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Rejected by acl "net_list_5". Falling back to Digest auth.
by Karol 20 Apr '11

20 Apr '11

OK, I am having some issues here. What I have going on here, is 2 phones (one aastra 6730i and 1 nortel 1535) at home. Using the suppiled router by verzion fios (actiontec), I connect to a vps out on the internet. This box is running the latest stable freeswitch (1.0.7). I keep getting a net_list_5 error, I've tried using UPNP and STUN server. Nothing has helped, I've also reset my router thinking something is wrong. Below is the errors, I see if trying to make a call. Phone to phones works and it should work. [DEBUG] sofia.c:6551 IP xxx.xxx.xxx.xxx Rejected by acl "net_list_5". Falling back to Digest auth. [WARNING] sofia_reg.c:1246 SIP auth challenge (INVITE) on sofia profile 'sipinterface_1' for [110(a)voip.lightnologies.com] from ip xxx.xxx.xxx.xxx Anyone have any ideas?? Thanks in advance.

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[freeswitch] <defunct> when using cmd="exec" in freeswitch.xml
by Mitja Thomas 20 Apr '11

20 Apr '11

Hi there, we tried to set up the FreeSWITCH and other Applications, so that we can configure them easier and more centralised. Thus we defined some Environment Variables (using CentOS) which hold often used Configuration Parameter like MySQL IP or FS Event Socket IP. We tried to integrate these Env Variables into the FS conf files by executing a shell Skript in freeswitch.xml via cmd="exec" which prepares an conf file which we include into freeswitch.xml: <X-PRE-PROCESS cmd="exec" data="sh conf/make_my_vars.sh"/> <X-PRE-PROCESS cmd="include" data="my_vars.xml"/> This works as expected and the pre defined variables in my_vars can be accessed from the other config Files, except that when we start our FreeSWITCH a zombie child process is spawned. # ps -eaf | grep free ippbx 22191 22190 4 09:41 pts/1 00:00:01 /opt/app/voip/ippbx/bin/freeswitch -waste -nonat -hp ippbx 22197 22191 0 09:41 pts/1 00:00:00 [freeswitch] <defunct> What I wanna know is: Is this a FS missbehaviour or do we use this in a wrong way? make_my_vars.sh: F="conf/my_vars.xml" echo "" > $F echo "<include>" >> $F fs_ip=`printenv MY_FS_IP` if test -n "$fs_ip" then echo '<X-PRE-PROCESS cmd="set" data="my_fs_ip='$fs_ip'"/>' >> $F fi ... echo "</include>" >> $F my_vars.xml (after FS startup):  <include> <X-PRE-PROCESS cmd="set" data="my_fs_ip=***.***.***.***"/> ... </include> Regards Mitja

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Help with rxfax debug "T4 expires in phase T30_PHASE_B_RX"
by Sadhika Sharma 20 Apr '11

20 Apr '11

Hi, I need help in debugging hangup cause 548, 'Disconnected after permitted retries' while attempting rxfax(). Logs show that T4 expires in phase T30_PHASE_B_RX. T30 PHASE B concerns exchange of DIS/DCS messages. After sending DIS, mod spandsp keeps waiting for DCS and exhausts its retries. Please guide me how to debug the root cause of loss of DCS message. Logs are mentioned below: 2011-03-07 15:53:48.675180 [DEBUG] switch_ivr.c:557 OpenZAP/4:1/43851609 Command Execute rxfax(/srv/fax/in/fcf15a4c-48a4-11e0-919e-7bf53bcc3f81.tiff) 2011-03-07 15:53:53.334614 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 17 2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 DIS: 2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 .... ...0= Store and forward Internet fax (T.37): Not set 2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 .... .0..= Real-time Internet fax (T.38): Not set : : 2011-03-07 15:53:53.875238 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Start T4 2011-03-07 15:53:53.895315 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17 2011-03-07 15:53:53.915402 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 HDLC signal status is Carrier down (-1) in state 17 2011-03-07 15:53:57.335032 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 T4 expired in phase T30_PHASE_B_RX, state 17 2011-03-07 15:53:57.335032 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Retry number 1 Please guide me how to debug the root cause of loss of DCS message. Thanks, Sadhika

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mod_cepstral not working
by covici＠ccs.covici.com 20 Apr '11

20 Apr '11

Hi. I am having trouble compiling mod_cepstral after moving to a 64-bit operating system and libraries, although maybe this has nothing to do with the problem. When the module tries to load, I get 2011-04-19 17:31:12.145808 [CRIT] switch_loadable_module.c:928 Error Loading module /usr/local/freeswitch/mod/mod_cepstral.so **/usr/local/freeswitch/mod/mod_cepstral.so: undefined symbol: swift_port_set_voice_by_name** Now swift does work by itself, so I am not sure what is happening here. Thanks in advance for any assistance. -- 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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Port mapping disabled warning in Freeswitch log
by Stephen Wilde 19 Apr '11

19 Apr '11

If I set log level to warning, I see in the log many row as: [WARNING] switch_nat.c:589 NAT port mapping disabled My freeswitch is running as: "./freeswitch -nonat" because it has a public IP and all connected gateways have also a public IP. To avoid this warning I have removed the parameter "-nonat" but if I do "show nat_map" in fs_cli I see "0 total.". I have always started Freeswitch with "-nonat", any advise to solve that?

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Re: [Freeswitch-users] Playing tones
by Clive Lansink 19 Apr '11

19 Apr '11

Hi Michael and others. The thing I really liked about Freeswitch when I downloaded the install, installed it on a Windows PC here and got it running, was how many things were working right out of the box. Of course there are the warnings about its default configuration being unsuitable for production use, but on the other hand it means I could get phones registered and start playing with stuff within half an hour, and functions like music on hold, transferring calls, voice mail etc, were working just as you would expect with a turnkey system. Well obviously I then spent time learning more about what goes on under the hood and getting us hooked up to a SIP provider etc but really that was quite straightforward, although I have had some experience with VOIP which obviously helped. My external script is written in Python and I do my Python developing on my laptop which is separate from the PC running Freeswitch. When I'm happy enough with it, I turn it into an executable and then run it on the same PC as Freeswitch. I think when you work in Windows as I do, the embedded scripting is not an option. In fact the Freeswitch scripts directory is empty I think. I had actually done some work with Yate earlier this year so I already had my programming framework established to handle messages between the script an external PABX so it didn't take long before I had the basic IVR functionality working. As an aside, we have quite a lot of home automation here including software that runs the house, and I very quickly realised that I could easily set up a system whereby you can enter simple commands on any phone to control certain things. All our phones, including cordless and mobile phones, are now remote controls that can control everything from entertainment to heating, and it only took a few hours. Such are the benefits of sockets programming. Anyway I digress. I'll have to experiment further with playing tones but won't have time today. Sometimes we have to do real work. I may look at it again tomorrow or Friday. And by the way, although right now my use of Freeswitch has been largely domestic, I do have a small and possibly commercial IVR application in mind to turn my attention to later when I am more familiar with it. Clive Lansink Email: Clive(a)Lansink.Co.NZ Phone: +64 9 520-4242 Mobile: +64 21 663-999 Fax: +64 21 789-150 -----Original message----- From: Michael Collins <msc(a)freeswitch.org> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Subject: Re: [Freeswitch-users] Playing tones Reply-to: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Date: Tue, 19 Apr 2011 09:36:44 -0700 Clive, Welcome to FreeSWITCH! If you're doing event socket stuff after only two weeks then I must tip my hat to you. Well done! I've never seen this symptom before so I'm at a bit of a loss as to what is happening. Out of curiosity, what language are you using for your event socket communications? I'm wondering if you can send the tone_stream without the "loops=-1" over and over again and receive the same symptom where the playback speeds up. The other test I would try is doing something like "loops=25" or some relatively high number. Again see if the symptom presents itself. (I'm wondering if this happens only on loops=-1 or if it happens any time there is a loop.) Let us know what you discover. -MC On Mon, Apr 18, 2011 at 7:01 PM, Clive Lansink <clive(a)lansink.co.nz> wrote: > Hello > > I am relatively new to Freeswitch; I've played with it for about two weeks. > It is now fully installed as our home PABX system. I've also done some > elementary IVR stuff using an external script communicating to event_socket. > It can monitor Freeswitch events, answer calls, play and record files and > get digits so I suppose it does all the basics. > > But I'm having difficulty playing tones. A simple tone is no problem like > the UK busy tone, %(400,200,400,450);%(400,2000,400,450), but things go > wrong when I try loops. > > In my external script when I want to play a tone, I sende a sendmsg request > with the appropriate UUID. In the headers, I have: > call-command: execute > execute-app-name: gentones > execute-app-arg: %(400,200,400,450);%(400,2000,400,450) > ] > That works, but if I want the ring tone to repeat for an indefinite period, > the header then becomes: > execute-app-arg: %(400,200,400,450);%(400,2000,400,450);loops=-1 > When I do this, I hear one ring tone, then a single DTMF digit is played, > presumably the 1 of loops=-1. I've also tried like loops=5 and got the same > result. > > After seeing some examples, I then tried the following, again using the > sendmsg command: > call-command: execute > execute-app-name: playback > execute-app-arg: > tone_stream://%(400,200,400,450);%(400,2000,400,450);loops=-1 > ] > This time it does at least seem to loop indefinitely, but after the first > two or so loops, the speed increases (not the pitch) so that the tone > repeats much more quickly than it should. > > Any help would be much appreciated. It seems to me that using playback in > this way must be accessing the same raw functionality as gentones, so I > don't understand why the result would be different. But in the end I guess I > don't care which method I use but it would be good to understand why both > methods currently don't work for me. > > By the way, I am using a recent version of Freeswitch, a pre-compiled > version running on Windows XP. > > Cheers. > > > Clive Lansink > Email: Clive(a)Lansink.Co.NZ > Phone: +64 9 520-4242 > Mobile: +64 21 663-999 > Fax: +64 21 789-150 > > _______________________________________________ > 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 > _______________________________________________ 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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Success
by Eric Beard 19 Apr '11

19 Apr '11

Hello, I just wanted to pass on my thanks and compliments to the FreeSwitch team. I have been working on a migration from a home grown sip server over to FreeSwitch during the last few weeks, and things are going very well so far. I am up to originating 140 concurrent calls, and the server (a relatively modest 1U Dell box) is humming along at around 7% CPU. Call quality is perfect. I plan to transition all of my production machines over to FS in the near future. Thanks everyone! ----------------------- Eric Z. Beard, CTO Loop LLC w (877) 850-2010 x9249 m (727) 776-2768 eric(a)loopfx.com<mailto:eric@loopfx.com>

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How does mod_lcr handle invalid ANIs?
by Max Clark 19 Apr '11

19 Apr '11

Hello, I am calling the lcr module from the dialplan like this: <action application="lcr" data="$2 default ${caller_id_number} intrastate"/> So I can get interstate vs. intrastate routing (I dropped the intralata column from the database). How does mod_lcr handle an invalid ANI? Which rate is used to route for this condition? Thanks, Max

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nibblebill cdr
by budi wibowo 19 Apr '11

19 Apr '11

hi is there any possibility that mod_nibblebill create cdr with call pricing? right now i just see the balance deducted, in fs_cli console i can see the call info regards budi

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BUG FIX: "Buffer size sanity check failed!" drops FAX receiving unneeded
by Markus Mueller 19 Apr '11

19 Apr '11

Hello FreeSwitch users and programmers, I found a problem on receiving faxes and want to share a working patch for this. The problem is that on receiving a fax, it is unneeded aborted by a sanity check. Sanity checks are fine, but a unneeded abort instead of a warning is in productive versions not the best solution. The message apearing is: 2011-04-04 10:44:52.060860 [CRIT] switch_core_codec.c:660 Buffer size sanity check failed! which is normaly aborting in receiving the fax. Simply decreasing this fault to a warning let the server receive the fax without any problems. After the patch the message apears up to five times per fax before the fax is beeing accepted. I am using libpri with the three HFC ISDN Cards and the DAHDI from Debian Squeeze 6.0. For more informations about my hardware just write me an email. Regards, Markus Mueller http://projekte.priv.de/ root@sip:/usr/local/src/freeswitch/src# diff -U 4 switch_core_codec.c* --- switch_core_codec.c 2011-03-14 10:49:17.000000000 +0100 +++ switch_core_codec.c.org 2011-03-14 10:47:02.000000000 +0100 @@ -657,9 +657,9 @@ uint32_t frames = encoded_data_len / codec->implementation->encoded_bytes_per_packet; if (frames && codec->implementation->decoded_bytes_per_packet * frames > *decoded_data_len) { switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "Buffer size sanity check failed!\n"); - // return SWITCH_STATUS_GENERR; + return SWITCH_STATUS_GENERR; } } if (codec->mutex) switch_mutex_lock(codec->mutex); root@sip:/usr/local/src/freeswitch/src#

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