWITCH-users - FreeSWITCH Mailing Lists

YeaLink :(
by Blake Priddy 09 Mar '14

09 Mar '14

Anyone had an issue when calling in to a yealink ext it just goes to vm and says NO_USER_RESPONSE in the logs. -- *Blakelund Priddy* Network & Systems Engineer Bryant Public School District Bryant, Arkansas 72022 http://www.bryantschools.org p 501-653-5038 f 501-847-5656

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Dialplan record on demand - email recording on exit?
by Sean Devoy 09 Mar '14

09 Mar '14

Hi All, I am learning to setup the record on demand (via dtmf) from the wiki. Seems pretty straight forward, except my user needs the sound file when done. How can I email the sound file when the session ends? Of course, there may not be a recording as it is "on demand". Thanks, Sean

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mod_dingaling fails to build on latest git as of this writing
by covici＠ccs.covici.com 09 Mar '14

09 Mar '14

Here is the output from the make, is this known or do I need to file a jira? making all mod_dingaling Makefile:559: /usr/src/freeswitch/libs/libdingaling/src/.deps/mod_dingaling_la-libdingaling.Plo: No such file or directory Makefile:560: /usr/src/freeswitch/libs/libdingaling/src/.deps/mod_dingaling_la-sha1.Plo: No such file or directory make[4]: *** No rule to make target '/usr/src/freeswitch/libs/libdingaling/src/.deps/mod_dingaling_la-sha1.Plo'. Stop. Makefile:544: recipe for target 'mod_dingaling-all' failed make[3]: *** [mod_dingaling-all] Error 1 Makefile:460: recipe for target 'all-recursive' failed make[2]: *** [all-recursive] Error 1 Makefile:2949: recipe for target 'all-recursive' failed make[1]: *** [all-recursive] Error 1 Makefile:930: recipe for target 'all' failed make: *** [all] Error 2 Thanks in advance for any suggestions. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici(a)ccs.covici.com

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Using reason header from SIP 183 message inside dialplan
by Bunea Lucian 08 Mar '14

08 Mar '14

Hello, I have a Vodafone SIP trunk connected to my freeswitch box. Everything works fine excepting one thing: I am unable to differentiate when a dialed number is busy or unavailable (line disconnected, mobile phone out of coverage area, &c). My provider signals this kind of situations inside a SIP 183 Session Progress message with reason header (user busy, subscriber absent, ...). How can I use/get that reason header inside dialplan? Sofia log uploaded here: http://pastebin.freeswitch.org/22101 Regards, Lucian

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Attended transfer issue
by Volkan Oransoy 08 Mar '14

08 Mar '14

Hi all, https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer#Example2 I am trying to apply this recipe on my system but there is a strange behavior. When B tries tries to connect C, B calls itself instead of C. I have captured the invite packet that B send to C, Request uri is B itself. I couldn't find what I am missing. Also if there is another way for attended transfer? I have found "uuid_transfer" in wiki but I am not sure that I can use for attended transfers. Thanks in advance, /Volkan

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JIRA is now https
by Brian West 08 Mar '14

08 Mar '14

FreeSWITCHers, I would say if you have a problem report it on JIRA, But chances are if you had a problem with https you couldn’t report it on JIRA. So just let me know if you see anything or can’t see anything. :) FYI people are still in the Friday Call… drop by. -- Brian West brian(a)freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)

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Recent changes in git Master
by Brian West 07 Mar '14

07 Mar '14

FreeSWITCHers, These two commits will require you to rebootstrap and configure. If you run into any issues please join us on #freeswitch or reply on the list. If its a real bug please file a JIRA. commit 2513388d8adf088acb69350ef9e00b8ad117698a Author: Michael Jerris <mike(a)jerris.com> Date: Mon Mar 3 16:39:18 2014 -0500 clean up some bootstrap warnings commit bcd9f49fbe7fb992c5e0fdac68e158731cc98420 Author: Michael Jerris <mike(a)jerris.com> Date: Thu Feb 27 14:16:54 2014 -0500 move applications to use automake Thanks, -- Brian West brian(a)freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)

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Proxy media for specific gateway
by Nikolay Rogoshchenkov 07 Mar '14

07 Mar '14

Hello, Is it possible to proxy media only(!) for specific gateway? By default in dialplan I have * <action application="set" data="bypass_media=true"/>* and I use a lua script for dynamic routing, so can I simply add *[bypass_media=false,...]* to my *dialstring *when specific gateway will be chosen? Thank you. PS: Freeswitch Version 1.5.6b -- Best regards Rogoshchenkov Nikolay

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Re: [Freeswitch-users] SIP REFER - URI Parameters
by Phil Dunks 07 Mar '14

07 Mar '14

Hi Anthony. Thanks for looking at this and exposing the sip_refer_to_params variable. I’ve updated to head, and can see the change in sofia.c. Sorry for being dense, but I’ve not had much luck getting the uri params from the REFER to the INVITE. My dial plan is very simple. <include> <extension name="public_did"> <condition field="destination_number" expression="(\d+)"> <action application="set" data="proxy_media=true"/> <action application="bridge" data="sofia/gateway/${distributor(load-balance ${sofia(profile external gwlist down)})}/$1"/> </condition> </extension> </include> I tried appending ${sip_refer_to_params} to the bridge uri, but no luck. It seems the REFER is handled by sofia_handle_sip_i_refer in sofia.c which sets up a blind transfer. I’ve been staring at sofia.c all morning - but not worked out what I need to do. Please could you shed some light on how to proceed. Thanks again, Phil From: Anthony Minessale <anthony.minessale(a)gmail.com> Subject: Re: [Freeswitch-users] SIP REFER - URI Parameters Date: 28 February 2014 19:15:11 GMT To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Reply-To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Added a patch to expose sip_refer_to_params vars. See latest HEAD. On Fri, Feb 28, 2014 at 11:17 AM, Phil Dunks <phil.dunks(a)netdev.co.uk> wrote: Hi Guys I’m trying to use FS (1.5.8b from last week) as an SBC and load-balancer (using mod_distributor). It is load balancing between several FS conference nodes. It’s working a treat, apart from one scenario : Say conference xyz is already running on conf-node-1. Next incoming call is sent to conf-node-2, pin is collected, and conference xyz is identified as the required conference. Conf-node-2 checks presence, and sees the conference is already running on conf-node-1, and does a deflect. We add some query params to the refer-to URI to tell conf-node-1 not to collect the pin, but put the caller straight into the conference. e.g <action application="deflect" data=“sip:1234@10.1.1.2;conf=xyz" /> In the REFER I can see the params : Refer-To: <sip:1234@10.1.1.2;conf=xyz> But they are not present on the request URI or To header in the INVITE to conf-node-1. So conf-node-1 requests the PIN again. Is there any way I can pass this info to the other conference node by using the REFER? Thanks for your help. Phil

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How do I Diagnose my FreeSwitch Crash? It just disappeared.
by Sean Devoy 07 Mar '14

07 Mar '14

Hi, I looked up and noticed my phones all had ORANGE lights. My net was up, but FS was down. I logged into my CENTOS server and did a "service freeswitch status" and I got "freeswitch dead but subsys locked". I followed up with service freeswitch stop then service freeswitch start. I have NEVER had FS die on me in over a year of use. So, I am clueless how to determine the problem. The log show absolutely nothing as far as I can tell. Normal call info: 2014-03-05 14:25:54.398282 [DEBUG] sofia.c:6364 Processing updated SDP 2014-03-05 14:25:54.398282 [DEBUG] switch_core_session.c:1016 Send signal sofia/external/201@<some domain address> [BREAK] 2014-03-05 14:25:54.398282 [DEBUG] sofia.c:5785 Channel sofia/external/201@<some domain address> entering state [completed][200] 2014-03-05 14:25:54.488278 [DEBUG] switch_core_session.c:1016 Send signal sofia/external/201@<some domain address> [BREAK] 2014-03-05 14:25:54.488278 [DEBUG] switch_core_session.c:1016 Send signal sofia/external/201@<some domain address> [BREAK] 2014-03-05 14:25:54.488278 [DEBUG] switch_core_session.c:1016 Send signal sofia/external/201@<some domain address> [BREAK] 2014-03-05 14:25:54.508277 [DEBUG] sofia.c:5785 Channel sofia/external/201@<some domain address> entering state [ready][200] 2014-03-05 14:26:23.528468 [NOTICE] sofia_reg.c:418 Registering vitelity-inbound Then what appears to be startup messages: 2014-03-05 14:30:01.349506 [NOTICE] switch_loadable_module.c:225 Adding Dialplan 'enum' 2014-03-05 14:30:01.349611 [NOTICE] switch_loadable_module.c:267 Adding Application 'enum' 2014-03-05 14:30:01.349716 [NOTICE] switch_loadable_module.c:313 Adding API Function 'enum' 2014-03-05 14:30:01.349801 [NOTICE] switch_loadable_module.c:313 Adding API Function 'enum_auto' 2014-03-05 14:30:01.350225 [DEBUG] mod_cdr_csv.c:364 Adding default template. 2014-03-05 14:30:01.350248 [DEBUG] mod_cdr_csv.c:411 Adding template sql. 2014-03-05 14:30:01.350264 [DEBUG] mod_cdr_csv.c:411 Adding template example. 2014-03-05 14:30:01.350280 [DEBUG] mod_cdr_csv.c:411 Adding template snom. 2014-03-05 14:30:01.350289 [DEBUG] mod_cdr_csv.c:411 Adding template linksys. 2014-03-05 14:30:01.350302 [DEBUG] mod_cdr_csv.c:411 Adding template asterisk. 2014-03-05 14:30:01.350313 [DEBUG] mod_cdr_csv.c:411 Adding template opencdrrate. 2014-03-05 14:30:01.350386 [NOTICE] switch_loadable_module.c:313 Adding API Function 'cdr_csv' ... Thanks in advance. Sean

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