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Re: [Freeswitch-users] Module initialization failure status
by Ahmed habiba 04 Apr '16

04 Apr '16

Any other return status would be considered as failed i.e “SWITCH_STATUS_FALSE <http://docs.freeswitch.org/switch__types_8h.html#fedb3923fc3ae51937cef145c5…>” but you should print a reasonable debug message to allow the user of your module the exact failure reason. > > From: Chris Mandra <mandra(a)gmail.com> > Subject: Re: [Freeswitch-users] Module initialization failure status > Date: April 4, 2016 at 1:16:48 AM GMT+3 > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Reply-To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > > > Anything? > > On Saturday, April 2, 2016, Chris Mandra <mandra(a)gmail.com <mailto:mandra@gmail.com>> wrote: > Hi Guys - Happy Saturday. > > Quick question: what is the proper status to return if a module fails on initialization? > That is: should it always return SWITCH_STATUS_SUCCESS ? > What if something goes wrong? What status should it return? > > Thanks! > chris > > > > > -- > mandra > c:410.258.5281 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org

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Re: [Freeswitch-users] Module initialization failure status
by jelena＠misticnabica.hr 03 Apr '16

03 Apr '16

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Auto answer via Esl
by Gregor Nanger 03 Apr '16

03 Apr '16

For example, if I have sip phone that supports auto answer feature, how can I send variable or sip header via Esl to uuid? Any hints? -- Gregor Nanger *CTO* t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia • www.infomedia.si

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Playback from remote server
by Gregor Nanger 02 Apr '16

02 Apr '16

Hi! Need some help. I am trying to playback file that is on remote server, but getting error: [ERR] switch_core_file.c:253 Invalid file format [http] for [ www.myserver.com/file.wav] File is accessible via browser and mod_sndfile is loaded. Any hint what could be wrong? Best regards, Gregor

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T38->g711 gateway issues with latest master and v1.6
by dave.may＠tresta.com 01 Apr '16

01 Apr '16

Is anybody else having trouble with T38 -> g711 gateway with either v1.6 or latest master? Initially I thought my problems might have been limited to calls bridging to mod_freetdm (i.e. T38 in -> PRI out), but today I was able to reproduce the problem using SIP only. I entered a Jira on the issue yesterday, but was hoping that more eyeballs on the problem might reveal something. https://freeswitch.org/jira/browse/FS-9008 Not sure if everyone can read Jira tickets, so here's the gist of what I can see happening: - FS1 txfax to FS2 - FS2 sets the following parameters, then bridges call to FS3 ** fax_enable_t38_request=true ** execute_on_answer=t38_gateway self - FS3 answers, then calls rxfax - bidirectional g711 between FS1/FS2 and FS2/FS3 - FS1 sends T38 reinvite to FS2 - FS2 responds 100 Trying - FS2 starts sending T38 packets to FS1 (unexpected), while still sending g711 to FS3 (expected) - 39 seconds later FS2 sends BYE to FS1, but using a new CSeq/branch not seen elsewhere in either leg (?) - FS2 finally responds to T38 request from FS1, but with 486 Busy Here instead of 200 OK - FS2 sends repeated BYE to FS1 (e.g. at 1,2,4 seconds) Thanks! Dave.

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Google Voice outbound calls fail with DESTINATION_OUT_OF_ORDER
by Mervyn Yeo 01 Apr '16

01 Apr '16

Hi all, I got my GV (Google Voice) setup following the wiki and inbound calls works fine. It's the outbound calls that dies immediately after dialing. I've enabled console and dingaling logging and attached the logs here. I've also attached various configurations related to GV setup. If you see something misconfigured or know what could be the cause of the problem please do let me know. client.xml - jingle_profiles/client.xml default - outbound dialplan. part of default.xml dingaling.conf.xml - as the name implies inbound.xml - inbound dialplan gv.log - console log Thanks, Mervyn

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Re: [Freeswitch-users] How to handle invite session from pjsua
by Michael Jerris 01 Apr '16

01 Apr '16

maybe better to ask them on their mailing list? > On Apr 1, 2016, at 4:50 AM, Rajib Dey <rajib.dey(a)mindteck.com> wrote: > > Hello, > > I am using pjsua sample application and I want to handle the invite sessions from my own logic. I need access/modify remote and local sdp sessions, endpoint and socket information. Is there any API available in pjsua that I can use to achieve this? > > Thanks & Regards, > Rajib Dey > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org <mailto:consulting@freeswitch.org> > http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/> > > Official FreeSWITCH Sites > http://www.freeswitch.org <http://www.freeswitch.org/> > http://confluence.freeswitch.org <http://confluence.freeswitch.org/> > http://www.cluecon.com <http://www.cluecon.com/> > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> > http://www.freeswitch.org <http://www.freeswitch.org/>

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How to handle invite session from pjsua
by Rajib Dey 01 Apr '16

01 Apr '16

Hello, I am using pjsua sample application and I want to handle the invite sessions from my own logic. I need access/modify remote and local sdp sessions, endpoint and socket information. Is there any API available in pjsua that I can use to achieve this? Thanks & Regards, Rajib Dey

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Can mod_cdr_csv write to multiple files?
by vfclists . 31 Mar '16

31 Mar '16

The cdr_csv.conf uses the default-template option to select the output format for the csv file. <param name="default-template" value="example"/> Is there some option for multiple files, such as generating output for Postgres and say Linksys? -- Frank Church ======================= http://devblog.brahmancreations.com

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How to use $0 in dialplan?
by vfclists . 31 Mar '16

31 Mar '16

This may be due to my ignorance on how regexes are matched to parts of the dialled number. I am testing a new trunk in the dialplan by routing a single number through it. All the other numbers use the existing trunk and will be switched over later. This means I match the whole number and use it in the dial plan. However when I use $0 on its on in the dial string it works okay, but when I prefix it, eg 240$0 the $0 is not translated to the dialled number and is sent literally to the provider with the obvious results. <!-- route 1234 numbers through trunkxxx -> <condition field="destination_number" break="on-true" expression="^1234567890$"> <action application="bridge" data="sofia/gateway/trunkxxx/240$0"/> </condition> What do I need to do to send the whole matched number prefixed or suffixed with some other numbers? -- Frank Church ======================= http://devblog.brahmancreations.com

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