September 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

play_and_get_digits - why is minimum 1 digit?
by Nathan Neulinger 20 Sep '13

20 Sep '13

Is there any particular technical reason that the minimum number of digits for play_and_get_digits is 1 instead of 0? There have been a few cases where I was wanting to have "enter a destination phone number followed by pound sign, or enter just pound sign to cancel" type of prompt. Lowering the minimum to zero would accomplish this, but is there some reason not to do that? -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

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Transferring XML metadata as part of SIP INVITE
by Siri MM 20 Sep '13

20 Sep '13

Hi, I have a requirement where, after recording sessions using FS, I need to make a SIP call to a third party device, and then transfer the recorded file to it as part of this call. In order to accomplish this, my plan is to originate a call to this third party SIP device, and bridge the call to my extension, and the dialplan of my extension would playback the recorded file to the third party device. Couple of questions: 1. Any other better ways of achieving this? 2. I also need to transfer the recorded file's metadata in the form of an XML, as part of the SIP INVITE to the device. Although I am aware of adding custom headers using "sip_h", I am not sure how to add a XML file as part of the SIP message. Would appreciate any inputs Thanks!

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Suitable log files for freeswitch?
by Melanie Treitinger 20 Sep '13

20 Sep '13

Hi, I started to work with freeswitch and I am really impressed with its features. But I'm quite unhappy with the log files freeswitch provides. Unlike Asterisk the call uuid is not included in every line so that I can't separate the log entries that belong to different calls. I already added <param name="uuid" value="true" /> to logfile.conf.xml, but it doesn't affect all functions that log something. So my question ist: how do you handle different calls in the freeswitch log files? Are there any methods you can recommend? Is anyone already working on a solution? Thanks a lot! antonellababy

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On-conference distortion for FreeTDM channels, but not SIP channels
by Bob Hartwig 20 Sep '13

20 Sep '13

I've tried everything I can think of on this, so now I'm reaching out to you good people. I have a FreeSWITCH system that has the latest version of FreeTDM, Dahdi, libpri, and a Rhino R4T1 quad T1 board with up-to-date firmware. When I call into one of the PRI channels, a "playback"-played file sounds clean, but when I go into conference, the conference audio is distorted. This does not happen with a SIP channel. I would classify the distortion as a "crackle", but it is very regularly periodic. Repeating ticks at (roughly) 10-20 Hz repeat rate, I would say. I can only hear them when audio is present, i.e. the audio seems to be modulated with the ticks. And that conference audio can be the MOH guitar recording. I've eliminated the following: - T1 clocking - Echo Cancellers (I've turned on and off all combinations of hardware and software EC) - T1 board firmware (it's the latest, and the problem only happens on-conference). - Packet length - it's the same (20 mS) for the conference engine, FreeTDM, and Dahdi - mu-law vs. a-law encoding - behaves identically. - Comfort noise - happens with and without. - conference sampling rate (tried 8000 and 16000, even though I'm using TDM, which is 8000). - FreeSWITCH version - happens on 1.2 stable, the latest master branch, and an older version that we were using. If you haven't heard of anything like this, could you please give me your thoughts on why this could happen only on-conference, not while playing WAV files, and only with FreeTDM channels? Thanks! Bob

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Cordless phones
by Jonas Gauffin 19 Sep '13

19 Sep '13

Hi, I'm looking for IP-dects where a base station can handle 5 phones (which all should be able to talk at the same time). All suggestions are welcome. Regards, Jonas

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Re: [Freeswitch-users] rpm build fails
by Babak Yakhchali 19 Sep '13

19 Sep '13

I tried to use scripts/ci/rpmbuilder.sh but it failed with these mssags: [babak@localhost freeswitch-stable]$ ./scripts/ci/rpmbuilder.sh stable cp: cannot stat `src_dist/*': No such file or directory error: Macro %BUILD_NUMBER has empty body error: Macro %BUILD_NUMBER has empty body error: File /home/babak/src/freeswitch-stable/rpmbuild/SOURCES/freeswitch-stable.0.0.tar.bz2: No such file or directory mkdir: cannot create directory `/home/babak/src/freeswitch-stable/RPMS': File exists mv: cannot stat `/home/babak/src/freeswitch-stable/rpmbuild/*/*.rpm': No such file or directory ---------------------------------------------------------------------- The vstable.0.0- RPMs have been rolled, now we just need to push them to the YUM Repo ---------------------------------------------------------------------- On Wed, Sep 18, 2013 at 11:29 PM, Babak Yakhchali < babak.freeswitch(a)gmail.com> wrote: > Hi > I was trying to create freeswitch rpm from stable git branch but I get > this error in the last line: > > Processing files: freeswitch-devel-stable-1.el6.x86_64 > error: File not found by glob: > /home/babak/rpmbuild/BUILDROOT/freeswitch-stable-1.el6.x86_64/usr/lib64/freeswitch/mod/*.a > RPM build errors: > File not found by glob: > /home/babak/rpmbuild/BUILDROOT/freeswitch-stable-1.el6.x86_64/usr/lib64/freeswitch/mod/*.a > > I attached the full log to this email. > >

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Suitable log files for freeswitch?
by Melanie Treitinger 19 Sep '13

19 Sep '13

Steve, you're right - with <param name="uuid" value="true" /> most log lines have a uuid. But look at the following example - we tested entering DTMF digits from two calls simultaneously. As you can see, for each call there is only one line which has a uuid and several others which have not. There is no way to separate these lines in case you need more information on this issue - especially if you image having 30 or 100 simultaneous calls. 2013-09-19 14:28:46.453687 [DEBUG] ftmod_wanpipe.c:1678 [s3c17][3:18] read wanpipe event 3 2013-09-19 14:28:46.473372 [DEBUG] ftmod_wanpipe.c:1678 [s3c16][3:17] read wanpipe event 3 2013-09-19 14:28:46.613375 [DEBUG] ftmod_wanpipe.c:1678 [s3c17][3:18] read wanpipe event 3 2013-09-19 14:28:46.613375 [DEBUG] ftmod_wanpipe.c:1547 [s3c17][3:18] Queuing wanpipe DTMF: 9 2013-09-19 14:28:46.613375 [DEBUG] ftdm_io.c:3778 [s3c17][3:18] Queuing DTMF 9 (debug = 0) 2013-09-19 14:28:46.633372 [DEBUG] ftmod_wanpipe.c:1678 [s3c16][3:17] read wanpipe event 3 2013-09-19 14:28:46.633372 [DEBUG] ftmod_wanpipe.c:1547 [s3c16][3:17] Queuing wanpipe DTMF: 3 2013-09-19 14:28:46.633372 [DEBUG] ftdm_io.c:3778 [s3c16][3:17] Queuing DTMF 3 (debug = 0) 2013-09-19 14:28:46.633372 [DEBUG] mod_freetdm.c:815 Queuing DTMF [9] in channel FreeTDM/3:17/453025556 device 3:17 2013-09-19 14:28:46.633372 [DEBUG] mod_freetdm.c:815 Queuing DTMF [3] in channel FreeTDM/3:16/453025556 device 3:16 90a6c771-f499-489b-b956-96807707e611 2013-09-19 14:28:46.633372 [DEBUG] switch_channel.c:471 RECV DTMF 9:2000 899aa236-8a86-4433-9f9c-734f985023b3 2013-09-19 14:28:46.633372 [DEBUG] switch_channel.c:471 RECV DTMF 3:2000

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Re: [Freeswitch-users] Operator features
by Mr.Gab 19 Sep '13

19 Sep '13

Thank you Andrew, Do you also know if there is a possibility to try this feature with a softphone before buying an hardware phone? Thanks On 19 September 2013 *11:56*, Andrew Cassidy <andrew at cassidywebservices.co.uk> <freeswitch-users%40lists.freeswitch.org?Subject=%5BFreeswitch-users%5D%20Op…> wrote: A number of phones can do this, and some even have add-on modules to increase the number of buttons you have to play with. The ones that come to mind are: - Snom 370 + Expansion - Cisco SPA508G + SPA500S Expansion Module There are most likely others too, I'm sure other people on this list will add other suggestions. Obviously choose what best fits your budget and requirements. You'll need to configure presence in FreeSWITCH inf your sofia profile, then configure the keys as required and the extras in the dialplan for call pickup. How to configure the phones varies.

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Re: [Freeswitch-users] Operator features
by Lloyd Aloysius 19 Sep '13

19 Sep '13

Also please check the default dialplan configuration. Call Pcikup and other most useful features in it. Thanks and regards, Aloysius Lloyd SunTel Technologies Inc. www.SunTelTechnologies.ca<http://www.sunteltechnologies.ca> ** t. +1-416-479-0505 x 203 e. lloyd.aloysius(a)sunteltech.ca *Business Phone System | Hosted PBX | IT Support | VoIP Solutions | Cloud Services* * * On Thu, Sep 19, 2013 at 3:56 AM, Andrew Cassidy < andrew(a)cassidywebservices.co.uk> wrote: > A number of phones can do this, and some even have add-on modules to > increase the number of buttons you have to play with. The ones that come to > mind are: > > - Snom 370 + Expansion > - Cisco SPA508G + SPA500S Expansion Module > > There are most likely others too, I'm sure other people on this list will > add other suggestions. > > Obviously choose what best fits your budget and requirements. > > You'll need to configure presence in FreeSWITCH inf your sofia profile, > then configure the keys as required and the extras in the dialplan for call > pickup. How to configure the phones varies. > > > On 18 September 2013 23:32, Mr.Gab <crazygabry(a)gmail.com> wrote: > >> Hello Everybody! >> >> i need to have the possibility of sip termination status checking >> (available, busy etc) from a central voip hardware phone (i read that it is >> possible with SNOM), and i need also the possibility to pick a call >> directed to another extension pressing the relative button. >> >> Can someone explain me how to configure freeswitch and suggest me a phone? >> >> Thanks in advice :) >> Gabriele >> >> _________________________________________________________________________ >> 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 >> >> > > > -- > *Andrew Cassidy BSc (Hons) MBCS SSCA* > Managing Director > > > *T <info(a)cassidywebservices.co.uk> *03300 100 960 *F<info(a)cassidywebservices.co.uk> > *03300 100 961 > *E <info(a)cassidywebservices.co.uk> *andrew(a)cassidywebservices.co.uk > *W <info(a)cassidywebservices.co.uk> *www.cassidywebservices.co.uk > > _________________________________________________________________________ > 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 > >

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H323 trunks
by Richard Mace 19 Sep '13

19 Sep '13

Hi all, we have an existing traditional PBX which supports h323. Would it be possible to have multiple sip phones talking to freeSWITCH and the h323 trunks going from freeSWITCH to the traditional PBX, which would then route the calls out to the existing isdn trunks? Richard

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