April 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Users : Alphanumeric mapping
by Jérôme M. 24 Apr '10

24 Apr '10

Hi everyone, I'm new to Freeswitch and just started to configure it. I have read the wiki, and my first problem is the following : - I want to use alphanumeric id for my users. In other words, they can register to the PBX with a id like firstname.lastname(a)domain.com and can be reached in this way, no need to know the phone number (300-399 in my case). So, i declared my users as mentionned on the wiki, using number-alias. In my dialplan, every call to 300-399 numbers are automatically associated to alphanumeric id's, everything is working fine. But in a real context, every user will only know the alphanumeric-id (it's easier to remember) of another users. I thought that the mapping was bidirectionnal, but it seems that I have to match every firstname.lastname(a)domain.com in my dialplan. I tried, but apparently destination_number contains only firstname.lastname ( no SIP URI, maybe because I use the local domain). I' m sure I could do it using a "dirty trick" but i really want to do it in a good way... Do you have any ideas ? Thank you so much for your help, sorry for my approximative english ;) ! Kind regards, Jerome

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Can FreeSWITCH act as a SIP Proxy with transcoding?
by Fabio Pietrosanti (naif) 24 Apr '10

24 Apr '10

Hi all, i would like to know if FreeSWITCH could be used as a simple SIP Proxy in order to: - receive SIP/TLS connections - receive SRTP secured connection - forward it in "unsecured" way to a backend PBX (plain RTP, plain SIP/UDP) - if required configured do simply transcoding before forwarding calls to backend PBX So to run it as a "SIP Security Proxy" in front of an existing SIP PBX doing also transcoding if required (in case backend SIP PBX does not support SIP UA codecs) . Is this setup feasible with a FreeSWITCH? Does anyone know if cudatel (https://www.cudatel.com/) support such kind of configuration scenario? Fabio

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How to set maximum ring duration if call is not answered and fake ringing tone
by Frank Church 24 Apr '10

24 Apr '10

How do you set the maximum ring duration for a call if it is not anwered? How do you set a fake ring tone, if there is a delay before call begins to ring? -- Frank Church ======================= http://devblog.brahmancreations.com

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Parsing XML files from Spidermonkey
by Alberto Escudero 24 Apr '10

24 Apr '10

Hi, After one year using FS i am starting to like XML so i am trying to get a Javascript script to read local XML files. I am using the XML method and getting Syntax errors from spidermonkey While something like this works: xmldata = new XML("<tag>foo</tag>"); I have not been able to read and parse XML local files, using File or FileIO methods A simple example like this returns Syntax error. var foo = apiExecute ("show", "channels as xml"); xmldata = new XML(foo); Has anyone managed to use the XML method from spidermonkey to read a XML stored file? There are some E4X bugs around and i wonder if those are the cause of the "Syntax Error" feedback even reading a very basic XML file -- Stopping junk mailers is good for the environment

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Transferring to Non-Existent Extension
by Jerry Richards 24 Apr '10

24 Apr '10

Is there a "built-in" way for a transferor to recover a transferee who was transferred to a non-existent extension? Any examples of this? Right now, FS will send a BYE to the transferee with cause "NO_ROUTE_DESTINATION". Best Regards, Jerry

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outbound sip call: NO_USER_RESPONSE
by Neil Patel 24 Apr '10

24 Apr '10

I'm trying to make a sip call using Gizmo. I set up the profile as specified here <http://wiki.freeswitch.org/wiki/Provider_Configuration:_Gizmo> and dropped it into /conf/sip_profiles/external/. I can see that the profile is registered when I check sofia status from CLI: Name Type Data State ================================================================================================= ... external::gizmo gateway sip:otalo@proxy01.sipphone.com<sip%3Aotalo(a)proxy01.sipphone.com> REGED ... ================================================================================================= On a call event, I invoke the lua commands: sessiondata = "sofia/gateway/gizmo/<num>" new_session = freeswitch.Session(sessiondata) >From this I'm getting a NO_USER_RESPONSE error: 2010-04-21 11:31:16.169900 [DEBUG] sofia.c:4153 Channel sofia/internal/ 1234(a)conference.freeswitch.org entering state [terminated][480] 2010-04-21 11:31:16.169900 [NOTICE] sofia.c:4789 Hangup sofia/internal/ 1234(a)conference.freeswitch.org [CS_CONSUME_MEDIA] [NO_USER_RESPONSE] 2010-04-21 11:31:16.169900 [DEBUG] switch_channel.c:2071 Send signal sofia/internal/1234(a)conference.freeswitch.org [KILL] 2010-04-21 11:31:16.169900 [DEBUG] switch_core_session.c:1018 Send signal sofia/internal/1234(a)conference.freeswitch.org [BREAK] 2010-04-21 11:31:16.169900 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1234(a)conference.freeswitch.org) Running State Change CS_HANGUP 2010-04-21 11:31:16.170748 [DEBUG] switch_core_state_machine.c:499 (sofia/internal/1234(a)conference.freeswitch.org) State HANGUP 2010-04-21 11:31:16.170748 [DEBUG] mod_sofia.c:408 sofia/internal/ 1234(a)conference.freeswitch.org Overriding SIP cause 408 with 480 from the other leg 2010-04-21 11:31:16.170748 [DEBUG] mod_sofia.c:414 Channel sofia/internal/ 1234(a)conference.freeswitch.org hanging up, cause: NO_USER_RESPONSE 2010-04-21 11:31:16.170748 [DEBUG] switch_core_state_machine.c:46 sofia/internal/1234(a)conference.freeswitch.org Standard HANGUP, cause: NO_USER_RESPONSE 2010-04-21 11:31:16.170748 [DEBUG] switch_core_state_machine.c:499 (sofia/internal/1234(a)conference.freeswitch.org) State HANGUP going to sleep 2010-04-21 11:31:16.170748 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/1234(a)conference.freeswitch.org) State Change CS_HANGUP -> CS_REPORTING 2010-04-21 11:31:16.170748 [DEBUG] switch_core_session.c:1018 Send signal sofia/internal/1234(a)conference.freeswitch.org [BREAK] 2010-04-21 11:31:16.171750 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1234(a)conference.freeswitch.org) Running State Change CS_REPORTING 2010-04-21 11:31:16.171750 [DEBUG] switch_core_state_machine.c:590 (sofia/internal/1234(a)conference.freeswitch.org) State REPORTING 2010-04-21 11:31:16.171750 [DEBUG] switch_core_state_machine.c:53 sofia/internal/1234(a)conference.freeswitch.org Standard REPORTING, cause: NO_USER_RESPONSE 2010-04-21 11:31:16.171750 [DEBUG] switch_core_state_machine.c:590 (sofia/internal/1234(a)conference.freeswitch.org) State REPORTING going to sleep 2010-04-21 11:31:16.171750 [DEBUG] switch_core_state_machine.c:327 (sofia/internal/1234(a)conference.freeswitch.org) State Change CS_REPORTING -> CS_DESTROY 2010-04-21 11:31:16.171750 [DEBUG] switch_core_session.c:1018 Send signal sofia/internal/1234(a)conference.freeswitch.org [BREAK] 2010-04-21 11:31:16.171750 [DEBUG] switch_core_session.c:1161 Session 4 (sofia/internal/1234(a)conference.freeswitch.org) Locked, Waiting on external entities 2010-04-21 11:31:16.177790 [DEBUG] switch_ivr_originate.c:3228 Originate Resulted in Error Cause: 18 [NO_USER_RESPONSE] *2010-04-21 11:31:16.177790 [ERR] mod_conference.c:4563 Cannot create outgoing channel, cause: NO_USER_RESPONSE* 2010-04-21 11:31:16.177790 [NOTICE] mod_conference.c:4566 Hangup sofia/internal/1001(a)server.IP [CS_EXECUTE] [NO_USER_RESPONSE] 2010-04-21 11:31:16.177790 [NOTICE] switch_core_session.c:1179 Session 4 (sofia/internal/1234(a)conference.freeswitch.org) Ended 2010-04-21 11:31:16.177790 [NOTICE] switch_core_session.c:1181 Close Channel sofia/internal/1234(a)conference.freeswitch.org [CS_DESTROY] 2010-04-21 11:31:16.177790 [DEBUG] switch_core_state_machine.c:428 (sofia/internal/1234(a)conference.freeswitch.org) Running State Change CS_DESTROY 2010-04-21 11:31:16.177790 [DEBUG] switch_channel.c:2071 Send signal sofia/internal/1001(a)server.IP [KILL] 2010-04-21 11:31:16.177790 [DEBUG] switch_core_state_machine.c:439 (sofia/internal/1234(a)conference.freeswitch.org) State DESTROY 2010-04-21 11:31:16.177790 [DEBUG] mod_sofia.c:341 sofia/internal/ 1234(a)conference.freeswitch.org SOFIA DESTROY 2010-04-21 11:31:16.177790 [DEBUG] switch_core_session.c:1018 Send signal sofia/internal/1001(a)server.IP [BREAK] Strangely, I get the same error even if I put in some arbitrary gateway name in the lua code, so it seems like possibly the profile isn't being properly read. I also thought that perhaps I should be using "sofia/gateway/external::gizmo/<num>" instead of "../gizmo/..", but that didn't seem to work. I haven't made any other changes to conf files, though perhaps I'm missing something? Thanks in advance, Neil

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Re: [Freeswitch-users] Grandstream gateways
by Samuel Mukoti 23 Apr '10

23 Apr '10

Thank you for those tips, I do have some small setups using gxw4108 they work or, except CID doesn't seem to work. I will try the channel bank route - just don't know too much about the setup options or how you'd purchase the correct config, eg. For 150 FXS channel bank, can I get a single PCI card for that? I may end up using the grandstream fxs gateways then use the T1 channel bank from sangoma, Thank you all.. Lastly, I know asterisk now has an offical skype_ module, Is there anything similar I could use? On 25 Nov,2009, at 9:52 PM, Cory Andrews <cory(a)voipsupply.com> wrote: > Samuel - you could go with FXS gateways or channel banks. If you go > the gateway route Grandstream or Audiocodes would work fine. > Audiocodes are a bit more telco grade. If you have 25 POTS incoming > you could use a 24FXO channel bank cross connected with Rhino T1 > cards, or individual FXO gateways but you may have a hard time > finding 24 ports of FXO in a single GW. Best performing T1 cards in > my experience (thousands of deployments) are Sangoma. Your server > configuration looks fine. > > Cory J. Andrews > Director New Market Initiatives > > Sayers Media Group > VoIP Supply, LLC > 454 Sonwil Drive > Buffalo, NY 14225 > 716-250-3402 OFFICE > 716-630-1548 FAX > 716-601-4474 MOBILE > candrews(a)sayersmedia.com > > > Have I exceeded your expectations? Please share your experience > with my boss, Benjamin P. Sayers, CEO > > NOTICE: The information contained in this email and any document > attached hereto is intended only for the named recipient(s). It is > the property of the VoIP Supply, LLC and shall not be used, > disclosed or reproduced without the express written consent of VoIP > Supply, LLC. If you are not the intended recipient, nor the employee > or agent responsible for delivering this message in confidence to > the intended recipient(s), you are hereby notified that you have > received this transmittal in error, and any review, dissemination, > distribution or copying of this transmittal or its attachments is > strictly prohibited. If you have received this transmittal and/or > attachments in error, please notify me immediately by reply e-mail > or telephone and then delete this message, including any > attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY > 14225 USA. > > > > -----Original Message----- > From: freeswitch-users-bounces(a)lists.freeswitch.org > [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of > Samuel Mukoti > Sent: Wednesday, November 25, 2009 2:40 PM > To: freeswitch-users(a)lists.freeswitch.org > Subject: [Freeswitch-users] Grandstream gateways > > Hi all, > > I'm wanting to try out a my first large scale setup at the office, 200 > extensions and 24 POTS incoming, also a T1 line once the telco guys > are ready. I wanted assistance with choosing the most appropriate > hardware. We already have about 150 analogue phones, and I was > wondering what's best? A couple of grandstream FXS GXW4024? Also for > my POTS lines, gxw4108 FXO gateway or is it better to buy a sangoma > or digium card? The best voice quality is paramount. Lastly for T1 > what cards are recommeded, > > I was also proposing to use a Dell T116 Quad core intel i7 8G DRAM, > would that perform? Or do I need hardware transcoding? > > Thank you, > > Sam > > Twitter: twitter.com/samuelmukoti > > > On 25 Nov,2009, at 8:05 PM, freeswitch-users-request(a)lists.freeswitch.org > wrote: > >> Send FreeSWITCH-users mailing list submissions to >> freeswitch-users(a)lists.freeswitch.org >> >> To subscribe or unsubscribe via the World Wide Web, visit >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> or, via email, send a message with subject or body 'help' to >> freeswitch-users-request(a)lists.freeswitch.org >> >> You can reach the person managing the list at >> freeswitch-users-owner(a)lists.freeswitch.org >> >> When replying, please edit your Subject line so it is more specific >> than "Re: Contents of FreeSWITCH-users digest..." >> >> >> Today's Topics: >> >> 1. Re: mod_conference kick to abort invitations (Michael Jerris) >> 2. Re: Handling the 302 Moved Temporarily response from >> JavaScript (Michael Jerris) >> 3. Re: No NOTIFY MWI when registering via proxy. (Brian West) >> 4. Re: remote_media_ip variable not set (Michael Jerris) >> 5. Re: How to find whether the destination extension supports >> encryption (Michael Jerris) >> 6. Re: Bypass_media and re_invite (srinivasula reddy) >> 7. Re: Handling the 302 Moved Temporarily response from >> JavaScript (Stephen Crosby) >> 8. Re: Handling the 302 Moved Temporarily response from >> JavaScript (Tihomir Culjaga) >> >> >> --- >> ------------------------------------------------------------------- >> >> Message: 1 >> Date: Wed, 25 Nov 2009 12:44:46 -0500 >> From: Michael Jerris <mike(a)jerris.com> >> Subject: Re: [Freeswitch-users] mod_conference kick to abort >> invitations >> To: freeswitch-users(a)lists.freeswitch.org >> Message-ID: <1CCC981C-9F4A-4D97-ACEA-A6DFB906C32B(a)jerris.com> >> Content-Type: text/plain; charset="windows-1252" >> >> Its a feature we don't have, patches welcome. >> >> Mike >> >> On Nov 24, 2009, at 5:35 PM, Jan Thiemo Fricke wrote: >> >>> Hi members, >>> I?m controlling freeswitch with the conference module via xmlrpc. >>> >>> Is it desired that the kick command can only kick users that are >>> connected to the conference? >>> Is there no chance abort an invitation? >>> The kick command has no effect until the person I invited with the >>> dial command is connected. >> >>

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ODBC DB Support in Windows
by Phillip Jones 23 Apr '10

23 Apr '10

Hi there, I am trying to set up ODBC support on Win2003. I have created and tested a system DSN and added this to sofia.conf.xml in the <global_settings> section: *<param name="odbc-dsn" value="FS:xx:password"/>* Is there more to do? Freeswitch starts with no problem - seemingly ignoring this request to use ODBC. What am I missing?? Phil

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single participant timeout for conference
by Woody Dickson 23 Apr '10

23 Apr '10

Hi, Is there anyway to configure conference in such a way that if there is only one participant in the conference, that single conference participant will be hung up after XX seconds? Woody

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recommended sip phone or adapter
by vip killa 22 Apr '10

22 Apr '10

I have a freeswitch server running at one location. I want to setup a sip phone at another location (outside the local freeswitch network). can someone recommend a sip phone or adapter that is not too expensive and will easily connect/work with the freeswitch server at location one?

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