- FreeSWITCH-users - FreeSWITCH Mailing Lists

find out terminators for play_and_get_digits
by Tony Tin 06 Sep '10

06 Sep '10

Hi, Is there a way to find out the terminator which is used during "play_and_get_digits", thanks. Regards, Tony

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Re: [Freeswitch-users] Troubleshooting in high usage environments
by Bernhard Suttner 06 Sep '10

06 Sep '10

Hi, nice. Maybe a good solution, but I thought about a "freeswitch service". A client connects to a number of freeswitch servers and tell the freeswitch servers to "search for any activity" of user XYZ. If such a request comes in on mod_sofia and will be handeld in the dialplan and in every other module, the logging output is sent to the client. Something like a service interface. Best regards, Bernhard ----- Original Message ----- From: Michael Scheidell [mailto:michael.scheidell@secnap.com] To: freeswitch-users(a)lists.freeswitch.org Sent: Sat, 04 Sep 2010 00:04:43 +0200 Subject: Re: [Freeswitch-users] Troubleshooting in high usage environments > On 9/3/10 5:32 PM, Bernhard Suttner wrote: > > 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" :-) > I know in sipx there is a program that looks at logs, called sipviewer. > you can drill down to a certain sender/recipient if you want. > might work here. has X windows and MS windows versions. > > -- > Michael Scheidell, CTO > o: 561-999-5000 > d: 561-948-2259 > ISN: 1259*1300 > > *| *SECNAP Network Security Corporation > > * Certified SNORT Integrator > * 2008-9 Hot Company Award Winner, World Executive Alliance > * Five-Star Partner Program 2009, VARBusiness > * Best in Email Security,2010: Network Products Guide > * King of Spam Filters, SC Magazine 2008 > > > ______________________________________________________________________ > This email has been scanned and certified safe by SpammerTrap(r). > For Information please see http://www.secnap.com/products/spammertrap/ > ______________________________________________________________________

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2 endpoints behing NATs
by Stas Khirman 06 Sep '10

06 Sep '10

Hi, Sorry for novice question: Successfully setup FreeSWITCH on publicly accessible IP using 1.0.6 standard configuration. No changes in configuration, dial plan, etc. Basic tests (music,echo,etc) are working ok. However, the problem observed with call between two softphones (X-lite) behind NATs - call signal are coming through but no media. Both NATs are kind of typical household WiFi routers. Everything ok if one of softphones is connected without NAT. Does two NATs configuration supported by stock FreeSWITCH ? Can I make it work by changing some configuration parameters or I need something like RTP proxy or similar? Regards Stas

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mod_managed API interface question
by Malay Thakershi 05 Sep '10

05 Sep '10

Hello, Please let me know if this is feasible / possible. I need to make outgoing calls based on a scheduler using FS (SIP). I am planning to write a windows service that runs at particular time interval (say 10 minutes), checks who needs to be called, fires commands to originate these calls. I have a single FS command line running (where FS console is running). I am planning to fire FS commands such as: ---------------------- >managed apidemo a1 123 API CALL [managed(apidemo a1 123)] output: ApiDemo executed with args 'a1 123' and event type API. >managedrun apidemo a1 123 API CALL [managedrun(apidemo a1 123)] output: +OK 2008-10-08 17:44:25 [DEBUG] switch_cpp.cpp:1059 console_log() ApiDemo on a background thread #(3), with args 'a1 123'. 2008-10-08 17:44:25 [DEBUG] switch_cpp.cpp:1059 console_log() ExecuteBackground in apidemo completed. ---------------------- So I will pass some arguments to the mod_managed script and it will fire an API to originate call. My main question is: How can I communicate windows service to fire FreeSwitch commands? Thank you for any help or guidance.

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Video calls
by Boris Kovalenko 05 Sep '10

05 Sep '10

Hello! I can't get video calls working with current FS trunk and two eyeBeam 1.5. Is this even possible? May be I haven't enabled importrant module or something else? May someone provide a good example or point me to documentation page. -- With respect, Boris

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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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