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Can not make outgoing call by FXO port Yeastar TDM400
by Nguyễn Đức Vinh 12 Oct '12

12 Oct '12

Hi all, I am trying to make an outgoing call using FXO connected to an external telephone line. I use Linphone (account 1001) to make a call to mobile number 0973385525 But I got this log: *outgoing_call method not implemented in this span* 2012-10-12 11:30:49.552355 [DEBUG] mod_freetdm.c:1238 Connect outbound channel FreeTDM/1:3/0973385525 2012-10-12 11:30:49.552355 [NOTICE] switch_channel.c:951 New Channel FreeTDM/1:3/0973385525 [71426de3-5e2d-44f0-ad1d-46328ce86294] 2012-10-12 11:30:49.552355 [DEBUG] mod_freetdm.c:1247 (FreeTDM/1:3/0973385525) State Change CS_NEW -> CS_INIT 2012-10-12 11:30:49.552355 [DEBUG] switch_core_session.c:1210 Send signal FreeTDM/1:3/0973385525 [BREAK] 2012-10-12 11:30:49.552355 [DEBUG] mod_freetdm.c:1266 Attached session 71426de3-5e2d-44f0-ad1d-46328ce86294 to channel 1:3 2012-10-12 11:30:49.552355 [ERR] ftdm_io.c:2681 [s1c3][1:3] *outgoing_call method not implemented in this span!* 2012-10-12 11:30:49.552355 [DEBUG] ftdm_io.c:3185 [s1c3][1:3] DTMF debug is already disabled 2012-10-12 11:30:49.552355 [DEBUG] ftdm_io.c:3217 [s1c3][1:3] No need to disable input dump 2012-10-12 11:30:49.552355 [DEBUG] ftdm_io.c:3248 [s1c3][1:3] No need to disable output dump 2012-10-12 11:30:49.552355 [DEBUG] ftdm_io.c:2990 [s1c3][1:3] channel done 2012-10-12 11:30:49.552355 [NOTICE] mod_freetdm.c:1756 Close Channel FreeTDM/1:3/0973385525 [CS_INIT] 2012-10-12 11:30:49.552355 [DEBUG] switch_core_state_machine.c:530 (FreeTDM/1:3/0973385525) Running State Change CS_DESTROY 2012-10-12 11:30:49.552355 [DEBUG] switch_core_state_machine.c:540 (FreeTDM/1:3/0973385525) State DESTROY 2012-10-12 11:30:49.552355 [DEBUG] switch_core_state_machine.c:540 (FreeTDM/1:3/0973385525) State DESTROY going to sleep 2012-10-12 11:30:49.552355 [NOTICE] switch_ivr_originate.c:2591 Cannot create outgoing channel of type [freetdm] cause: [DESTINATION_OUT_OF_ORDER] 2012-10-12 11:30:49.552355 [DEBUG] switch_ivr_originate.c:3508 Originate Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER] 2012-10-12 11:30:49.552355 [INFO] mod_dptools.c:3027 Originate Failed. Cause: DESTINATION_OUT_OF_ORDER * Could anyone help me out in getting FXO working? * ** *Here is detail setup.* Yeastar TDM400 with 2 O2 modules installed (The O2 is a module with 2 FXO ports to terminate two analog PSTN lines). I installed DAHDI driver (follow this guide http://www.yeastar.com/download/TDM_DriverInstall_dahdi.pdf) and installed freeTDM with Freeswitch ( http://wiki.freeswitch.org/wiki/FreeTDM#FreeTDM_with_FreeSWITCH) *1. Result of dahdi_hardware* pci:0000:02:01.0 wctdm+ e159:0001 Wildcard S400P Prototype *2. Result of dahdi_cfg -vv* DAHDI Tools Version - 2.6.1 DAHDI Version: 2.3.0.1 Echo Canceller(s): MG2 Configuration ====================== Channel map: Channel 01: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 04) 4 channels to configure. Setting echocan for channel 1 to mg2 Setting echocan for channel 2 to mg2 Setting echocan for channel 3 to mg2 Setting echocan for channel 4 to mg2 *3. File freeswitch/conf/freetdm.conf* [span zt FXO1] fxo-channel => 1,2,3,4 *4. File freeswitch/conf/autoload_configs/freetdt.conf.xml* <analog_spans> <span name="FXO1" > <param name="dialplan" value="XML"/> <param name="context" value="default"/> </span> </analog_spans> *5. Result of CLI command: ftdm dump 1* +OK span_id: 1 chan_id: 1 physical_span_id: 1 physical_chan_id: 1 physical_status: alarmed physical_status_red: 1 physical_status_yellow: 0 physical_status_rai: 0 physical_status_blue: 0 physical_status_ais: 0 physical_status_general: 0 signaling_status: DOWN type: FXS state: DOWN last_state: DOWN txgain: 0.00 rxgain: 0.00 cid_date: cid_name: cid_num: ani: aniII: dnis: rdnis: cause: NONE session: (none) span_id: 1 chan_id: 2 physical_span_id: 1 physical_chan_id: 2 physical_status: alarmed physical_status_red: 1 physical_status_yellow: 0 physical_status_rai: 0 physical_status_blue: 0 physical_status_ais: 0 physical_status_general: 0 signaling_status: DOWN type: FXS state: DOWN last_state: DOWN txgain: 0.00 rxgain: 0.00 cid_date: cid_name: cid_num: ani: aniII: dnis: rdnis: cause: NONE session: (none) span_id: 1 chan_id: 3 physical_span_id: 1 *physical_chan_id: 3* *physical_status: ok //(I have a telephone line connect to port 3)* physical_status_red: 0 physical_status_yellow: 0 physical_status_rai: 0 physical_status_blue: 0 physical_status_ais: 0 physical_status_general: 0 signaling_status: DOWN type: FXS state: DOWN last_state: DOWN txgain: 0.00 rxgain: 0.00 cid_date: cid_name: cid_num: ani: aniII: dnis: rdnis: cause: NONE session: (none) span_id: 1 chan_id: 4 physical_span_id: 1 physical_chan_id: 4 physical_status: alarmed physical_status_red: 1 physical_status_yellow: 0 physical_status_rai: 0 physical_status_blue: 0 physical_status_ais: 0 physical_status_general: 0 signaling_status: DOWN type: FXS state: DOWN last_state: DOWN txgain: 0.00 rxgain: 0.00 cid_date: cid_name: cid_num: ani: aniII: dnis: rdnis: cause: NONE session: (none) *6. A snipet from CLI command: reload mod_freetdm* 2012-10-12 11:31:43.672340 [INFO] ftdm_io.c:860 Auto-loaded I/O module 'zt' 2012-10-12 11:31:43.672340 [DEBUG] ftdm_io.c:5160 created span 1 (FXO1) of type zt 2012-10-12 11:31:43.672340 [DEBUG] ftdm_io.c:5178 span 1 [fxo-channel]=[1,2,3,4] 2012-10-12 11:31:43.672340 [DEBUG] ftdm_io.c:5213 setting trunk type to 'FXO' start(KEWL), mode(CPE) 2012-10-12 11:31:43.672340 [INFO] ftmod_zt.c:405 configuring device /dev/dahdi/channel channel 1 as FreeTDM device 1:1 fd:40 2012-10-12 11:31:43.672340 [INFO] switch_time.c:1163 Timezone reloaded 530 definitions 2012-10-12 11:31:43.672340 [INFO] ftmod_zt.c:405 configuring device /dev/dahdi/channel channel 2 as FreeTDM device 1:2 fd:41 2012-10-12 11:31:43.672340 [INFO] ftmod_zt.c:405 configuring device /dev/dahdi/channel channel 3 as FreeTDM device 1:3 fd:42 2012-10-12 11:31:43.672340 [INFO] ftmod_zt.c:405 configuring device /dev/dahdi/channel channel 4 as FreeTDM device 1:4 fd:43 2012-10-12 11:31:43.672340 [DEBUG] ftdm_io.c:5853 Creating new group:__default 2012-10-12 11:31:43.672340 [INFO] ftdm_io.c:5416 Configured 4 channel(s) *7. Diaplan default.xml* <action application="bridge" data="freetdm/1/3/0973385525"/> *8. File /etc/dahdi/system.cfg* # Span 1: WCTDM/4 "Wildcard S400P Prototype Board 5" (MASTER) fxsks=1 echocanceller=mg2,1 fxsks=2 echocanceller=mg2,2 fxsks=3 echocanceller=mg2,3 fxsks=4 echocanceller=mg2,4 # Global data loadzone = us defaultzone = us =========================== (I am from Vietnam)* * Thanks in advance, BR, Nguyen Duc Vinh, Mr

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FreeSWITCH Conference Call Today
by Michael Collins 10 Oct '12

10 Oct '12

Hello folks, Today's conference call agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2012_10_10 Ken Rice is going to be leading us in a discussion on how to gather data for troubleshooting and reporting on Jira. It should be very informative and will be good knowledge to spread amongst the community and especially for those who are new to FreeSWITCH. 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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FS Core ODBC Support - Specified key was too long; max key length is 1000 bytes
by Muhammad Shahzad 08 Oct '12

08 Oct '12

Hi, I have compiled FS with core odbc support and get below error at start up, ======================================================= 2012-10-09 01:49:33.596802 [ERR] switch_odbc.c:494 ERR: [create index complete11 on complete (a1,a2,a3,a4,a5,a6,a7,a8,a9,a10,hostname)] [STATE: HY000 CODE 1071 ERROR: [unixODBC][MySQL][ODBC 5.1 Driver][mysqld-5.1.63-0+squeeze1]Specified key was too long; max key length is 1000 bytes ] 2012-10-09 01:49:33.596864 [ERR] switch_core_sqldb.c:526 SQL ERR [STATE: HY000 CODE 1071 ERROR: [unixODBC][MySQL][ODBC 5.1 Driver][mysqld-5.1.63-0+squeeze1]Specified key was too long; max key length is 1000 bytes ] create index complete11 on complete (a1,a2,a3,a4,a5,a6,a7,a8,a9,a10,hostname) ======================================================= I know its MySQL specific error as specified at, http://stackoverflow.com/questions/7846298/max-key-length-is-1000-bytes-in-… However, i was wondering if there is any work around for it? Can we ignore this error or it has to be fixed before putting FS in production? Thank you. -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 MSN: shari_786pk(a)hotmail.com Email: shaheryarkh(a)googlemail.com

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FreeSWITCH Weekly News and Notes
by Michael Collins 08 Oct '12

08 Oct '12

Hello all! Last week was mostly status quo - busy but not too busy. We spent time working on chapters 5 and 6 of the second edition of the FreeSWITCH book<https://www.packtpub.com/freeswitch-1-0-6-build-robust-high-performance-tel…>. We also had an interesting discussion<http://wiki.freeswitch.org/wiki/FS_weekly_2012_10_03>about Lumicall <http://www.lumicall.org>, an Android application that allows for mobile VoIP. This week<http://wiki.freeswitch.org/wiki/FS_weekly_2012_10_10>we our very own Ken Rice (with an assist from yours truly) will be discussing FreeSWITCH data gathering and troubleshooting techniques. We will also be talking about how to get all of this data into a Jira in such a way that the developers will be thankful. The following week we've scheduled Chad Engler to come discuss with us his node.js/ESL interface<https://github.com/englercj/node-esl> . In other news we just wanted to let you know that our friends over at Vestec<http://vestec.com>have closed a new round of funding. Check out this release <http://www.prweb.com/releases/2012/10/prweb9961259.htm> for details. We are getting ready to put the finishing touches on the voice-enabled application contest. In the meantime keep working on those application ideas - you may just be the one to win some cash! We thank the Vestec folks for being so supportive. Have a great week! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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Maintenance Note: FreeSWITCH.org Sites
by Michael Collins 03 Oct '12

03 Oct '12

Hello all! Just a note: this coming Friday night at 9PM CST we will be performing some maintenance on a number of sites that inhabit the freeswitch.org domain. This includes wiki.freeswitch.org and www.freeswitch.org. The maintenance probably won't last more than an hour or two. Thanks! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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FreeSWITCH Conference Call Today
by Michael Collins 03 Oct '12

03 Oct '12

Hey all! Don't forget about our conference call today: http://wiki.freeswitch.org/wiki/FS_weekly_2012_10_03 We are going to discuss mobile VoIP and Lumicall for Android devices. If you have an Android device you might want to get Lumicall from the Play Store (it's free) so you'll be ready to try it out on today's call. 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 01 Oct '12

01 Oct '12

Welcome to October! I hope the weather is nice where you are. Here it's still above 100F. :) Last week was a little bit quieter than the previous few weeks. I had a chance to work on the FreeSWITCH change log and I made a list<http://wiki.freeswitch.org/wiki/FS_weekly_2012_09_26#News.2C_Notes.2C_.26_M…>of some of the APIs, dialplan tools, and channel variables. These have all been added since 1.2.0 was initially released in early August. All of them have wiki entries - thanks to those who took the initiative to do add them. Feel free to add your knowledge and experience to the mix. Last week's conference call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_09_26>was an object lesson in the challenges of getting a SIP proxy working with TLS and FreeSWITCH. This week<http://wiki.freeswitch.org/wiki/FS_weekly_2012_10_03>we are going to change direction and look at something that has been slowly (and painfully) advancing the past few years: mobile VoIP. We will be having Daniel Pocock share with us some information about Lumicall<http://www.lumicall.org>, an open source mobile VoIP client for Android devices. There is also a service component to Lumicall and we'll be learning about that as well. Come join us to see the state of mobile VoIP<http://en.wikipedia.org/wiki/Mobile_VoIP>. We are working on some fresh presentations for later this month. We hope to have an update on e164.org and how we can all get involved. We are also preparing a presentation on how to perform some of the data-gathering techniques <http://wiki.freeswitch.org/wiki/Reporting_Bugs> that are needed for basic and advanced troubleshooting. If you have some input on these or other presentations please let me or Ken Rice know. Thanks! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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Unreachable code in mod_spidermonkey.c (event_chat_execute)
by Niels Thomsen 01 Oct '12

01 Oct '12

There's a bit of unreachable code in mod_spidermonkey which means that the event_chat_execute function becomes a no-op. This may be intentional, of course, but it doesn't look like it is... static JSBool event_chat_execute(JSContext * cx, JSObject * obj, uintN argc, jsval * argv, jsval * rval) { struct event_obj *eo = JS_GetPrivate(cx, obj); if (eo) { if (argc > 0) { char *app = JS_GetStringBytes(JS_ValueToString(cx, argv[0])); char *arg = NULL; if (argc > 1) { arg = JS_GetStringBytes(JS_ValueToString(cx, argv[1])); } goto end; //-- this code is never executed switch_core_execute_chat_app(eo->event, app, arg); *rval = BOOLEAN_TO_JSVAL(JS_TRUE); return JS_TRUE; } } end: *rval = BOOLEAN_TO_JSVAL(JS_FALSE); return JS_FALSE; } - Niels

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Possible bug in video code
by Rick van Rein 01 Oct '12

01 Oct '12

Hello, I found this surprising fragment in the sofia_glue.c routine tech_pvt->adv_sdp_video_port = sdp_port; switch_channel_set_variable(tech_pvt->channel, SWITCH_LOCAL_VIDEO_IP_VARIABLE, tech_pvt->adv_sdp_audio_ip); switch_channel_set_variable_printf(tech_pvt->channel, SWITCH_LOCAL_VIDEO_PORT_VARIABLE, "%d", sdp_port); I'm almost certain that "adv_sdp_audio_ip" should be "adv_sdp_video_ip", except that it isn't there in mod_sofia.h; is this code assuming that Audio and Video come from the same source? I don't think this assumption will remain valid with the advent of IPv6 and potential LAN-mounted camera's. I am actually coding open source phone firmware that explicitly will pull in networked media devices by passing on their direct IPv6 addresses. This could be as simple as a webcam on a host. If you don't mind I'll leave it to the regular developers to judge and fix this in the git repository. I would like to learn the outcome. Cheers, -Rick

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Handling multipart/mixed SDP in INVITE
by Debasish Chandra 01 Oct '12

01 Oct '12

Hi, I am getting following error (415 Unsupported Media Type) while my FS receiving INVITE with multipart bodies. However it seems sofia is responding with 415. Does sofia handle multipart bodies? I am using latest freeswitch build. I am attaching the log below. ------------------------------------------------------------------------------------------------------------------------------------------------------------------- tport_wakeup_pri(0x7ff2c80041f0): events IN tport_recv_event(0x7ff2c80041f0) tport_recv_iovec(0x7ff2c80041f0) msg 0x7ff2c8005f30 from (udp/ 192.168.1.10:5060) has 1015 bytes, veclen = 1 tport_deliver(0x7ff2c80041f0): msg 0x7ff2c8005f30 (1015 bytes) from udp/ 192.168.1.10:5060/sip next=(nil) nta: received INVITE sip:222222@192.168.1.10:5060 SIP/2.0 (CSeq 1) nta: canonizing sip:222222@192.168.1.10:5060 with contact nta: INVITE (1) going to a default leg nta: timer set to 200 ms nua: nua_stack_process_request: entering nua: nua_invite_server_respond: entering tport_tsend(0x7ff2c80041f0) tpn = UDP/192.168.1.10:32896 tport_resolve addrinfo = 192.168.1.10:32896 tport_by_addrinfo(0x7ff2c80041f0): not found by name UDP/192.168.1.10:32896 tport_vsend(0x7ff2c80041f0): 711 bytes of 711 to udp/192.168.1.10:32896 tport_vsend returned 711 nta: sent 415 Unsupported Media Type for INVITE (1) tport_wakeup_pri(0x7ff2c80041f0): events IN tport_recv_event(0x7ff2c80041f0) tport_recv_iovec(0x7ff2c80041f0) msg 0x7ff2c8009800 from (udp/ 192.168.1.10:5060) has 386 bytes, veclen = 1 tport_deliver(0x7ff2c80041f0): msg 0x7ff2c8009800 (386 bytes) from udp/ 192.168.1.10:5060/sip next=(nil) nta: received ACK sip:222222@192.168.1.10:5060 SIP/2.0 (CSeq 1) nta: ACK (1) is going to INVITE (1) nta: timer set next to 4801 ms nta: timer I fired, terminate 415 response incoming_reclaim_all((nil), (nil), 0x7ff2f2712b50) nta_incoming_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free nta: timer not set -------------------------------------------------------------------------------------------------------------------------------------------------------------------- Please let me know whether it is possible to use multipart with FS. Best Regards, Debasish

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