September 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Calling all bug reporters and patch submitters
by Michael Jerris 27 Sep '10

27 Sep '10

Please take a few minutes to update any issues you have on http://jira.freeswitch.org today. If you have an open bug, please re-test it with current git master and confirm if it still exists, if you have a patch posted, please update it to current git master. Please do this even if your issue has been sitting for a long time. We are going to aim for a commit/fix fest this week and this will assist us in processing your issues. Also, please note that we have merged quite a number of different projects in jira into a single FreeSWITCH project with many components so that we can improve our release management. If you don't have any issues on jira, but you would like to lend a hand, find an open bug that has not been updated recently and see if you can reproduce the issue with current git master. Thanks Mike

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Struggling with installing.
by Peter Schrock 26 Sep '10

26 Sep '10

Hello, I am new to FreeSWITCH email group. I am having some issues, so I apologize if this issue is old history. If it is, please help direct me to the appropriate archive that I may look up the information. I am having an issue with installing FreeSWITCH on my MacMini OS X 10.6.4. I have purchased the book on FreeSWITCH to help wrap my brain around it. I am following the install instructions and it says to modify the "modules.config", which I have to include flite. The problem is that I keep getting these errors and I don't know how to resolve the error. This is what it says when I use the command "sudo make": (extra info deleted for brevity) /usr/include/unistd.h:495: error: conflicting types for ‘swab’ /usr/include/string.h:137: error: previous declaration of ‘swab’ was here make[7]: *** [find_sts_main.o] Error 1 make[6]: *** [build/i386-darwin10.4.0/obj//.make_build_dirs] Error 2 make[5]: *** [/usr/local/freeswitch-1.0.6/libs/flite-1.3.99/build/libs/libflite_cmu_us_awb.a] Error 2 make[4]: *** [all] Error 1 make[3]: *** [mod_flite-all] Error 1 make[2]: *** [all-recursive] Error 1 And then it ends the "sudo make" command. Can some one please help? Any help would be greatly appreciated. Thanks in advance. Peter Schrock -- "People should not be afraid of their governments, governments should be afraid of their people." -V-

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Perl background script crashes FreeSWITCH...
by Klaus Hochlehnert 25 Sep '10

25 Sep '10

Hi, I'm currently testing the newest trunk of FS with Ubuntu 10.04. I have a perl background script that will be started through FS (perl.conf.xml). This script uses Perl-DBI with sqlite. And when adding this script to perl.conf.xml freeswitch will immediately crash when this script starts. On my old FS installation (somewhere between 1.0.4 and 1.0.6) it works fine. It also works fine when started standalone. Any ideas? Here's the backtrace: (gdb) bt #0 0x00007ffa2e64bbac in sqlite_st_execute () from /usr/lib/perl5/auto/DBD/SQLite/SQLite.so #1 0x00007ffa2e641f7f in XS_DBD__SQLite__st_execute () from /usr/lib/perl5/auto/DBD/SQLite/SQLite.so #2 0x00007ffa2f1a8a37 in XS_DBI_dispatch () from /usr/lib/perl5/auto/DBI/DBI.so #3 0x00007ffa3564f045 in Perl_pp_entersub () from /usr/lib/libperl.so.5.10 #4 0x00007ffa3564d336 in Perl_runops_standard () from /usr/lib/libperl.so.5.10 #5 0x00007ffa355f4548 in Perl_call_sv () from /usr/lib/libperl.so.5.10 #6 0x00007ffa2f1a6d1a in XS_DBI_dispatch () from /usr/lib/perl5/auto/DBI/DBI.so #7 0x00007ffa3564f045 in Perl_pp_entersub () from /usr/lib/libperl.so.5.10 #8 0x00007ffa3564d336 in Perl_runops_standard () from /usr/lib/libperl.so.5.10 #9 0x00007ffa355f0df7 in Perl_eval_sv () from /usr/lib/libperl.so.5.10 #10 0x00007ffa355f11b0 in Perl_eval_pv () from /usr/lib/libperl.so.5.10 #11 0x00007ffa35916dc3 in Perl_safe_eval (my_perl=0x1e8b950, string=0x0) at /usr/src/freeswitch/src/mod/languages/mod_perl/mod_perl.c:69 #12 0x00007ffa35917c80 in perl_parse_and_execute (my_perl=<value optimized out>, input_code=<value optimized out>, setup_code=<value optimized out>) at /usr/src/freeswitch/src/mod/languages/mod_perl/mod_perl.c:138 #13 0x00007ffa35917f70 in perl_thread_run (thread=<value optimized out>, obj=0x19b5bf0) at /usr/src/freeswitch/src/mod/languages/mod_perl/mod_perl.c:275 #14 0x00007ffa498159ca in start_thread () from /lib/libpthread.so.0 #15 0x00007ffa48d4d6fd in clone () from /lib/libc.so.6 #16 0x0000000000000000 in ?? () Thanks, Klaus

2 2

Sonus possibly stomping all over DTMFs that *it* sends out...
by Mahesh Paolini-Subramanya 25 Sep '10

25 Sep '10

This doesnt directly apply to Freeswitch, but I just wanted to know if anybody else has noticed anything along these lines. The specific problem applies to the path below, for rfc2833 DTMFs Cell Phone --------> Sonus SIP Provider --------> Freeswitch In particular, if the user on the Cellphone hammers home a bunch-a DTMFs really really rapidly, only a few of them get to Freeswitch. I *think* I know where the problem lies (hint: S***s). Lets see, where do I begin. Sonus does DTMF handling *differently* from normal people. e.g., anybody who cares should know about http://www.submityoursip.com/wiki/Sonus_NBS) In addition to that, Sonus seems to have set their Minimum DTMF Duration to 320ms. Which would be fine and dandy, except that due to the miracle of How Sonus Works, it just stomps all over data when it extends the duration of the DTMF. e.g., Consider the situation where the Cell Phone user hammers home a couple of DTMFs (987) really really rapidly. Just for arguments sake, consider that each DTMF that was entered was around 130ms in length, and had, oh, 50ms between them. Here is what *should* be sent from Sonus to FS Time: 0 --> DTMF 9 Time: 130 --> DTMF 9 ends Time: 180 --> DTMF 8 Time: 310 --> DTMF 8 ends Time: 360 --> DTMF 7 Time: 490 --> DTMF 7 ends i.e., FS should actually receive 987 However, here is what actually happens when Sonus decides to do *its* version of Minimum DTMF Duration. Time: 0 --> DTMF 9 Time:320 --> DTMF 9 ends ******* There is no DTMF 8 here. Ever, given the above durations. ******* Time: 360 --> DTMF 7 Time: 490 --> DTMF 7 ends i.e., FS actually receives 97 To recap, what Sonus seems to do is t 1) Look at the incoming DTMF stream 2) If the DTMF is greater than Minimum DTMF Duration, leave it alone 3) If the DTMF is less than Minimum DTMF duration, then create additional RTP EVENT packets to make up the difference, and (this is the goofy part) replace the packets in the original stream with the ones it created. For outbound DTMF through Sonus, this is rarely a problem since most people leave their min_dtmf_duration as the default (400ms). If, however, you reduce min_dtmf_duration to, say, 150ms, you *will* have issues on outbound DTMF! That said, for SIP calls *from* Sonus *to* Freeswitch, you have no such guarantees on the originator of the DTMFs - especially if that is the PSTN. There is probably no real solution to this, apart from either dumping your Sonus based provider for someone else, or getting them to change their configuration (yeah, likely) So, back to the original question - anyone else noticed this? Mahesh Paolini-Subramanya | CTO | mahesh(a)aptela.com | 703.386.1500 Ext. 9100 2250 Corporate Park Drive | Suite 150 | Herndon, VA | www.aptela.com Check out our Blog | Follow us on Twitter | Refer a Friend

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where to put perl code to add cdr info
by covici＠ccs.covici.com 25 Sep '10

25 Sep '10

I would like to insert the cdr information when the call hangsup, rather than some kron job or something -- is there a way I can use hangup_api_hook to execute the perl code, or is this too early for cdr_csv to have populated the variables? Thanks. -- 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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FreeSwitch - mod_skypopen alsa driver on centos 64 bit, x86_64
by Shamun toha md 25 Sep '10

25 Sep '10

Hello, Alsa driver is installed, but i have no sound. I was trying to use alsamixer or etc. But nothing is just allowing me. ex: http://gist.github.com/raw/596026/41aff3b9d8fe24492cfa8fab57ecaa3da02e69f7/… Thanks for guiding..

2 11

No caller ID sent to a bridged outside call
by Mario 25 Sep '10

25 Sep '10

I am bridging a call from either inside or outside to an emergency number on an att cellphone which works fine. I am trying to pass the original caller id or anything else for that matter as the caller id/name. I only get "unknown". I also tried to send to a VOIP number and it also gets "unknown" so I am thinking it is not att. I tried various combinations of things I found on the web but nogo. Anyone see the problem below? Thanks! Mario <extension name="mx911"> <condition field="destination_number" expression="^2911$"> <action application="set" data="effective_caller_id_name=Emergency"/> <action application="set" data="effective_caller_id_number=9161234567"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="sofia/gateway/uid1/19161112222"/> </condition> </extension>

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[OT] Own Posts
by Ross McKillop 25 Sep '10

25 Sep '10

I don't receive my own posts to this list, despite enabling it on the Mailman control panel ? Is it disabled globally or am I doing something wrong? Ross

2 1

Call Forwarding Problem
by Mike 24 Sep '10

24 Sep '10

Hi, Here's a problem I posted about on the IRC some time ago - I said I'd do some traces and put them in pastebin, which I've done here: http://pastebin.freeswitch.org/13970 A brief description of the problem and the calls involved in the pastebin. Incoming calls are delivered to the Freeswitch from a SIP trunking provider who are also used to route outgoing calls. This is not via a gateway. The SIP trunking provider insists that for outgoing calls, the data in the "From" field in the INVITE message contains a PSTN number that they have 'associated' with the SIP trunking service, e.g. From: "Zoltan" <sip:02031375892@79.125.110.116> is OK. From: "Zoltan" <sip:anonymous@79.125.110.116> would be rejected by them, as would From: "Zoltan" <sip:01344299800@79.125.110.116> as 01344299800 is 'someone elses' number, not one of theirs that they have allocated to us. The first call in the pastebin shows a Polycom phone making an outgoing call that works / is allowed as the effective_caller_id_number is set to an "allowed" number 02031375892. The second call, starting at line 871 shows a call originating in the PSTN from number 01344299800 to the Polycom. The SIP provider delivers the call to the FreeSwitch which in turn sends the call to the Polycom. The Polycom has been set to divert all incoming calls to number 07770380800 and replies with a "SIP/2.0 302 Moved Temporarily" with the forwarded to number as the contact. This in turn causes the FreeSwitch to generate an INVITE message which is sent to the trunk provider with: From: "01344299800" <sip:01344299800@79.125.110.116> and To: <sip:07770380800@79.125.4.174> Since 01344299800 is not an "allowed" number according to the provider, the call is rejected. Putting aside the argument for now of whether the SIP provider should be carrying out this rather brutal form of "CLI screening" (the data in the 'From' field appears as the A Party Number / CLI once the call hits the PSTN / mobile networks), is there any way to modify the behaviour of this call so that in response to the "SIP/2.0 302 Moved Temporarily" message, the INVITE which is sent out has the number from the effective_caller_id_number variable for the forwarding extension, rather than the number from the "From" field of the incoming invite? Hope this makes sense, Mike

2 1

mod_callcenter dropping call
by Troy Anderson 24 Sep '10

24 Sep '10

Hi, I'm trying to play around with mod_callcenter. The wiki is well written and thorough, except that I can't get it to work ;) Hopefully I'm just missing something simple. I have it set up with the sample callcenter.conf.xml from a fresh install. I added this to my default dialplan: <extension name="callcenter test"> <condition field="destination_number" expression="^11$" break="on-true"> <action application="answer"/> <action application="playback" data="ivr/ivr-hello.wav"/> <action application="callcenter" data="support@default"/> <action application="playback" data="misc/error.wav"/> </condition> </extension> When I dial 11 from a local phone, I hear "hello" and then "error" and in fs_cli, with logging set to 7, there is not one line of output related to callcenter except for the fact that it executed the above extension. It's as if the callcenter application never got executed. However, I can add and remove members, tiers, and agents to using callcenter_config. Is this the correct way to drop a call into a callcenter queue? Thanks for any help! Troy

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