September 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] setting switch_r_sdp with outbound event socket
by Christian Löschenkohl 05 Sep '10

05 Sep '10

hello looks like there is no switch_url_decode in mod_event_socket.c would "static switch_status_t read_packet(listener_t *listener, switch_event_t **event, uint32_t timeout)" be a good place to decode the url encoded string? or would "static switch_status_t parse_command(listener_t *listener, switch_event_t **event, char *reply, uint32_t reply_len)" fit better, around if (!strncasecmp(cmd, "sendevent", 9)) { ... ... } i think it would be helpfull if everybody can set url encoded strings with the socket interface. i will give my best to get a patch for this function. br On 2010-09-02 19:36, Anthony Minessale wrote: > maybe FS is not urldecoding it in mod_event_socket > > 2010/9/2 Christian Löschenkohl<christian.loeschenkohl(a)xpirio.com>: >> hello >> >> message on loopback look now like this >> >> sendmsg >> call-command: execute >> execute-app-name: set >> execute-app-arg: >> switch_r_sdp=v%3D0%0D%0Ao%3Droot+2099281814+2099281814+IN+IP4+xx.xx.xx.xx%0D%0As%3Dcall%0D%0Ac%3DIN+IP4+xx.xx.xx.xx%0D%0At%3D0+0%0D%0Am%3Daudio+18888+RTP%2FAVP+9+99+3+18+4+101%0D%0Aa%3Drtpmap%3A9+g722%2F8000%0D%0Aa%3Drtpmap%3A3+gsm%2F8000%0D%0Aa%3Drtpmap%3A18+g729%2F8000%0D%0Aa%3Dfmtp%3A18+annexb%3Dno%0D%0Aa%3Drtpmap%3A4+g723%2F8000%0D%0Aa%3Drtpmap%3A101+telephone-event%2F8000%0D%0Aa%3Dfmtp%3A101+0-16%0D%0Aa%3Dptime%3A20%0D%0A >> >> server respons now with "SIP/2.0 500 Internal Server Error" when trying to >> bridge this call with this sdp >> >> why does that not guarentee it will work? >> >> what could be a possible solution here? >> >> bg >> >> >> >> On 2010-09-01 03:31, Anthony Minessale wrote: >> >>> You have to urlencode it too and that does not guarentee it will work.... >>> >>>> On Aug 31, 2010 6:01 PM, "Christian Löschenkohl" >>>> <christian.loeschenkohl(a)xpirio.com >>>> <mailto:christian.loeschenkohl@xpirio.com>> wrote: >>>> >>>> thank you for your reply >>>> >>>> i read it here >>>> http://wiki.freeswitch.org/wiki/Codec_negotiation#Rewriting_SDP and it >>>> seemed >>>> logic to me (which isn't true as i seems). >>>> >>>> of course i tried it without CDATA - then the command on the loopback >>>> looks like this >>>> >>>> >>>> sendmsg >>>> call-command: execute >>>> execute-app-name: set >>>> >>>> execute-app-arg: switch_r_sdp=v=0 >>>> o=root 1051875459 1051875459 IN IP4 x.x.x.x >>>> >>>> s=call >>>> c=IN IP4 x.x.x.x >>>> t=0 0 >>>> >>>> m=audio 19156 RTP/AVP 0 9 99 3 18 4 101 >>>> >>>> a=rtpmap:0 pcmu/8000 >>>> a=rtpmap:9 g722/8000 >>>> a=rtpmap:3 gsm/8000 >>>> a=rtpmap:18 g729/8000 >>>> a=fmtp:18 annexb... >>>> >>>> and i also do get [ERR] switch_ivr.c:469 Invalid Command! >>>> >>>> these lines print out the error (switch_ivr.c:469) >>>> >>>> if (zstr(cmd)) { >>>> switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), >>>> SWITCH_LOG_ERROR, "Invalid Command!\n"); >>>> return SWITCH_STATUS_FALSE; >>>> } >>>> >>>> br >>>> >>>> >>>> >>>> On 2010-09-01 00:34, Brian West wrote: >>>> >>>>> Why are you trying to put it in the CDATA? You know you... >>>> >>>> -- >>>> Ing. Christian Löschenkohl >>>> Technische Leitung, Forschung& Entwicklung VoIP >>>> >>>> xpirio >>>> Telekommunik... >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users(a)lists.freeswitch.org >>>> <mailto:FreeSWITCH-users@lists.freeswitch.org> >>>> http://lists.freeswitch.org/mail... >>>> >>> >>> >>> _______________________________________________ >>> 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 >> >> >> -- >> Ing. Christian Löschenkohl >> Technische Leitung, Forschung& Entwicklung VoIP >> >> xpirio >> Telekommunikation& Service GmbH >> Lakeside B04 >> 9020 Klagenfurt >> Austria >> >> T +43 (0) 5 77 11 - 1000 >> F +43 (0) 5 77 11 - 1002 >> E christian.loeschenkohl(a)xpirio.com >> > > > -- Ing. Christian Löschenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl(a)xpirio.com

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Fwd: Forward phone calls
by Krystian Szczęsny 04 Sep '10

04 Sep '10

Hi, I would like to build a small box with a GSM Phone connected through USB to receive and forward calls. I am wondering if it is possible using freeswitch. This is the scenario I would like it to cater for: I have a small company and have to give people my phone number. I don't want them to call me on my private phone. But at the same time, I hate having 2 phones with me!. I want to setup a server with a GSM Phone connected. I want all calls coming to this phone to be forwarded to my private phone [calls within the same network are free] I want all calls from my private phone coming to this phone to be answered and forwarded to a specified number in this fashion: Let's say the number attached to the server is 600123456 I want to dial on my private number 600123456w600654987 and call 600654987 through the phone connected to the server. I hope this scenario is clear. Thanks in advance Krystian -- Krystian Szczesny

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Does execute_extension terminate a javascript/ lua script?
by Neil Burgess 04 Sep '10

04 Sep '10

Hi, A quick question. If I use something like:- session.execute("execute_extension", "myExtension XML default"); inside a javascript file, should this return to the execute the following lines in the script once the extension has completed, or will the script terminate at that point. I know that is the case with a transfer, but not sure about the expected behaviour with an "execute_extension"? Similarly is this the case with an lua script? Many thanks, Neil

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Troubleshooting in high usage environments
by Bernhard Suttner 03 Sep '10

03 Sep '10

Hi, what is the best approach to debug SIP registration or the call dialplan processing on systems under very high usage? Its very difficult to view the complete log file for all users in debug mode because there are to much call attempts / sip registrations. Are there tools where I can turn on debugging only for a certain user or IP? It would be really cool to have an debugging interface which can be used to turn on debugging for a user on freeswitch. Maybe also in combination with not only one server but e.g. a "server farm" :-) Thanks in advance. Best regards, Bernhard Suttner

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Ubuntu Hi-Res timer - small world
by lloyd thomas 03 Sep '10

03 Sep '10

Went to my cousins party on saturday (Leeds UK) and bumped into a friend of hers, who just so happens know a bit about FS and has a couple of production boxes running FS for his wholesale carrier business. He recommended if I were running FS on ubuntu to recompile it, so that dynamic ticks were switch off and set the timer frequency to 1000 Hz. I have done that, but the CPU idle seems to have increased 5% to around 15% and spikes to 25% when I make a single call. Should that be expected on a lowly box with an old celeron CPU and 500mb RAM? model name : Celeron (Coppermine) stepping : 10 cpu MHz : 901.571 cache size : 128 KB MemTotal: 508316 kB MemFree: 276248 kB

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recording file without the terminator key sound
by Tony Tin 03 Sep '10

03 Sep '10

Hi, I try to record file with "record" application, it works fine. However, the sound of terminator key is also included at the end of the recorded file. I'm wondering whether there is a way get rid of it. Thanks. Regards, Tony

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Monitoring Valet Parking slots
by Tony Graziano 03 Sep '10

03 Sep '10

Is there a way to monitor a Valet Parking slot with blf? -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: tgraziano(a)voice.myitdepartment.net Fax: 434.984.8431 Email: tgraziano(a)myitdepartment.net LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpdesk(a)voice.myitdepartment.net Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec.

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Re: [Freeswitch-users] mod_freetdm dialing delay on analog line
by Moises Silva 03 Sep '10

03 Sep '10

On Thu, Sep 2, 2010 at 6:01 PM, Mark Maly <mark.maly(a)molcs.org> wrote: > Hello, > > > > Wondered why there would be an approximate 6-7 second delay in calling > analog/POTS via Sangoma USB100. Have used a TDM400P and there’s a delay, > but up to 7 seconds seems excessive. > > > > Any comments? > > > > 2010-09-02 16:56:28.586905 [DEBUG] mod_freetdm.c:427 FreeTDM/1:1/xxxxxxx > CHANNEL ROUTING > > 2010-09-02 16:56:28.586905 [DEBUG] switch_ivr_originate.c:66 > (FreeTDM/1:1/xxxxxxx) State Change CS_ROUTING -> CS_CONSUME_MEDIA > > 2010-09-02 16:56:28.586905 [DEBUG] switch_core_session.c:1039 Send signal > FreeTDM/1:1/xxxxxxx [BREAK] > > 2010-09-02 16:56:28.586905 [DEBUG] switch_core_state_machine.c:341 > (FreeTDM/1:1/xxxxxxx) State ROUTING going to sleep > > 2010-09-02 16:56:28.586905 [DEBUG] switch_core_state_machine.c:314 > (FreeTDM/1:1/xxxxxxx) Running State Change CS_CONSUME_MEDIA > > 2010-09-02 16:56:28.586905 [DEBUG] switch_core_state_machine.c:360 > (FreeTDM/1:1/xxxxxxx) State CONSUME_MEDIA > > 2010-09-02 16:56:28.586905 [DEBUG] switch_core_state_machine.c:360 > (FreeTDM/1:1/xxxxxxx) State CONSUME_MEDIA going to sleep > > 2010-09-02 16:56:30.264549 [DEBUG] ftmod_analog.c:720 [s1c1][1:1] Detected > tone DIAL on 1:1 > > 2010-09-02 16:56:30.284864 [DEBUG] ftdm_io.c:3041 [s1c1][1:1] Generating > DTMF [xxxxxxx] > > 2010-09-02 16:56:30.624887 [DEBUG] ftdm_io.c:2865 [s1c1][1:1] Queuing DTMF > 2 > > 2010-09-02 16:56:31.224598 [DEBUG] ftdm_io.c:2865 [s1c1][1:1] Queuing DTMF > 3 > > 2010-09-02 16:56:34.364947 [DEBUG] ftmod_analog.c:413 [s1c1][1:1] Changed > state from DIALING to UP > > > > Please pastebin the wanpipex.conf, freetdm.conf and freetdm.conf.xml files. You can also find more interactive help in the #sangoma IRC channel Moises Silva Senior Software Engineer Sangoma Technologies Inc. | NEW 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6 Canada t. 1 905 474 1990 x128 | e. moy(a)sangoma.com

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Help with [Exchange_Routing_Error]: is there a maximum number of transfers on a call?
by Victor Chukalovskiy 03 Sep '10

03 Sep '10

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Problems getting dtmf events
by Nigel Kent 03 Sep '10

03 Sep '10

I am trying to catch dtmf events but am having problems. It seems no dtmfs are being recieved. I can filter to other events and it gets picked up so this section of the code is doing the indented action of getting events. However I notice that during the execution of the playback command while the sound file is playing if a key press is done at this time the dtmf does get received. So my thought are what conditions does the playback command set in order to be able to have dtmf get recognized.

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