November 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

busy tone detection
by Gopal krishnan 05 Jan '09

05 Jan '09

Hi, Is it possible to have busy tone detection thru console dialing? -- Thank you with regards, Gopal,

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Conference
by Baskar 03 Jan '09

03 Jan '09

Hi, *Just i want to call the 1002==>9841799874 and i want to call another no 9840544078 and add in to the conference room * *First method i tried :* api originate sofia/internal/1002%172.20.191.227 &bridge(sofia/default/ 9841799874(a)172.20.191.228) ====>1002 call 9841799874 api uuid_transfer 04120c08-19e7-4881-8873-0834a11524ff -both 3001 ====>Both caller and extension will be transfer into conference room Then i call the another caller api originate sofia/internal/1002%172.20.191.227 &bridge(sofia/default/ 9840544078(a)172.20.191.228)====> 1002 call another caller 9840544078 api uuid_transfer c564f668-6778-4e78-85f6-a78c9c67cfa4 -bleg 3001 ====>Transfer only the caller 9840544078 to conference room 3001 while calling the second no 9840544078 the first line get hold after the transfer of bleg they all come in to the conference room but the first line caller is in hold so we cant able to unhold the call through api we have to manually go the soft phone and unhold the call . but i need the unhold should be done in api what is the solution . *i have tried in another method also * api originate sofia/internal/1002%172.20.191.227 &bridge(sofia/default/ 9841799874(a)172.20.191.228)====>1002 call 9841799874 api uuid_transfer 04120c08-19e7-4881-8873-0834a11524ff -both 3001 ====>Transfer only the caller 9841799874 to conference room 3001 api uuid_kill 04120c08-19e7-4881-8873-0834a11524ff before i hangup i have transfer both the caller in to conference room so when i hangup the extension why my extension got hangup Can any one suggest me how can i try the conference are else correct where i am wrong. -- Warm Regards, N.Baskar

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OpenZap Not working Stable
by palani sivagurunathan 17 Dec '08

17 Dec '08

Hi, I have installed freeswitch through SVN in centos.OutBound is working fine .But not always .Once we make 8 to 10 call from a sip extension we get service unavailable error in xlite . Error and configuration are pasted below .we observed the problem is getting rectified after 5 to 10 mins .and we are able to make calls and again after 8 to 10 calls Errors happen how to check. what will be the issue my configuration is: *wanpipe1.conf * [devices] wanpipe1 = WAN_AFT_TE1, Comment [interfaces] w1g1 = wanpipe1, , TDM_VOICE_API, Comment [wanpipe1] CARD_TYPE = AFT S514CPU = A CommPort = PRI AUTO_PCISLOT = NO PCISLOT = 9 PCIBUS = 0 FE_MEDIA = E1 FE_LCODE = HDB3 FE_FRAME = CRC4 FE_LINE = 1 TE_CLOCK = NORMAL TE_REF_CLOCK = 0 TE_SIG_MODE = CCS TE_HIGHIMPEDANCE = NO LBO = 120OH FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 1 TDMV_DCHAN = 16 TDMV_HW_DTMF = NO [w1g1] ACTIVE_CH = ALL TDMV_ECHO_OFF = NO TDMV_HWEC = YES MTU = 80 * openzap.conf* [span wanpipe] name => OpenZAP number => 1 trunk_type => E1 b-channel => 1:1-15 d-channel => 1:16 b-channel => 1:17-31 *openzap.conf.xml * <configuration name="openzap.conf" description="OpenZAP Configuration"> <settings> <param name="debug" value="0"/> </settings> <pri_spans> <span id="1"> <param name="mode" value="user"/> <param name="dialect" value="euro"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> </span> </pri_spans> </configuration> *default.xml* <?xml version="1.0" encoding="utf-8"?>  <include> <context name="default"> <extension name="Long Distance - wanpipe"> <condition field="destination_number" expression="^9(\d{5,15})$"> <action application="set" data="dialed_ext=$1"/> <action application="bridge" data="OpenZAP/1/a/${dialed_ext}"/> </condition> </extension> </context> </include> After starting the freeswitch . In console i type oz dump 1 a *my status before calling is:* span_id: 1 chan_id: 16 physical_span_id: 1 physical_chan_id: 16 type: DQ921 state: UP last_state: DOWN cid_date: cid_name: cid_num: ani: aniII: dnis: rdnis: cause: NONE * my (oz dump 1 a) status After calling is:* span_id: 1 chan_id: 16 physical_span_id: 1 physical_chan_id: 16 type: DQ921 state: UP last_state: DOWN cid_date: cid_name: cid_num: ani: aniII: dnis: rdnis: cause: NONE But while i try to make a outbound call continously, i am getting following error [08 02 00 00 4e 18 03 a1 83 81 79 01 80] 2008-09-23 01:05:50 [DEBUG] Span:0 Q.931() Sending message to Layer4 (size: 119) 2008-09-23 01:05:50 [DEBUG] ozmod_isdn.c:319 zap_isdn_931_34() Yay I got an event! Type:[46] Size:[119] CRV: 0 (0, CTX: Originator) 2008-09-23 01:05:50 [DEBUG] ozmod_isdn.c:351 zap_isdn_931_34() zchan 0 (-1:-1) source isdn_data->channels_remote_crv[0] 2008-09-23 01:05:50 [DEBUG] ozmod_isdn.c:399 zap_isdn_931_34() Changing state on 1:1 from DOWN to RESTART 2008-09-23 01:05:50 [DEBUG] Span:0 Q.931() Sending message to Q.921 (size: 119) 2008-09-23 01:05:50 [DEBUG] Span:0 Q.931() Creating Q.931 Message Header: ProtDisc 8 (0x8), CRV 0 (0), CRVflag: 0 (0), MesType: 78 (0x4e) 2008-09-23 01:05:50 [DEBUG] ozmod_isdn.c:1499 q931_rx_32() WRITE 13 -------------------------------------------------------------------------------- [08 02 00 00 4e 18 03 a1 83 81 79 01 80] 2008-09-23 01:05:50 [DEBUG] ozmod_isdn.c:796 state_advance() 1:1 STATE [RESTART] 2008-09-23 01:05:50 [DEBUG] mod_openzap.c:1368 on_clear_channel_signal() got clear channel sig [RESTART] 2008-09-23 01:05:50 [DEBUG] ozmod_isdn.c:882 state_advance() Changing state on 1:1 from RESTART to DOWN 2008-09-23 01:05:50 [DEBUG] ozmod_isdn.c:796 state_advance() 1:1 STATE [DOWN] 2008-09-23 01:05:50 [DEBUG] zap_io.c:1088 zap_channel_done() channel done 1:1 -- Thanks & Regards, Palanivel.S

6 15

OpenZAP: "Received unhandled message 125"
by Peter P GMX 16 Dec '08

16 Dec '08

I have installed OpenZAP with a TE220 card and EuroISDN. When I bridge an outgoing call I get a "Received unhandled message 125" Any idea what that means? As far as I know there is no <external> result code 125 defined in the ISDN protocol. Best regards Peter See the following logs. 2008-11-27 17:35:17 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 14 -------------------------------------------------------------------------------- [08 02 80 03 7d 08 04 82 e3 98 28 14 01 01] 2008-11-27 17:35:17 [DEBUG] Span:0 Q.931() Received message from Q.921 (ind 4, tei 0, size 18) MesType: 125, CRVFlag 1 (Terminator), CRV 3 (Dialect: 2) 2008-11-27 17:35:17 [DEBUG] Span:0 Q.931() Sending message to Layer4 (size: 114) 2008-11-27 17:35:17 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got an event! Type:[7d] Size:[114] CRV: 3 (0x3, CTX: Terminator) 2008-11-27 17:35:17 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan 70d230 (1:1) source isdn_data->channels_local_crv[0x3] 2008-11-27 17:35:17 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() Received unhandled message 125 (0x7d) 2008-11-27 17:35:17 [DEBUG] Span:0 Q.921() Sending frame >oz list API CALL [oz(list)] output: +OK span: 1 (span1) type: isdn chan_count: 31 dialplan: XML context: default dial_regex: fail_dial_regex: hold_music: analog_options none +OK span: 2 (span2) type: isdn chan_count: 31 dialplan: XML context: default dial_regex: fail_dial_regex: hold_music: analog_options none

4 4

Console Dialing in Freeswitch
by Baskar 11 Dec '08

11 Dec '08

*Hi,* *It is possible to dial outbound through console dialing. Yes means me How?* *Another question whether there is any api command for console dialing.* * Thank you with regards, N.Baskar *

6 18

Autoanswer
by Jonas Gauffin 08 Dec '08

08 Dec '08

Hello I send an API command through the event socket that looks like this (the first two variables are used by our server): api originate {gate_user_id=44,gate_site_id=1,sip_invite_params=intercom=true,sip_h_Call-Info=<sip:$${domain}>;answer-after=0,sip_auto_answer=true,origination_caller_id_name='Jonas Gauffin',origination_caller_id_number=+4623666XXXX,sip_auto_answer=true}sofia/localdomain/u1000044% 192.168.1.112 '&execute_extension(8901 XML)' The command works just as it should. The problem is that the auto-answer variables seems to stick to the b-leg (execute extensions), which means that both calls gets auto answered. What I want is that only the first call gets answered. (A logged in user presses the "call" icon in our webdirectory, which makes freeswitch use his phone to call 8901) //Jonas

2 3

Windows is slow?
by Per Møller 06 Dec '08

06 Dec '08

I have installed FS 1.0.0 on a Mac using the precompiled .dmg and FS 1.0.1 on a Windows Vista machine using the precompiled .msi - actually the same machine). Using the default configuration files, and using 2 Snom 360 phones I dialed from extension 1000 to extension 1001. On the Mac, 1001 starts ringing instantly, but under Windows it takes 1-2 seconds before it starts ringing. It seems to be in the dialplan the time is spent. From the time I see this line on the console: [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 1000->1000 in context default Until the next thing happens it always takes at least 1 full second, but on the Mac it happens instantly. Why is the Windows build this much slower? Is it a known problem? I get the feeling that the majority of the FS community is Unix based, which is fine by me, but I would really like to know just how well supported and stable the Win32 build is and if this is currently a viable way to go, or if I should stick to Linux/BSD/Mac for production use? // Per

5 7

Redirect calls between FS
by Barray McKee 03 Dec '08

03 Dec '08

Hello, I am implementing 2 load balancing FS behind a pair of sip proxies. Since traffic is routed to one of the two FS on a round-robin basis, I need a mechanism to reroute call from one FS to another FS under a specific special circumstance. I am think to use the bridge command with bypass_media = true, so that the call can just pass through one FS and reach the second FS. Is this the approach that I should use? I am wondering if there is a better approach out there that I just can't think of. Thanks in advance, Barray

2 1

Unstable att_xfer
by x y 03 Dec '08

03 Dec '08

Hy! I have tested several times so far the att_xfer function of the freeswitch, and I've found it unstable. I'm using a similar code to the example at the freeswitch wiki, and sometimes a call isnt transfered. The att_xfer works fine in the cases as I figured out from the log, when the switch_ivr_bridge comes back with a "ending bridge by request from read function" message. When it comes back with a "request from write function", the two channels dont get bridged together: they cant hear each other form the other side, and they dont even realize that the other hangs up. I have pasted some logs about this a week ago into paste bin, but if you wish, I could paste new logs with the newest fs. Could it has something with silence, because I think there are more failures when I dont talk into the phones? Im using latest fs, and sip phones without built in transfer. Cheers, Viktor Hirdetés (x) Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu - a kötelező biztosítások kiindulópontja!

6 22

Problems with Mod_openMRCP
by mszlazak＠aol.com 02 Dec '08

02 Dec '08

I'm getting the following errors when trying to run the example in the wiki: http://wiki.freeswitch.org/wiki/Mod_openmrcp 2008-11-28 09:59:54 [DEBUG] switch_core_session.c:435 switch_core_session_receive_message() Send signal sofia/internal/1000(a)10.0.0.2 [BREAK] 2008-11-28 09:59:54 [DEBUG] sofia.c:2269 sofia_handle_sip_i_state() Channel sofia/internal/1000(a)10.0.0.2 entering state [completed] 2008-11-28 09:59:54 [NOTICE] mod_spidermonkey.c:2034 session_answer() Channel [sofia/internal/1000(a)10.0.0.2] has been answered 2008-11-28 09:59:54 [DEBUG] mod_spidermonkey.c:1851 init_speech_engine() Raw Codec Activation Success L16@8000hz 1 channel 20ms 2008-11-28 09:59:54 [DEBUG] mod_openmrcp.c:634 openmrcp_tts_open() Create Synthesizer Channel 2008-11-28 09:59:54 [DEBUG] sofia.c:2269 sofia_handle_sip_i_state() Channel sofia/internal/1000(a)10.0.0.2 entering state [ready] 2008-11-28 09:59:59 [ERR] mod_openmrcp.c:643 openmrcp_tts_open() No response from client stack 2008-11-28 09:59:59 [ERR] mod_openmrcp.c:647 openmrcp_tts_open() No synthesizer channel available 2008-11-28 09:59:59 [ERR] mod_spidermonkey.c:1859 init_speech_engine() Invalid TTS module! 2008-11-28 09:59:59 [ERR] inline:1 mod_spidermonkey()? Cannot allocate speech engine! 2008-11-28 09:59:59 [NOTICE] switch_core_state_machine.c:160 switch_core_standard_on_execute() Hangup sofia/internal/1000(a)10.0.0.2 [CS_EXECUTE] [NORMAL_CLEARING] 2008-11-28 09:59:59 [DEBUG] switch_channel.c:1449 switch_channel_perform_hangup() Send signal sofia/internal/1000(a)10.0.0.2 [KILL] 2008-11-28 09:59:59 [DEBUG] switch_core_session.c:727 switch_core_session_signal_state_change() Send signal sofia/internal/1000(a)10.0.0.2 [BREAK] 2008-11-28 09:59:59 [DEBUG] switch_core_state_machine.c:432 switch_core_session_run() (sofia/internal/1000(a)10.0.0.2) State EXECUTE going to sleep 2008-11-28 09:59:59 [DEBUG] switch_core_state_machine.c:367 switch_core_session_run() (sofia/internal/1000(a)10.0.0.2) Running State Change CS_HANGUP 2008-11-28 09:59:59 [DEBUG] switch_core_state_machine.c:395 switch_core_session_run() (sofia/internal/1000(a)10.0.0.2) State HANGUP 2008-11-28 09:59:59 [DEBUG] mod_sofia.c:276 sofia_on_hangup() Channel sofia/internal/1000(a)10.0.0.2 hanging up, cause: NORMAL_CLEARING 2008-11-28 09:59:59 [DEBUG] mod_sofia.c:333 sofia_on_hangup() Sending BYE to sofia/internal/1000(a)10.0.0.2 2008-11-28 09:59:59 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/internal/1000(a)10.0.0.2 Standard HANGUP, cause: NORMAL_CLEARING 2008-11-28 09:59:59 [DEBUG] switch_core_state_machine.c:395 switch_core_session_run() (sofia/internal/1000(a)10.0.0.2) State HANGUP going to sleep 2008-11-28 09:59:59 [DEBUG] switch_core_session.c:860 switch_core_session_thread() Session 1 (sofia/internal/1000(a)10.0.0.2) Locked, Waiting on external entities 2008-11-28 09:59:59 [NOTICE] switch_core_session.c:878 switch_core_session_thread() Session 1 (sofia/internal/1000(a)10.0.0.2) Ended 2008-11-28 09:59:59 [NOTICE] switch_core_session.c:880 switch_core_session_thread() Close Channel sofia/internal/1000(a)10.0.0.2 [CS_HANGUP] 2008-11-28 10:00:26 [DEBUG] mod_openmrcp.c:167 openmrcp_on_session_terminate() on_session_terminate called I believe I followed the instructions correctly but I can't get openmrcp to connect with Cepstrals TTS.

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