November 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Spandsp tone_detect segment frequencies
by Nandy Dagondon 09 Nov '13

09 Nov '13

Hello everyone, I'm trying to fine tune call progress tone using tone_detect_start. I set spandsp.conf.xml: <descriptions log-level="2"> Log shows these lines: Tone segment f1=-1, f2=-1, duration=336ms How can I tell which frequency represents -1, 4 or 5? What is the frequency assignment of silence? Let's use North America as an example. /Nandy  <descriptor name="1"> <tone name="CED_TONE" description="ANS / ANSam"> <element freq1="2100" freq2="0" min="700" max="0"/> </tone> <tone name="SIT" description="Special Information Tone"> <element freq1="950" freq2="0" min="256" max="400"/> <element freq1="1400" freq2="0" min="256" max="400"/> <element freq1="1800" freq2="0" min="256" max="400"/> </tone> <tone name="RING_TONE" description="North America ring"> <element freq1="440" freq2="480" min="1200" max="0"/> </tone> <tone name="REORDER_TONE" description="North America reorder"> <element freq1="480" freq2="620" min="224" max="316"/> <element freq1="0" freq2="0" min="168" max="352"/> <element freq1="480" freq2="620" min="224" max="316"/> </tone> <tone name="BUSY_TONE" description="North America busy"> <element freq1="480" freq2="620" min="464" max="536"/> <element freq1="0" freq2="0" min="464" max="572"/> <element freq1="480" freq2="620" min="464" max="536"/> </tone> </descriptor>

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Re: [Freeswitch-users] Modify INVITE
by Aziz Bayd 09 Nov '13

09 Nov '13

2013/11/9, Aziz Bayd <bayd80(a)gmail.com>: > Hello. > > I have a question about FreeSWITCH configuration. > Can I edit or modify the INVITE message content? For example, remove > "To" or "Via" fields in the INVITE content. > If yes, how can I do? > Thanks. >

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Re: [Freeswitch-users] Errors during compiling from today's 1.2.stable build
by bedgar＠vseinc.com 09 Nov '13

09 Nov '13

Ken and Anthony, This is on a physical server that was already successfully running 1.2.14. It is CentOS 6.4. It was a fresh git to an empty directory. Only special thing was the configure requirement that I always do for CentOS. Brian Brian Sent from my Galaxy S®III -------- Original message -------- From: Ken Rice <krice(a)freeswitch.org> Date: 11/08/2013 6:29 PM (GMT-05:00) To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Subject: Re: [Freeswitch-users] Errors during compiling from today's 1.2.stable build What platform is this on? On 11/8/13 5:57 PM, "bedgar(a)vseinc.com" <bedgar(a)vseinc.com> wrote: I had trouble building from a git of 1.2.stable this afternoon. bootstrap successful configure successful make – not so good /bin/sh: fork: retry: Resource temporarily unavailable /bin/sh: fork: retry: Resource temporarily unavailable /bin/sh: fork: retry: Resource temporarily unavailable /bin/sh: fork: retry: Resource temporarily unavailable /bin/sh: fork: Resource temporarily unavailable /bin/sh: fork: retry: Resource temporarily unavailable /bin/sh: fork: retry: Resource temporarily unavailable /bin/sh: fork: retry: Resource temporarily unavailable /bin/sh: fork: retry: Resource temporarily unavailable Had to kill the make with Ctrl-C make[392]: *** [/home/fsbuild/src/freeswitch/libs/apr-util/libaprutil-1.la] Interrupt make[391]: *** [core] Interrupt rsync’d from another server from a build that retrieved this morning and successfully installed after I did a git clean -d -f –x Successful - FreeSWITCH Version 1.2.14+git~20131108T143841Z~0339e80780~64bit (git 0339e80 2013-11-08 14:38:41Z 64bit) Did another git of the stable build and still had the same error. Did something get borked in stable today? Brian ________________________________ _________________________________________________________________________ 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 G+ ClueCon : http://fs0.us/cluecon-gplus FB ClueCon : http://fs0.us/cluecon-fb G+ FreeSwitch : http://fs0.us/freeswitch-gplus FB FreeSWITCH : http://fs0.us/freeswitch-fb Twitter : @FreeSWITCH_WIRE irc.freenode.net #freeswitch

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Audio Quality of recordings
by Scott Ross 09 Nov '13

09 Nov '13

I've developed a tool which leverages freeswitch event_socket to originate 2 calls (sofia/external/number@ip) whom then navigate an IVR, connect to the same conference bridge then take turns playing back a .wav while the partner call records channel input. The goal is to run the two recorded files against a PESQ library to monitor whether our conference bridges are performing as expected. Problem: I'm hearing poor channel quality (static, degradation) from the recordings and when I listen in on the calls while the test is running. My PESQ scores are coming out much lower than expected. Out of curiosity I tried asterisk and scores were about a point higher (2.9 vs 3.9) so I should be able to rule out the hardware/environment/destination lines I'm running it on. I need help improving freeswitch audio quality for my use-case. I know this is asked alot - I've read all the threads/wiki articles I can find and tried: loading only necessary modules, running a single codec (ulaw), I've tried ensuring my playback reference is also ulaw format without headers, I've tried running on several different OSs: Ubuntu 11.04, RHEL 6.8, RHEL 5.6. Even originating on two separate sofia profiles (although I don't think this should have mattered regardless). I prefer freeswitch over asterisk in all other aspects but need to solve this (important) quality issue. Let me know what outputs I can provide.

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Freeswitch Segfaults
by JC 08 Nov '13

08 Nov '13

Hi everyone, We've been having some server crashes lately. We thought our most recent upgrade had fixed it, but unfortunately, it crashed again the other day. Based on the information below, does anyone have any recommendations or thoughts? Always seems to segfault. We are connected to it via RTMP from a Flash Media Server if that helps at all. > >> >> Crash this time: >> >> Core was generated by `/usr/local/freeswitch/bin/freeswitch -nc -hp'. >> Program terminated with signal 11, Segmentation fault. >> #0 rtmp_send_message (rsession=0x0, amfnumber=3 '\003', timestamp=0, type=20 '\024', stream_id=0, message=0x2adf53c96160 "\002", >> len=69, flags=0) at rtmp.c:562 >> 562 if ((rsession->send_ack + rsession->send_ack_window) < rsession->send && >> >> >> >> (gdb) bt >> #0 rtmp_send_message (rsession=0x0, amfnumber=3 '\003', timestamp=0, type=20 '\024', stream_id=0, message=0x2adf53c96160 "\002", >> len=69, flags=0) at rtmp.c:562 >> #1 0x00002adf4a14a0dd in rtmp_send_invoke_v (rsession=0x0, amfnumber=3 '\003', type=20 '\024', timestamp=0, stream_id=0, >> list=0x2adf53c969d0, freethem=SWITCH_TRUE) at rtmp.c:548 >> #2 0x00002adf4a14a2c8 in rtmp_send_invoke_free (rsession=0x0, amfnumber=3 '\003', timestamp=20, stream_id=0) at rtmp.c:507 >> #3 0x00002adf4a14a429 in rtmp_notify_call_state (session=<value optimized out>) at rtmp.c:484 >> #4 0x00002adf4a147073 in rtmp_receive_message (session=0x2adf4c33d5a8, msg=0x2adf53c96c90) >> at /usr/local/src/freeswitch/src/mod/endpoints/mod_rtmp/mod_rtmp.c:549 >> #5 0x00002adf42bc36cb in switch_core_session_perform_receive_message (session=0x2adf4c33d5a8, message=0x3, >> file=<value optimized out>, func=<value optimized out>, line=<value optimized out>) at src/switch_core_session.c:801 >> #6 0x00002adf42ba3bea in switch_channel_perform_pre_answer (channel=0x2adf4c343190, file=0x2adf42d075d8 "src/switch_core_session.c", >> func=0x2adf42d08400 "switch_core_session_execute_application_get_flags", line=2635) at src/switch_channel.c:3347 >> #7 0x00002adf42bc4711 in switch_core_session_execute_application_get_flags (session=0x2adf4c33d5a8, >> app=0x2adf4d533cc8 "spandsp_start_dtmf", arg=<value optimized out>, flags=<value optimized out>) at src/switch_core_session.c:2635 >> #8 0x00002adf42bc8153 in switch_core_session_run (session=0x2adf4c33d5a8) at src/switch_core_state_machine.c:230 >> #9 0x00002adf42bc3030 in switch_core_session_thread (thread=<value optimized out>, obj=0x2adf4c33d5a8) >> at src/switch_core_session.c:1529 >> #10 0x00002adf42bbf9d0 in switch_core_session_thread_pool_worker (thread=0x2adf545d3600, obj=<value optimized out>) >> at src/switch_core_session.c:1621 >> #11 0x00000030b000683d in start_thread () from /lib64/libpthread.so.0 >> #12 0x00000030af8d4f8d in clone () from /lib64/libc.so.6 >> >> >> >> >> Last two times: >> >> >> Loaded symbols for /lib64/libnss_dns.so.2 >> Core was generated by `/usr/local/freeswitch/bin/freeswitch -nc -hp'. >> Program terminated with signal 11, Segmentation fault. >> #0 rtmp_send_message (rsession=0x0, amfnumber=3 '\003', timestamp=0, type=20 '\024', stream_id=0, message=0x2b8329b6b410 "\002", len=68, flags=0) at rtmp.c:562 >> 562 if ((rsession->send_ack + rsession->send_ack_window) < rsession->send && >> >> >> Core dump from last time: >> >>>>>> >>>>>> >>>>>> Program terminated with signal 11, Segmentation fault. >>>>>> #0 rtmp_send_message (rsession=0x0, amfnumber=3 '\003', timestamp=0, type=20 '\024', stream_id=0, message=0x46eaf410 "\002", len=71, flags=0) at rtmp.c:562 >>>>>> 562 if ((rsession->send_ack + rsession->send_ack_window) < rsession->send && >>>>> > > >

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WebRTC video in mod_conference
by Joao Barbosa 08 Nov '13

08 Nov '13

Hi, i have searched for information about FS support to WebRTC video and the problems in mod_conference are related with video transcoding. My question is, what if there are three or more persons using the same WebRTC client and want to join a conference, how does a participant receives the video from the remaining ones? In this case there is no need for transcoding in video but i don't know how one person receives the video from the others. The client is based on JsSIP library. Thank you -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/WebRTC-video-in-mod-conferenc… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Sending Sip Message when receiver client is offline
by Huseyin TURAN 08 Nov '13

08 Nov '13

Hi I have FS server and CSipSimple client network. I have a problem receiving Sip Message when the client is offline. Sender client sends a Sip Message when the receiver is offline and then receiver gets online but receiver can not get the Sip message. Any advice? Thanks in advance. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Sending-Sip-Message-when-rece… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Freeswitch and a legacy combination SPA962+SPA932 regards BLF
by support＠ecn.net.au 08 Nov '13

08 Nov '13

Hi All We're attempting to sort out some issues between Freeswitch and a legacy combination SPA962+SPA932 regards BLF. The issue is on register the SPA962 subscribes for each BLF, however the SPA932 button stays flashing orange (sub the SPA is successfully registered). Once a call is made or received on a BLF monitor extension the SPA932 button lights RED then GREEN and stays functional (until the next phone reboot). The SIP trace is here: ------------------------------------------------------------------------ recv 421 bytes from udp/[10.0.2.127]:5060 at 00:45:11.742131: ------------------------------------------------------------------------ SUBSCRIBE sip:303@10.0.2.250 SIP/2.0 Via: SIP/2.0/UDP 10.0.2.127:5060;branch=z9hG4bK-7f609a1f From: "322" <sip:322@10.0.2.250>;tag=a18f86cd934221c To: <sip:303@10.0.2.250> Call-ID: f1821561-f766c498(a)10.0.2.127 CSeq: 18390 SUBSCRIBE Max-Forwards: 70 Contact: "322" <sip:322@10.0.2.127:5060> Accept: application/dialog-info+xml Expires: 60 Event: dialog User-Agent: Linksys/SPA962-6.1.5(a) Content-Length: 0 ------------------------------------------------------------------------ send 732 bytes to udp/[10.0.2.127]:5060 at 00:45:11.744088: ------------------------------------------------------------------------ SIP/2.0 202 Accepted Via: SIP/2.0/UDP 10.0.2.127:5060;branch=z9hG4bK-7f609a1f From: "322" <sip:322@10.0.2.250>;tag=a18f86cd934221c To: <sip:303@10.0.2.250>;tag=cfF5YNkjtE2v Call-ID: f1821561-f766c498(a)10.0.2.127 CSeq: 18390 SUBSCRIBE Contact: <sip:303@10.0.2.250:5060> Expires: 60 User-Agent: FreeSWITCH-mod_sofia/1.5.5b+git~20130801T015433Z~f255f65a82 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=60 Content-Length: 0 ------------------------------------------------------------------------ send 983 bytes to udp/[10.0.2.127]:5060 at 00:45:11.745843: ------------------------------------------------------------------------ NOTIFY sip:322@10.0.2.127:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.2.250;rport;branch=z9hG4bKgr3D387pp3Q8g Max-Forwards: 70 From: <sip:303@10.0.2.250>;tag=cfF5YNkjtE2v To: "322" <sip:322@10.0.2.250>;tag=a18f86cd934221c Call-ID: f1821561-f766c498(a)10.0.2.127 CSeq: 698819123 NOTIFY Contact: <sip:303@10.0.2.250:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.5.5b+git~20130801T015433Z~f255f65a82 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: dialog Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=60 Content-Type: application/dialog-info+xml Content-Length: 148 <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" entity="sip:303@ecninternal"> </dialog-info> It seems the NOTIFY message sent to the Phone doesn't satisfy the phones "BLF needs" to make the lamp solid Green. The result is the reception flashes an odd orange until calls are made through the unit. Has anyone encountered this issue with SPA932 and Freeswitch (oddly Asterisk which they migrated from doesn't present the same issue - so they must craft the NOTIFY messages differently I assume? Many Thanks Mark The information contained in this mail message is confidential and may also be legally privileged. If you are not the intended recipient, please note that any use, dissemination, further distribution, or reproduction of this message in any form whatsoever, is strictly prohibited. If the mail is in error, please notify me by return E-mail, delete your copy of the message, and accept my apologies for any inconvenience caused.

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A few interesting historical links for telephony folks
by William King 07 Nov '13

07 Nov '13

Analysis of the modem tones: http://i.imgur.com/5Dq6K2U.png http://www.youtube.com/watch?v=vvr9AMWEU-c Historical telephony switching: http://www.youtube.com/watch?v=xZePwin92cI -- William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518 william.king(a)quentustech.com

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Additional Feature added to Jira! Now with Draw.IO goodness
by Ken Rice 07 Nov '13

07 Nov '13

Hey Guys, Ever tried explaining a setup where a diagram would be much easier on your tickets? Now you can... Open your ticket on Jira as normal. Once its opened, pull your ticket up click the ³More Actions² button on your ticket, then click the ³Add Drawio Diagram² link. Draw your diagram and save it to the ticket! -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org G+ ClueCon : http://fs0.us/cluecon-gplus FB ClueCon : http://fs0.us/cluecon-fb G+ FreeSwitch : http://fs0.us/freeswitch-gplus FB FreeSWITCH : http://fs0.us/freeswitch-fb Twitter : @FreeSWITCH_WIRE irc.freenode.net #freeswitch

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