June 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_conference noise?
by Markus von Arx 24 Jun '13

24 Jun '13

Hi I'm trying to set up audio conferences with mod_conference. Functionality works fine, but I have a problem with the audio quality of the audio conferences. If a SIP phone enters a conference as the first participant, FreeSwitch sends back a very noisy audio to that phone. As soon as a 2nd SIP phone joins the same conference, the noise goes away. The same noise is there if all conference members are muted. I tried to disable all kinds of comfort noise (e.g. comfort-noise = -1 in conference.conf.xml) etc., but cannot get rid of that noise. My setup: - FreeSwitch 1.2.10, Windows, x86 compilation. - SIP calls with uLAW, ptime 20. - disabled all wave files in conference.conf.xml Any ideas, how to get rid of that noise? Thanks, Markus

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Polish language support - mod_say_pl
by Mariusz Czulada 24 Jun '13

24 Jun '13

Hello all/Czołem, ----------------- As you may notice, I added polish language module mod_say_pl to master (it is also in v1.2.stable branch). If you have any comments regarding this module please post to freeswitch-users list. Of course any "say" module without sound files is almost useless. That's why I also prepared archives with 8k and 16k rate, available on http://files.freeswitch.org/. I created them with espeak (http://espeak.sourceforge.net/) In mod_say_pl source directory there is a bash script I used. In conf/dialplan/default there is also a file (test_say_pl.xml). You can use it to check diffent scenarios: numbers, dates and IP address, by dialing 44xx extensions. ----------------- Jak być może zauważyliście, popełniłem moduł "mod_say_pl". Przygotowałem również pliki dźwiękowe z pomocą espeak'a. Jakość głosu nie jest oczywiście porównywalna np. z Ivoną, tym nie mniej sądzę że są wystarczająco dobre do celów rozwojowych/testowych. Na obecnym etapie moduł powinien prawidłowo czytać liczebniki główne i prządkowe, literować liczby, czytać daty (również w wersji skróconej) oraz adresy IP. Uwagi odnośnie modułu mile widziane - przez listę, zaś dostrzeżone błedy i poprawki - standardowo przez jirę. ----------------- Regards/Pozdrawiam, Mariusz

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origin/master's HEAD is segfaulting
by arif hossain 24 Jun '13

24 Jun '13

Hi, I'm on this commit : arif@khost:~/src/freeswitch$ git show HEAD commit f8122e39c51240b99bf07a767afe2f8a698f1486 Now its segfaulting on startup. >From generated core : Program terminated with signal 11, Segmentation fault. #0 0x00007f245cc28df1 in su_casematch (s1=0x7f245cc7fc57 "", s2=0x3000000030 <Address 0x3000000030 out of bounds>) at su_string.c:224 224 unsigned char a = *s1++, b = *s2++; Clearly s2 is wrongly initialized. The backtrace : (gdb) bt #0 0x00007f245cc28df1 in su_casematch (s1=0x7f245cc7fc57 "", s2=0x3000000030 <Address 0x3000000030 out of bounds>) at su_string.c:224 #1 0x00007f245cc377ff in tport_server_addrinfo (mr=0x7f243c003c30, canon=0x7f243c003bfe "[::1]", family=0, host=0x7f245c958670 "::1", service=0x7f243c003c04 "5060", protocol=0x7f245cc7fc56 "*", transports=0x7f245c9589c0, return_addrinfo=0x7f245c958608) at tport.c:1814 #2 0x00007f245cc370a1 in tport_bind_server (mr=0x7f243c003c30, tpn=0x7f245c958770, transports=0x7f245c9589c0, public=tport_type_local, tags=0x7f245c958730) at tport.c:1652 #3 0x00007f245cc36762 in tport_tbind (self=0x7f243c003c30, tpn=0x7f245c958a50, transports=0x7f245c9589c0, tag=0x7f245cecae20 <tptag_ident>, value=139794152182707) at tport.c:1524 #4 0x00007f245cb8882d in nta_agent_add_tport (self=0x7f243c001e40, uri=0x7f24400099c0, tag=0x7f245cecae20 <tptag_ident>, value=139794152182707) at nta.c:2237 #5 0x00007f245cbc4e71 in nua_stack_init_transport (nua=0x7f2440007970, tags=0x7f24400098f0) at nua_register.c:1216 #6 0x00007f245cbd6b6b in nua_stack_init (root=0x7f243c001130, nua=0x7f2440007970) at nua_stack.c:194 #7 0x00007f245cc24441 in su_pthread_port_clone_main (varg=0x7f245c9d03d0) at su_pthread_port.c:340 #8 0x00007f2460ffae9a in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 ---Type <return> to continue, or q <return> to quit--- #9 0x00007f2460d27ccd in clone () from /lib/x86_64-linux-gnu/libc.so.6 #10 0x0000000000000000 in ?? () In tport_server_addrinfo() : (gdb) list tport_server_addrinfo ... 1800 char const * const transports[], ... ... 1811 for (i = 0, N = 0; transports[i] && N < TPORT_N; i++) { 1812 su_addrinfo_t *ai = &hints[N]; 1813 1814 if (!su_casematch(protocol, transports[i]) && !su_strmatch(protocol, "*")) So char const * const transports[] is not correctly initialized before using its values in the for loop. (gdb) p transports[0] $3 = 0x3000000030 <Address 0x3000000030 out of bounds> (gdb) p transports[1] $4 = 0x7f245c958d00 "`\215\225\\$\177" So transports[0] is the problematic initialization here. Going back further, in libs/sofia-sip/libsofia-sip-ua/nta/nta.c: 2150 char const * const * tports = tports_sip; And here (gdb) ptype tports_sip type = const char * const[5] (gdb) p tports_sip[0] $6 = 0x7f245cc7fc38 "udp" (gdb) p tports[0] $7 = 0x3000000030 <Address 0x3000000030 out of bounds> (gdb) ptype tports_sip type = const char * const[5] (gdb) p tports $11 = (const char * const *) 0x7f245c9589c0 (gdb) p &tports_sip $12 = (const char * const(*)[5]) 0x7f245cebf660 <tports_sip> As it can be seen, tports diverged from initial tports_sip. It has been done in following snippet : 2197 if (strchr(tp, ',')) { 2198 int i; char *t, *tps[9] = {0}; 2199 2200 /* Split tp into transports */ 2201 for (i = 0, t = tp; t && i < 8; i++) { 2202 tps[i] = t; 2203 if ((t = strchr(t, ','))) 2204 *t++ = '\0'; 2205 } 2206 2207 tps[i] = NULL; 2208 tports = (char const * const *)tps; Now : (gdb) p tp $20 = "udp\000tcp\000H\215\225\\$\177\000\000\240\252\354\\".. So above code should parse "udp" in tports[0]. Which is not the case here. I'm assuming this bug has to generated from a more higher level, rather than happened in a file which has not been touched since 2006( I git blamed it). But bt does not have any calls to "non-sofia" stuff. This su_casematch() was called from two threads, 4 Thread 0x7f245eccb700 (LWP 7622) 0x00007f245cc28df1 in su_casematch ( s1=0x7f245cc7fc57 "", s2=0x3000000030 <Address 0x3000000030 out of bounds>) at su_string.c:224 * 1 Thread 0x7f245c959700 (LWP 7623) 0x00007f245cc28df1 in su_casematch ( s1=0x7f245cc7fc57 "", s2=0x3000000030 <Address 0x3000000030 out of bounds>) at su_string.c:224 Both calls to su_casematch seems problematic. Sorry for a long BUGREPORT. Actually i thought i should post my investigation if it helps the devs(After git blaming, i highly suspect that my investigation unearthed nothing , problem generated from elsewhere, which i could not find). I've not started git bisecting. If it helps, i will be happy to do that. -- Cheers arif

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remove in local SDP word FREESWITCH
by Abdullah 24 Jun '13

24 Jun '13

HI ALL , please help me ,how to change or remove o=*"FreeSWITCH"* in free switch Cli Log . any idea ?? o=FreeSWITCH 1369449071 1369449072 IN IP4 10.10.50.1 s=FreeSWITCH c=IN IP4 10.10.50.1 many thanks -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/remove-in-local-SDP-word-FREE… Sent from the freeswitch-users mailing list archive at Nabble.com.

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help on mod_callcenter
by Ashish gautam 24 Jun '13

24 Jun '13

Hi, I am working with mod_callcenter. I want FS to send event data to an external script on two events when they happen. first when a caller is just answered and is in waiting state. Second, when he is connected to any agent. I have written this line in the dialplan: <action application="socket" data="127.0.0.1:8022 async full"/> is it going to do something? Please throw some light on it. -- Ashish Gautam

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Attended transfers failing - need help
by Garey Arrington 24 Jun '13

24 Jun '13

I am having an issue with attended transfers failing, it seems that the server is trying to send the invite after the REFER to itself. Almost as if it is resolving the registration/location domain (a.domain1.com in this case) and sending the invite to that IP. SIP trace available here: http://pastebin.com/dyvYpZNc I have googled this and found several others having this issue in the past but no resolution was posted afterwards. Please help. :)

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Request to change vim modelines in freeswitch sources
by Nathan Neulinger 23 Jun '13

23 Jun '13

I'd like to change the editor modelines in freeswitch source files from: * For VIM: * vim:set softtabstop=4 shiftwidth=4 tabstop=4: */ to: * For VIM: * vim:set softtabstop=4 shiftwidth=4 tabstop=4 noexpandtab: */ This will help formatting if an existing vimrc is setting expandtab by default. Since these appear to be consistent across most of the codebase, figured it would make more sense to do with a global search and replace. Would this be an acceptable change? -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

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m=audio 5016 RTP/AVP 96 3 107 110 0 8 101., what does it mean?
by Sayyed Mohammad Emami Razavi 23 Jun '13

23 Jun '13

m=audio 5016 RTP/AVP 96 3 107 110 0 8 101. Is there any digits represents sip codecs in subject?! Can any body describe this statement?! For example in response to one of my questions, Steven told me: "Your phone is not offering any codecs in the INVITE: m=audio 15696 RTP/AVP 19 The only payload type listed is 19 (comfort noise) - there are no codecs listed." base on my logs: http://pastebin.freeswitch.org/21084

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weird behavior when originating call with 3cx
by Sayyed Mohammad Emami Razavi 23 Jun '13

23 Jun '13

When i send call to one user that register with 3cx, 3cx response unalocate number. I can listen to IVR with 3cx when call IVR extension but when i want to originate one call with user at 3cx i can not. All tcp logs attached and FS logs are here: http://pastebin.freeswitch.org/21095

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opensips route one registration to two FS
by ZhouJianhua 22 Jun '13

22 Jun '13

Hi all, OpenSIPs is used for load balancing in from of two FS. And in some cases, there are more than one item inFS's "sip_registrations" table for one sip client. Then, the execution of "originate sofia/internal/SIP_ACCOUNT%FS_IP", will fail if FS_IP is wrong. I use the "expires" field in table of "sip_registrations" to figure out which is the newest FS_IP for the sip client, but sometimes it fails. I use the config file from http://wiki.freeswitch.org/wiki/OpenSIPS_configuration_for_2_or_more_FreeSW… Any suggestions will be appreciated!

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