- FreeSWITCH-users - FreeSWITCH Mailing Lists

alert_call using bgapi
by Jason Holden 10 Apr '14

10 Apr '14

Hi. Is the normal process for the dialplan to stall on an alert_call bgapi command and complete processing of the line item before continueing on in the dialplan? If so how can I have it run the alert_call and continue in the dialplan at the same time and process additional bridge statements?

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snom 320 phone external calls not hangup
by Lloyd Aloysius 10 Apr '14

10 Apr '14

Hello I am testing the Snom320 with the new firmware. Internal calls between extensions no issues. But calling external and Hangup the line, not hangup the calls. I think this is a issue with the snom firmware. Does anyone see the similar issue. Please let me know. Thanks Lloyd

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Re: [Freeswitch-users] [ERR] sofia_reg.c:2110 "some gateway" Registration Failed with status Unauthorized [401]. failure #37 (Snabel Kaabiya)
by Snabel Kabiya 10 Apr '14

10 Apr '14

Hi, This seems to be a freeswitch bug: http://jira.freeswitch.org/browse/FS-6287 Thanks, Snabel Kaabiya On Tue, Apr 8, 2014 at 9:25 PM, < freeswitch-users-request(a)lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: mod_skinny --> Calling numbers with various lenght (MS-R-IT) > 2. Re: Delayed sound after answer (Anthony Minessale) > 3. Re: Delayed sound after answer (Oleg Stolyar) > 4. Re: [ERR] sofia_reg.c:2110 "some gateway" Registration Failed > with status Unauthorized [401]. failure #37 (Snabel Kaabiya) > (Brian West) > > > ---------- Forwarded message ---------- > From: "MS-R-IT" <martin.schoepfer(a)ms-r-it.de> > To: "'FreeSWITCH Users Help'" <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 8 Apr 2014 19:58:15 +0200 > Subject: Re: [Freeswitch-users] mod_skinny --> Calling numbers with > various lenght > Hello Nathan, > > thanks for your reply can i have checked this hack but i don´t think this > is > a solution to communicate. Are there any plans to get this function into > freeswitch in the future? > I check now how I could handle the cisco phones temporary with a small > asterisk chan_sccp connected to freeswitch. Is there any way to get > chan_sccp transformed to > Freeswitch! Because its Opensource too! I only able to program some smalls > things in C or C++ so I have to ask if it is any reason and then I can > check > it by myself! > > Thank you for your informations. I think you have luck with NANP!! :-) > > Mit freundlichen Grüßen > > Martin Schöpfer > > > ____________________________________________________________________________ > __________ > > MS-R-IT > Martin Schöpfer > Regelungs- und Informationstechnik > Bergham 3 > 83052 Bruckmühl > Tel.: 08062 7238870 > Fax.: 08062 7238871 > Mobil: 0176 80088253 > > > -----Ursprüngliche Nachricht----- > Von: freeswitch-users-bounces(a)lists.freeswitch.org > [mailto:freeswitch-users-bounces@lists.freeswitch.org] Im Auftrag von > Nathan > Neulinger > Gesendet: Dienstag, 8. April 2014 19:41 > An: FreeSWITCH Users Help > Betreff: Re: [Freeswitch-users] mod_skinny --> Calling numbers with various > lenght > > On 04/08/2014 12:10 PM, MS-R-IT wrote: > > Hello, > > > > yes they can overlap you can have different length with overlap like > > my company siemens the central has the number 3-digit prefix (yes this > can > also be) then 4-digit four our site (like 9221) and 0 for our central and > 5910 for me directly. > > > > There are many else expressions I could give you emergency call in > > Germany is 110 and 112 but there can be number with 110XXXXX too! > > This is not something you'll be able to do with the current built-in dial > handling for mod_skinny currently because of how it processes the dialing. > > The key part of the current processing for this is in > skinny_handle_keypad_button_message() where it calls > skinny_session_process_dest() with the non-null 2nd to last argument > (append_dest). > > The processing sets call to routing state after each digit in that routine. > > What would need to happen is some sort of time delayed callback to have it > not send the call to routing state until a certain time has elapsed. > > > You could do a workaround of requiring a terminating digit (# or something > like that) to "dial" - that would work with > current code, but is definitely a hack. > > > -- Nathan > > ------------------------------------------------------------ > Nathan Neulinger nneul(a)mst.edu > Missouri S&T Information Technology (573) 612-1412 > System Administrator - Architect > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > > > ---------- Forwarded message ---------- > From: Anthony Minessale <anthony.minessale(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 8 Apr 2014 13:04:48 -0500 > Subject: Re: [Freeswitch-users] Delayed sound after answer > Its probably based on your network. Are you behind NAT? Sometimes it > takes a few seconds for the automatic translation to kick in. > You can add a sleep 1000 after answer to ensure you have established media > before you play the tone. > > > > On Mon, Apr 7, 2014 at 5:10 PM, Oleg Stolyar <olegstolyar(a)gmail.com>wrote: > >> Hi guys, >> >> I have the following in my dialplan: >> >> <condition field="${<some field>}" expression="<some expression>"> >> <action application="answer"/> >> <action application="set" data="api_hangup_hook=curl <some URL>"/> >> * <action application="playback" >> data="tone_stream://v=0;%(1000,0,600)" />* >> <action application="conference" data="$1@default"/> >> </condition> >> >> Notice the line in bold. In most cases the tone plays fine but in some >> cases it is shorter than 1 second and in a few cases it does not play at >> all. Has anyone run into this? Is there a way to make the occasional >> media delay after answer go away? >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > > -- > Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ > > ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ > http://twitter.com/FreeSWITCH > ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ > <http://freeswitch.org/g+>* > > ClueCon Weekly Development Call > ☎ sip:888@conference.freeswitch.org ☎ +19193869900 > > > > ---------- Forwarded message ---------- > From: Oleg Stolyar <olegstolyar(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 8 Apr 2014 12:12:42 -0600 > Subject: Re: [Freeswitch-users] Delayed sound after answer > Thanks Anthony, I'll try that. > > > On Tue, Apr 8, 2014 at 12:04 PM, Anthony Minessale < > anthony.minessale(a)gmail.com> wrote: > >> Its probably based on your network. Are you behind NAT? Sometimes it >> takes a few seconds for the automatic translation to kick in. >> You can add a sleep 1000 after answer to ensure you have established >> media before you play the tone. >> >> >> >> On Mon, Apr 7, 2014 at 5:10 PM, Oleg Stolyar <olegstolyar(a)gmail.com>wrote: >> >>> Hi guys, >>> >>> I have the following in my dialplan: >>> >>> <condition field="${<some field>}" expression="<some expression>"> >>> <action application="answer"/> >>> <action application="set" data="api_hangup_hook=curl <some >>> URL>"/> >>> * <action application="playback" >>> data="tone_stream://v=0;%(1000,0,600)" />* >>> <action application="conference" data="$1@default"/> >>> </condition> >>> >>> Notice the line in bold. In most cases the tone plays fine but in some >>> cases it is shorter than 1 second and in a few cases it does not play at >>> all. Has anyone run into this? Is there a way to make the occasional >>> media delay after answer go away? >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting(a)freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> >> -- >> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >> >> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >> http://twitter.com/FreeSWITCH >> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >> <http://freeswitch.org/g+>* >> >> ClueCon Weekly Development Call >> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > > ---------- Forwarded message ---------- > From: Brian West <brian(a)freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 8 Apr 2014 13:25:05 -0500 > Subject: Re: [Freeswitch-users] [ERR] sofia_reg.c:2110 "some gateway" > Registration Failed with status Unauthorized [401]. failure #37 (Snabel > Kaabiya) > This is a bug in the provider, If they want to start auth over they should > say stale=‘true’, this way the NC will start back at 1. not 18e > -- > 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) > > > > > > > > > > > > > > On Apr 8, 2014, at 7:07 AM, Steven Ayre <steveayre(a)gmail.com> wrote: > > > It's not the first request because it's sending a challenge response > using a nonce, which it must have received in a previous challenge (401)... > It can't just make that up. > > > _______________________________________________ > 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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avmd usage help or issue?
by Malay Thakershi 10 Apr '14

10 Apr '14

Hello, Trying to detect VM from C# mod_managed. I have seen weird behavior making concurrent outbound calls. Even though VM is detected on call A, call B (answered by real person) also gets hung up. Is it possible that the beep from call A is getting mixed up with call B? I use session.Execute("avmd", "start") command and then listen in EventReceivedFunction for avmd_detect variable. Do I have to check for uuid? Please help. Code: mObjMainSession.EventReceivedFunction = (ev) => { Log.WriteLine(LogLevel.Notice, "~~avmd_detect:" + mObjMainSession.GetVariable("avmd_detect")); if (mObjMainSession.GetVariable("avmd_detect") == "TRUE") { Log.WriteLine(LogLevel.Notice, "~~VM detected"); Log.WriteLine(LogLevel.Notice, aObjCntx.Session.uuid + "-" + "VM detected"); mObjMainSession.Execute("avmd", "stop"); return "AVMD"; } return ""; }; Log.WriteLine(LogLevel.Notice, "~~Started AVMD"); mObjMainSession.Execute("avmd", "start");

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This channel was not configured to use INFO DTMF message while reviewing VoiceMail messages
by Sean Devoy 09 Apr '14

09 Apr '14

Hi, I get "sofia.c:7679 IGNORE INFO DTMF(7) (This channel was not configured to use INFO DTMF!)" or similar for any key pressed while reviewing VoiceMail. I will say up front, this is just a warning and everything WORKS fine. However, I would like to understand what I am missing in my config. I have googled it with very limited success. endpoint is Cisc0 SPA504g FS Version: FreeSWITCH Version 1.2.22+git~20140309T212137Z~65fed130e5~64bit (git 65fed13 2014-03-09 21:21:37Z 64bit) Thanks, Sean

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MediaWiki->Confluence Migration (Call Tomorrow April 9th 2014 1PM Central)
by Brian West 09 Apr '14

09 Apr '14

FreeSWITCHers, We are starting the migration from MediaWiki, we’ve tried various ways to import the data we have now and it didn’t end well at all. We would have to do more work to fix all those broken links. So we are fleshing out the new layout and migrating the data little by little from MediaWiki to Confluence. If you want an account to the sandbox area to assist in migrating the documentation we have, formatting it and reading thru it as we migrate it over from MediaWiki, Please contact wking(a)freeswitch.org for an account. Tomorrow at 1PM Central we will have a call in sip:888@conference.freeswitch.org or +1-919-386-9900. If you’re interested in helping out please show up. We do need as many hands as possible working on this to get it done quickly. Once the migration is complete we’ll go thru and fill in docs on areas that are currently either missing or not well thought out. The new wiki will not copy the layout exactly that is on MediaWiki but more well thought out TOC. To view our work in progress please visit http://confluence.freeswit.ch This is the time for everyone that can’t contribute code to step up and give back. You can get information about Confluence here: https://confluence.atlassian.com/display/DOC/Confluence+User%27s+Guide Hope to see a few people there tomorrow. 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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mod_sofia sig11 with openssl-1.0.1g on lucid_x64
by Mihály Árva-Tóth 09 Apr '14

09 Apr '14

Hello, I compiled latest FS master branch (7bc92a11) on Ubuntu Lucid x64 with openssl-1.0.1g. Based on this thread: http://lists.freeswitch.org/pipermail/freeswitch-users/2014-February/102724… wget http://www.openssl.org/source/openssl-1.0.1g.tar.gz tar -zxvf openssl-1.0.1g.tar.gz cd openssl-1.0.1g ./config --prefix=/usr/local/openssl101g -fPIC make make install ln -s /usr/local/openssl101g /usr/local/openssl And then compile FS with: ./configure --enable-sse MOD_CFLAGS="-g -ggdb3" CFLAGS="-I/usr/local/openssl/include" LDFLAGS="-L/usr/local/openssl/lib" make make install Compile and install were successful. But when I try to start freeswitch, I get segmentation fault when loading mod_sofia. I'm not advanced debugger, when I run gdb: Program terminated with signal 11, Segmentation fault. #0 0x00007f89b0e876b8 in ssl_cipher_strength_sort () from /usr/local/lib/libfreeswitch.so.1 Have you got any idea what's the wrong? If I follow these steps on Ubuntu Precise that is works and no sig11. Thank you! -- Best regards, Mihály Árva-Tóth System Engineer

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Re: [Freeswitch-users] [ERR] sofia_reg.c:2110 "some gateway" Registration Failed with status Unauthorized [401]. failure #37 (Snabel Kaabiya)
by Snabel Kabiya 09 Apr '14

09 Apr '14

Hi, This seems to be a freeswitch bug: http://jira.freeswitch.org/browse/FS-6287 Thanks, Snabel Kaabiya ---------- Forwarded message ---------- > From: Brian West <brian(a)freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 8 Apr 2014 13:25:05 -0500 > Subject: Re: [Freeswitch-users] [ERR] sofia_reg.c:2110 "some gateway" > Registration Failed with status Unauthorized [401]. failure #37 (Snabel > Kaabiya) > This is a bug in the provider, If they want to start auth over they should > say stale='true', this way the NC will start back at 1. not 18e > -- > 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) > > > > > > On Apr 8, 2014, at 7:07 AM, Steven Ayre <steveayre(a)gmail.com> wrote: > > > It's not the first request because it's sending a challenge response > using a nonce, which it must have received in a previous challenge (401)... > It can't just make that up. > > > _______________________________________________ > 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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ClueCon Weekly! Today 1PM Easter! Topic Heartbleed SSL!
by Ken Rice 09 Apr '14

09 Apr '14

Hey FreeSWITCHers, As you are all aware, Heartbleed has popped up showing us a bad weakness in SSL/TLS. Join us today on ClueCon Weekly to discuss how this affects your FreeSWITCH SIP/TLS installations. When: Today, Wed Apr 9 at 1PM Eastern, 10AM Pacific Where: SIP:888@conference.freeswitch.org or visit http://wiki.freeswitch.org/wiki/Weekly_Conference_Call_Calling_Instructions for other access methods. For you guys that want to get those speaking slots at ClueCon, Get those proposals in Now to marketing(a)cluecon.com! Also don¹t forget Fridays Bug Hunting Sessions following the VUC also on the FreeSWITCH Community Bridge. K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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context problem
by Koby Peleg Hen 09 Apr '14

09 Apr '14

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