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Question about placing call needed time
by Georgi Stefanov 11 Jul '13

11 Jul '13

Hello All, I have made some other experiments. I have added mod_event socket to the extension used in originate command <action application="answer"/> <action application="socket" data="127.0.0.1:8084 async full"/> On the other side of socket I log the time the call is answered. Even I have simultaneous execution of "originate" command, I have sequential socket opening Does anybody have any clue ?

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FreeSWITCH Community Conference Call Today
by Michael Collins 10 Jul '13

10 Jul '13

Hello all, Today's conference call agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2013_07_10 We are going to be discussing several topics: * ClueCon 2013 <http://www.cluecon.com> - register now and triple your chances to win * New FreeSWITCH wiki site under development * Community discussion of some mailing list questions, like the broker's call<http://lists.freeswitch.org/pipermail/freeswitch-users/2013-July/097491.html>feature Talk to you soon! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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FreeSWITCH Weekly News and Notes
by Michael Collins 08 Jul '13

08 Jul '13

Greetings FreeSWITCH users! WebRTC continues to be a hot topic. We've had a lot of interest in the subject since the announcement <http://freeswitch.org/node/455> of FreeSWITCH 1.4 beta. On last week's conference call<http://wiki.freeswitch.org/wiki/FS_weekly_2013_07_03>we had Ken Rice show us just how easy it is to get a WebRTC-enable page up and running very easily using sipML5 <http://www.sipml5.org> or JsSIP<http://www.jssip.net>. FreeSWITCH users are also welcome to view our WebRTC page's<https://webrtc.freeswitch.org/webrtc/portal.html>underlying code for inspiration. This week we will be doing a community discussion<http://wiki.freeswitch.org/wiki/FS_weekly_2013_07_10>highlighting some recent mailing list topics. We will also be entertaining questions from the audience, so if you have have a FreeSWITCH question please join us on Wednesday. In ClueCon news we are happy to let you know that everyone who registers by Friday, July 12th will triple their chances of winning some great prizes, including the 10 Android tablets that will be given away this year. Simply click the "Register Now" button on the ClueCon site <http://www.cluecon.com>to complete the registration process. Also, plan on being in Chicago on Monday August 5th so that you can hang out with the team and meet other ClueCon attendees. Can you believe that ClueCon 2013 is less than a month away? Have a great week and we will talk to you on Wednesday. -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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sofia profile stop response
by Xijing Dai 06 Jul '13

06 Jul '13

hey all, I have experienced a lot of times that sofia profile stops to response for any sip messages. And when you type sofia status, that profile is gone, but if we check the port by netstat, the port is still opened by freeswitch. This happens if we keep freeswitch running in 1 - 2 days. And there is no error log. Any idea on how to debug? Cheers.

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ClueCon 2013 - Register Right Away to Increase Your Chances of Winning
by Michael Collins 05 Jul '13

05 Jul '13

ClueCon 2013 - Register Today! At ClueCon we love giving stuff away. Our sponsors have given us many cool toys to give away this year. We also have books from Packt Publishing. To top it all off we will once again be giving away ten Android 4 tablets! You can increase your chances of winning by registering right away. Everyone who registers by Friday July 12 will *triple* their chances of winning, so don't delay! You can register online right now. Just visit the ClueCon <https://www.cluecon.com/2013/> site and click the "Register 2013" link. To book your hotel room online, simply click the "Book Your Room" link. Rooms are going fast so don't wait! Great Presentations: WebRTC, VoIP Security, and More 2013 will be the best ClueCon yet. Check out this year's schedule<https://www.cluecon.com/schedule/>and you'll see some familiar favorites as well as new faces. WebRTC is a hot topic, as is security. We are happy to let you know that Phil Zimmermann from Silent Circle will be with us again. Be sure to attend the VoIP Security meetup on Wednesday evening. Feel free to contact us if you have any questions about this year's event. We look forward to seeing everyone in Chicago next month! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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mod_callcenter makes me BREAK_OUT
by Sayyed Mohammad Emami Razavi 05 Jul '13

05 Jul '13

After configuring agents and queues, when i call and join to the queue, call center makes me BREAK_OUT Can any body guess my problem?! any other log file you want to solve the problem? FS logs pasted bin http://pastebin.freeswitch.org/21144

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Video input for FreeSwitch
by Thorhallur Sverrisson 05 Jul '13

05 Jul '13

Hi All, I've been working on setting up FreeSWITCH on my Raspberry PI as a replacement for my aging front door intercom. I have most of the quirks worked out using PortAudio for the audio and some shell script magic to handle the button input. As this is project is nearing its end I'ld like to add on the camera module for rpi and get streaming video from the front door. I know FS has h.264 passthrough, and I've seen some mp4 stuff showing up in the source tree although I have not dug too deep into that yet. I'm wondering if anyone has looked into this before, or if my idea is too wild to even consider. The rpi has a nice and simple command line program to interface to the camera, so 'all' that would be needed, if I understand correctly, is to execute the program with the right parameters and stream the standard output to the remote end. Any information or tips would be highly appreciated, Thor

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origin/master's HEAD is segfaulting
by arif hossain 05 Jul '13

05 Jul '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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Wrong match between codecs offered by caller UAC and enabled codecs when multiple sample rates are in use
by Giacomo Vacca 05 Jul '13

05 Jul '13

Hi all, I've seen a behaviour similar to http://jira.freeswitch.org/browse/FS-5398and I have a proposed patch that fixes it. Scenario: - Caller offers, in order, AMR-WB, iLBC, SILK/12K, SILK/24K, SILK/8, speex/8 - FS has global_codec_prefs set to SILK@24000h@20i,SILK@16000h @20i,SILK@8000h@20i,speex@16000h@20i,speex@8000h@20i,PCMU,PC MA - FS decides to accept with SILK/16K (not even offered): the caller BYEs immediately FS logged: 7dc4b656-a7cf-46fe-93f0-b10b497a1f7d 2013-06-27 14:17:30.726808 [DEBUG] switch_core_media.c:2662 Audio Codec Compare [SILK:97:12000:20:0]/[SILK:120:24000:20:40000] 7dc4b656-a7cf-46fe-93f0-b10b497a1f7d 2013-06-27 14:17:30.726808 [DEBUG] switch_core_media.c:2662 Audio Codec Compare [SILK:97:12000:20:0]/[SILK:119:16000:20:30000] 7dc4b656-a7cf-46fe-93f0-b10b497a1f7d 2013-06-27 14:17:30.726808 [DEBUG] switch_core_media.c:1772 Set Codec sofia/internal/4656729(a)truphone.comSILK/16000 20 ms 320 samples 30000 bits To fix this I've added the comparison of sample rates in switch_core_media.c, with this patch (applied on latest master): diff --git a/src/switch_core_media.c b/src/switch_core_media.c index 9270b0b..4b926ed 100644 --- a/src/switch_core_media.c +++ b/src/switch_core_media.c @@ -2788,7 +2788,7 @@ SWITCH_DECLARE(uint8_t) switch_core_media_negotiate_sdp(switch_core_session_t *s if ((zstr(map->rm_encoding) || (smh->mparams->ndlb & SM_NDLB_ALLOW_BAD_IANANAME)) && map->rm_pt < 96) { match = (map->rm_pt == imp->ianacode) ? 1 : 0; } else { - match = strcasecmp(rm_encoding, imp->iananame) ? 0 : 1; + match = (!strcasecmp(rm_encoding, imp->iananame) && (map->rm_rate == codec_rate)) ? 1 : 0 } Hope this helps, even if a better solution exists. Regards, Giacomo

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Help : Freeswitch Conference call
by Dona Liz Peter 05 Jul '13

05 Jul '13

Hi everyone , I am new to this field. Just started trying FreeSWITCH for my project. How can I implement the idea suggested here in a dialplan? Can you please give an example dialplan or code? Is it possible to mute a member in a conference using configuration ? Please give me an example. Thanks in advance, Regards, Peter -- *Mit freundlichen Grüßen,* * * *Peter.*

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