July 2013 - freeswitch-dev@lists.freeswitch.org

[Freeswitch-dev] 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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[Freeswitch-dev] 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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Re: [Freeswitch-dev] [Freeswitch-users] origin/master's HEAD is segfaulting
by Ken Rice 05 Jul '13

05 Jul '13

Oh come on people... Jira... That's where you report bugs... And yes a segfault is a bug On 6/22/13 9:03 AM, "arif hossain" <aftnix(a)live.com> wrote: > 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 > _________________________________________________________________________ > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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[Freeswitch-dev] 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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[Freeswitch-dev] 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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[Freeswitch-dev] Error building RPM
by Worker Bee 05 Jul '13

05 Jul '13

Hello Everyone; I am new to FreeSwitch. I am attempting to create an rpm distribtion on CentOS6, Freeswitch version 1.2.9 I keep running into the following error: libtool: link: `/usr/lib64/libogg.la' is not a valid libtool archive I would be very appreciative if anyone could shed some light or point me in the right direction. Thanks! Bee

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[Freeswitch-dev] Question about placing call needed time
by Georgi Stefanov 05 Jul '13

05 Jul '13

Hello All, I have a small question about time needed to place call initiated with originate command. I hope someone to give me some clues, because I am stucked Here is the test FreeSwitch2 dialplan: public: blind transfer to default <extension name="public_did"> <condition field="destination_number" expression=""> <action application="set" data="domain_name=$${domain}"/> <action application="bridge" data="user/${destination_number}"/> </condition> </extension> default: just play a pretty music <extension name="Local_Extension"> <condition field="destination_number" expression="^(10[01][0-9])$"> <action application="export" data="dialed_extension=$1"/> <action application="answer"/> <action application="playback" data="/usr/local/freeswitch/sounds/music/8000/ponce-preludio-in-e-major.wav"/> </condition> </extension> FreeSwitch1 Using mod_event_socket I connect to FS1 and pass 20 commands like (20 connections): originating from FS1 to user to FS2 - I am using "originate sofia/internal/1000(a)10.101.3.60 6001" for user 1000 til 1020 As result I have 20 calls in FS2 This is cool for now :-) Here comes the question arised from looking FS2 logs: root@GeorgiStefanov:/usr/local/freeswitch/log# grep "NEW RTP SESSION CREATING" freeswitch.log 2013-07-05 17:46:34.897918 [CRIT] switch_rtp.c:1907 NEW RTP SESSION CREATING 2013-07-05 17:46:34.897918 [CRIT] switch_rtp.c:1907 NEW RTP SESSION CREATING 2013-07-05 17:46:34.897918 [CRIT] switch_rtp.c:1907 NEW RTP SESSION CREATING 2013-07-05 17:46:34.917913 [CRIT] switch_rtp.c:1907 NEW RTP SESSION CREATING 2013-07-05 17:46:34.917913 [CRIT] switch_rtp.c:1907 NEW RTP SESSION CREATING 2013-07-05 17:46:34.917913 [CRIT] switch_rtp.c:1907 NEW RTP SESSION CREATING 2013-07-05 17:46:34.917913 [CRIT] switch_rtp.c:1907 NEW RTP SESSION CREATING 2013-07-05 17:46:34.937911 [CRIT] switch_rtp.c:1907 NEW RTP SESSION CREATING 2013-07-05 17:46:34.937911 [CRIT] switch_rtp.c:1907 NEW RTP SESSION CREATING 2013-07-05 17:46:34.937911 [CRIT] switch_rtp.c:1907 NEW RTP SESSION CREATING 2013-07-05 17:46:34.937911 [CRIT] switch_rtp.c:1907 NEW RTP SESSION CREATING 2013-07-05 17:46:34.937911 [CRIT] switch_rtp.c:1907 NEW RTP SESSION CREATING 2013-07-05 17:46:34.937911 [CRIT] switch_rtp.c:1907 NEW RTP SESSION CREATING 2013-07-05 17:46:34.937911 [CRIT] switch_rtp.c:1907 NEW RTP SESSION CREATING 2013-07-05 17:46:34.937911 [CRIT] switch_rtp.c:1907 NEW RTP SESSION CREATING 2013-07-05 17:46:34.937911 [CRIT] switch_rtp.c:1907 NEW RTP SESSION CREATING 2013-07-05 17:46:34.937911 [CRIT] switch_rtp.c:1907 NEW RTP SESSION CREATING 2013-07-05 17:46:34.957911 [CRIT] switch_rtp.c:1907 NEW RTP SESSION CREATING 2013-07-05 17:46:34.957911 [CRIT] switch_rtp.c:1907 NEW RTP SESSION CREATING 2013-07-05 17:46:34.957911 [CRIT] switch_rtp.c:1907 NEW RTP SESSION CREATING root@GeorgiStefanov:/usr/local/freeswitch/log# grep "NEW RTP SESSION CREATING" freeswitch.log | wc -l 20 root@GeorgiStefanov:/usr/local/freeswitch/log# grep " Correct ip/port confirmed" freeswitch.log 2013-07-05 17:46:36.857908 [DEBUG] switch_rtp.c:3706 Correct ip/port confirmed. 2013-07-05 17:46:36.857908 [DEBUG] switch_rtp.c:3706 Correct ip/port confirmed. 2013-07-05 17:46:38.077915 [DEBUG] switch_rtp.c:3706 Correct ip/port confirmed. 2013-07-05 17:46:38.077915 [DEBUG] switch_rtp.c:3706 Correct ip/port confirmed. 2013-07-05 17:46:39.317915 [DEBUG] switch_rtp.c:3706 Correct ip/port confirmed. 2013-07-05 17:46:39.317915 [DEBUG] switch_rtp.c:3706 Correct ip/port confirmed. 2013-07-05 17:46:40.557916 [DEBUG] switch_rtp.c:3706 Correct ip/port confirmed. 2013-07-05 17:46:40.557916 [DEBUG] switch_rtp.c:3706 Correct ip/port confirmed. 2013-07-05 17:46:41.777917 [DEBUG] switch_rtp.c:3706 Correct ip/port confirmed. 2013-07-05 17:46:41.777917 [DEBUG] switch_rtp.c:3706 Correct ip/port confirmed. 2013-07-05 17:46:42.977908 [DEBUG] switch_rtp.c:3706 Correct ip/port confirmed. 2013-07-05 17:46:42.977908 [DEBUG] switch_rtp.c:3706 Correct ip/port confirmed. 2013-07-05 17:46:43.617914 [DEBUG] switch_rtp.c:3706 Correct ip/port confirmed. 2013-07-05 17:46:43.617914 [DEBUG] switch_rtp.c:3706 Correct ip/port confirmed. 2013-07-05 17:46:45.477914 [DEBUG] switch_rtp.c:3706 Correct ip/port confirmed. 2013-07-05 17:46:45.477914 [DEBUG] switch_rtp.c:3706 Correct ip/port confirmed. 2013-07-05 17:46:46.657910 [DEBUG] switch_rtp.c:3706 Correct ip/port confirmed. 2013-07-05 17:46:46.677908 [DEBUG] switch_rtp.c:3706 Correct ip/port confirmed. 2013-07-05 17:46:47.297931 [DEBUG] switch_rtp.c:3706 Correct ip/port confirmed. 2013-07-05 17:46:47.297931 [DEBUG] switch_rtp.c:3706 Correct ip/port confirmed. root@GeorgiStefanov:/usr/local/freeswitch/log# grep " Correct ip/port confirmed" freeswitch.log | wc -l 20 I have modified switch_rtp.c I have added log "NEW RTP SESSION CREATING" in very begging of function switch_rtp_create - just want to know when media connection preparation is started Does anybody can help me to understand the debug "Correct ip/port confirmed." info. I am worried about the not so much of these 10 sec latency, but the sequence I thought they should be paralel, not one after another. What is FS saying with "Correct ip/port confirmed."? Does FS respond in paralel to all calls?

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[Freeswitch-dev] Developing a new module
by Lists 02 Jul '13

02 Jul '13

Hi, I posted on the main list asking how to implement a modem using FreeSwitch. I had some helpful replies and I have since found that various existing modules include the bulk of what I need, in particular mod_fsk. I need to program remote analog phone devices that are programmed using data sent by V.23 and dtmf, similar to that used in the FSK caller id protocol. I decided to write a new module and took the mod_skel as recommended in the Authoring modules section of the documentation. I have that compiled and running as a skin for my app. My problem is that the documentation is somewhat fragmented and incomplete for a newbie like me, I'm really struggling to understand all of the complex issues around writing apps that use switch core functions and deal with rtp packets in real time. I'm part of a small startup team and would appreciate any help in getting up to speed quickly with this as we want to launch our new product soon. Can anyone recommend the best way to learn, other than scouring through other peoples code, I would be happy to pay for some support from another developer who could guide me along. I feel capable I will get there eventually, but I'm not sure I have the time to work it all out the hard way! All the best, paul(a)collisters.com lists(a)collisters.com Liverpool, England

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[Freeswitch-dev] FreeSWITCH Weekly News and Notes
by Michael Collins 02 Jul '13

02 Jul '13

Welcome to July! We've had lot's of great news<http://online.wsj.com/article/PR-CO-20130624-907435.html> lately with FreeSWITCH and now there's even more. Our friends at Yealink have reported on a very large VoIP installation in Spain that utilizes Yealink phones and FreeSWITCH servers. The complete writeup can be found here <http://www.yealink.com/News_info.aspx?NewsId=330&CateId=306>. We look forward to reporting on more success stories in the future. With the recent announcement <http://freeswitch.org/node/455> about FreeSWITCH 1.4 beta and WebRTC support there has been a lot of interest in just how to make this all work. On this week's conference call<http://wiki.freeswitch.org/wiki/FS_weekly_2013_07_03> we will be taking a closer look at the code and showing you how to setup both sides of a WebRTC connection with FreeSWITCH. Please join usthis Wednesday at 1PM Eastern, 10AM Pacific for this interesting discussion. We also had an interesting discussion on last week's conference call<http://wiki.freeswitch.org/wiki/FS_weekly_2013_06_26>. Lorenzo Magani and Alexandr Dubovikov from the HOMER<http://www.sipcapture.org/> project gave us a tour of their new version 3.5 release. The HOMER SIP capture system is an excellent - and free - tool that can be put into production with any FreeSWITCH server. HOMER is an invaluable tool used for SIP diagnostics and troubleshooting and we highly recommend it for anyone using FreeSWITCH and SIP in a production environment. Have a great week and we'll talk to you on Wednesday. -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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