December 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Oddities with Polycom's vesion 4.0.0
by Yehavi Bourvine 21 Feb '12

21 Feb '12

Hello, I've upgraded our Polycom phones from firmware 3.3.2 to 4.0.0 and have the following oddities: - On shared lines the call lists are "mixed": - No calls are in outgoing calls. - answered incoming and outgoing calls are placed in incoming calls. - unanswered incoming and outgoing calls are placed in the answered calls. - When the destination is busy there is no busy sign on the screen - the call is disconnected immediately and the phone returns to an idle state. I've opened today a support call with Polycom, and will update when I get replies. Regards, __Yehavi:

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Extending the field size of core's db_data table?
by Yehavi Bourvine 30 Jan '12

30 Jan '12

Hello, We use the DB api for storing and caching various data. We need to store data that is longer than 255 characters, which is the current limit on the field size there. The backend for this API is MySQL (via ODBC). Can I just increase the field size in MySQL, or is there some dependency in FreeSwitch on this size? Thanks, __Yehavi:

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Not recognizing voice in mod_pocketsphinx
by Thangappan.M 30 Jan '12

30 Jan '12

Dear all, I have installed the mod_pocketsphinx module successfully and enabled with FreeSWITCH. Copied the relevant files for pizza application and placed into appropriate directory. Made a call to "74992" and able to hear the voice. When I said "TAKEOUT' it was not recognizing my words. In the freeswitch CLI it shown the following messages. EXECUTE sofia/internal/1000(a)192.168.1.222 javascript(ps_pizza.js) 2010-09-06 13:18:48.023673 [DEBUG] sofia_glue.c:2594 AUDIO RTP [sofia/internal/1000(a)192.168.1.222] 192.168.1.222 port 32590 -> 192.168.6.60 port 8000 codec: 8 ms: 20 2010-09-06 13:18:48.024705 [DEBUG] switch_rtp.c:1182 Starting timer [soft] 160 bytes per 20ms 2010-09-06 13:18:48.025710 [DEBUG] sofia_glue.c:2774 Set 2833 dtmf send payload to 101 2010-09-06 13:18:48.026799 [DEBUG] sofia_glue.c:2779 Set 2833 dtmf receive payload to 101 2010-09-06 13:18:48.026799 [DEBUG] mod_sofia.c:636 Local SDP sofia/internal/ 1000(a)192.168.1.222: v=0 o=FreeSWITCH 1283726738 1283726739 IN IP4 192.168.1.222 s=FreeSWITCH c=IN IP4 192.168.1.222 t=0 0 m=audio 32590 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2010-09-06 13:18:48.028235 [DEBUG] switch_core_session.c:641 Send signal sofia/internal/1000(a)192.168.1.222 [BREAK] 2010-09-06 13:18:48.028235 [DEBUG] sofia.c:4153 Channel sofia/internal/ 1000(a)192.168.1.222 entering state [completed][200] 2010-09-06 13:18:48.029256 [NOTICE] mod_spidermonkey.c:2066 Channel [sofia/internal/1000(a)192.168.1.222] has been answered 2010-09-06 13:18:48.030269 [DEBUG] switch_ivr_play_say.c:1152 Codec Activated L16@8000hz 1 channels 20ms 2010-09-06 13:18:48.123555 [DEBUG] sofia.c:4153 Channel sofia/internal/ 1000(a)192.168.1.222 entering state [ready][200] 2010-09-06 13:18:48.171705 [DEBUG] switch_rtp.c:2066 Correct ip/port confirmed. 2010-09-06 13:18:50.212417 [DEBUG] switch_ivr_play_say.c:1444 done playing file EXECUTE sofia/internal/1000(a)192.168.1.222 detect_speech(pocketsphinx pizza_order undefined) 2010-09-06 13:18:51.768030 [DEBUG] switch_core_media_bug.c:360 Attaching BUG to sofia/internal/1000(a)192.168.1.222 EXECUTE sofia/internal/1000(a)192.168.1.222 detect_speech(resume) 2010-09-06 13:18:51.774074 [DEBUG] switch_ivr_play_say.c:1152 Codec Activated L16@8000hz 1 channels 20ms 2010-09-06 13:18:54.492023 [DEBUG] switch_ivr_play_say.c:1444 done playing file EXECUTE sofia/internal/1000(a)192.168.1.222 detect_speech(resume) EXECUTE sofia/internal/1000(a)192.168.1.222 detect_speech(resume) 2010-09-06 13:19:01.534474 [DEBUG] switch_ivr_play_say.c:1152 Codec Activated L16@8000hz 1 channels 20ms Please help me..... -- Regards, Thangappan.M

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int/ext dial tones not synchronous
by georg＠riseup.net 27 Jan '12

27 Jan '12

Hello all, I made the first experiences with the dialplan, and after some while, I really got calls from the outside routed to one of my sip phones. However, my phone is ringing something like 1-3 seconds earlier than the dial tone the caller is hearing. I googled around, but found no solution. As a start, I just used this extension: <extension name="Local_Extension"> <condition field="destination_number" expression="^(4930XXXXXXXX)$"> <action application="bridge" data="user/12(a)192.168.1.93"/> </condition> </extension> Thanks, Georg

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BLF gets confused with multiple calls
by Tim St. Pierre 17 Jan '12

17 Jan '12

Hello, I'm having all sorts of problems with BLFs not being in the correct state. It works fine in some cases, but others are wrong. Here's what I can see: Single endpoint, single call, everything works fine. Light flashes on ring, goes steady on answer, goes out on hangup. Here's where it gets tricky: If there are two phones registered to the extension, it flashes when they ring, but then goes out when one of them answers. If either phone places an outgoing call, the lamp comes on. If a single phone gets a second call, their lamp flashes again, but then goes out when they answer the second call. I'm constantly getting complaints from users where the lamps are stuck on. It happens more often when they have a lot of phones in ring groups. The lamps work fine for ring groups - they all flash, and whomever picks up the call stays steady while the rest go out. Is there anything I can do to get freeswitch to base the state on whether or not that user has any active calls, rather than just what the last thing that the phone did was? This happens on every model of Aastra phone, and I have all of them. I haven't had a chance to try it yet on Polycom. I'm running FreeSWITCH Version 1.0.head (git-7531fed 2011-08-17 11-27-20 -0500) Any ideas would be more than welcome. -Tim

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audo sync issues with record_session to mp3
by Tom Parrott 13 Jan '12

13 Jan '12

Hi, We are running a cluster of 3 SIP servers all running Freeswitch. Each call is recorded using the record_session application to record to mp3 files using mod_shout. Longer calls, after about 10 minutes start to introduce sync issues between the A-leg and the B-leg. We are running record_session on the A-leg, and it seems to get ahead of the B-leg. For example the caller on the A-leg will be heard to answer a question whilst the person on the B-leg is asking it. I am aware that recording to .mp3 can use additional resources, but these are quad core servers with 4GB of ram and they are only doing around 10 concurrent calls each. Additionally I have seen others experiencing these issues with Wav files too: http://lists.freeswitch.org/pipermail/freeswitch-users/2010-July/060107.html I am using the git head version from Tue 11 Oct 2011. I have also tried turning off enable_file_write_buffering on the A-leg however this makes no difference either. Does anyone have any suggestions? Thanks

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Overlap dialing problem
by Štefan Čudai 10 Jan '12

10 Jan '12

Hi, does FreeSwitch support for overlap dialing or did somebody try to use it? I am using Sangoma card A102 under Windows 7 with FreeTDM. I have configured my freetdm.conf.xml with this two lines: <param name="overlap" value="yes"/> <param name="min_digits" value="0"/> But when I'm dialing number (which has 14 digits) I hear disconnect tone after dialing first 5 digits. I think that reason is because FreeTDM module is waiting number with minimum digits equals to "0" (see my min_digits). I have also try set min_digits to 20 and then dial same number with 14 digits but result was same. So, I looked to log file and found that FreeTDM module got in thist case STATUS CONFIRM message but he was still waiting for next digits and his time expired. Can you help me please? Thanks a lot. Stefan

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Help on RTMP
by RuiQiang Huang 06 Jan '12

06 Jan '12

Hi, I'm trying to add RTMP to my freeswitch server. I compiled mod_rtmp.so, mod_rtmp.la and put it under freeswitch/mod/. My freeswitch version is FreeSWITCH version: 1.0.head (git-c8c94f0 2011-08-19 11-52-40 -0500) My rtmp.conf.xml is <configuration name="rtmp.conf" description="RTMP Endpoint"> <profiles> <profile name="default"> <settings> <param name="bind-address" value="0.0.0.0:1935" /> <param name="context" value="public" /> <param name="dialplan" value="XML" />  <param name="auth-calls" value="true" />  <param name="buffer-len" value="50" />  <param name="chunksize" value="512" /> </settings> </profile> </profiles> </configuration> However, I didn't see freeswitch listening on 1935 port and in fs_cli when I run freeswitch > rtmp status It's showing unknown command. My question is, is there anything else needed to get RTMP working? How to know which modules are loaded? Thanks.

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How to troubleshoot "503 Maximum Calls In Progress"?
by Chris Fowler 05 Jan '12

05 Jan '12

Hi, On our Production FreeSWITCH box: FreeSWITCH Version 1.0.head (git-4c435ec 2011-03-14 11-54-08 -0500) UP 0 years, 14 days, 6 hours, 28 minutes, 44 seconds, 474 milliseconds, 751 microseconds 12519 session(s) since startup 2 session(s) 0/30 1000 session(s) max min idle cpu 0.00/100.00 At 14:30:37 a reloadxml command was issued. At 14:30:43 the box started rejecting all calls with 503 error. At 14:38 it started working again. Looking at the code I see three conditions can trigger this behavior: if (sess_count >= sess_max || !sofia_test_pflag(profile, PFLAG_RUNNING) || !switch_core_ready()) { nua_respond(nh, 503, "Maximum Calls In Progress", SIPTAG_RETRY_AFTER_STR("300"), TAG_END()); switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "No more sessions allowed at this time.\n"); goto done; } Logs show it wasn't lack of available sessions. How can sofia_test_pflag(profile, PFLAG_RUNNING) or switch_core_ready() fail and busy the system? Thoughts? Thx, Chris.

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Sangoma A200 FXO Outgoing Problem
by Ryan V 05 Jan '12

05 Jan '12

Hi, I have configured A200 and able to receive incoming calls. I am unable to dial out. Here are the logs. 2011-12-08 16:39:15.990089 [DEBUG] ftmod_analog.c:62 [s3c1][2:1] Changed state from DOWN to DIALING 2011-12-08 16:39:15.990089 [DEBUG] ftmod_analog.c:439 [s3c1][2:1] ANALOG CHANNEL thread starting. 2011-12-08 16:39:15.990089 [DEBUG] ftmod_analog.c:459 [s3c1][2:1] Initialized DTMF detection 2011-12-08 16:39:15.990089 [DEBUG] ftmod_analog.c:640 [s3c1][2:1] Completed state change from DOWN to DIALING in 0ms 2011-12-08 16:39:15.990089 [DEBUG] ftmod_analog.c:646 [s3c1][2:1] Executing state handler on 3:1 for DIALING 2011-12-08 16:39:21.009503 [DEBUG] ftmod_analog.c:505 [s3c1][2:1] Changed state from DIALING to BUSY 2011-12-08 16:39:21.030044 [DEBUG] ftmod_analog.c:640 [s3c1][2:1] Completed state change from DIALING to BUSY in 20ms 2011-12-08 16:39:21.030044 [DEBUG] ftmod_analog.c:646 [s3c1][2:1] Executing state handler on 3:1 for BUSY 2011-12-08 16:39:21.030044 [DEBUG] ftmod_analog.c:802 [s3c1][2:1] Changed state from BUSY to DOWN 2011-12-08 16:39:21.049928 [DEBUG] ftmod_analog.c:640 [s3c1][2:1] Completed state change from BUSY to DOWN in 20ms 2011-12-08 16:39:21.049928 [DEBUG] ftmod_analog.c:646 [s3c1][2:1] Executing state handler on 3:1 for DOWN 2011-12-08 16:39:21.049928 [DEBUG] ftmod_analog.c:944 [s3c1][2:1] Going onhook2011-12-08 16:39:21.049928 [DEBUG] switch_core_state_machine.c:362 (FreeTDM/3:1/9449905000) Running State Change CS_HANGUP Here is our dial plan <extension name="outgoing_bsnl"> <condition field="destination_number" expression="^9(\d{10})$"> <action application="bridge" data="freetdm/3/a/$1"/> </condition> </extension> I suspect FS is not detecting the dial tone on analogue channel. BTW, outgoing on our digital span works fine. Any suggestions? Thanks, Ryan

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