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Call problems
by Jonas Gauffin 25 Oct '07

25 Oct '07

Hello I have three phones, all registered with FreeSwitch. I did the following: a) Called from phone A to phone B b) B picked up c) A put B on hold c) A called phone C d) C picked up e) Pressed transfer on A's phone Both B and C got disconnected instead of being connected to each other. I did another variant: A calls B, B calls C, B presses transfer. A things that he is still connected, B and C gets disconnected. Are these problems in my phones, or are they FS problems? //Jonas

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Re: [Freeswitch-users] Outbound gateway
by Anthony Minessale 24 Oct '07

24 Oct '07

you may also note that it's possible to setup a gateway that does not register by setting register to false in the gateway params. That will allow you to make calls using the gateway syntax and possibly provide authentication parameters on gateways that do not need you to register but do require authentication or a paticular from addr. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Jonas Gauffin <jonas.gauffin(a)gmail.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Wednesday, October 24, 2007 5:22:30 AM Subject: Re: [Freeswitch-users] Outbound gateway I can now answer my own questions is someone else is wondering: > 1. Do I need to configure anything more? Gateways that use no authentication do not need to be configured at all. Simply attach the gateway to addresses when calling. For instance: sofia/default/+462311111(a)sipgw1.thegateway.com > 2. How can I see if the registration to the gateway have succeeded You cant for unregistered gateways, since they arent registered =) > 3. Is it possible to debug just all proxy communication (and not all > internal sip communication)? Guess not. But you can set the SOFIA_DEBUG flag to 7, it will give you losts of information. > 4. I trying to call with: "pa call > sofia/mydomain.com/sipgw2.thegateway.com/4670123456", is that correct? The command can currently not be used do dial profiles, only to dial stuff that exists in the dialplan. If you want to try your gateway add something like this: <extension name="5556"> <condition field="destination_number" expression="5556"> <action application="bridge" data="sofia/mydomain.com/+4670111111@sipgw2.thegateway.com" /> </condition> </extension> Replace +4670111111 with a real phone number (my provider uses E.164 format). If you want to dial using a registered gateway, the format is this: sofia/<profilename>/gateway/<number> This stuff will also be put in the user book that we are writing. _______________________________________________ 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 __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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Outbound gateway
by Jonas Gauffin 24 Oct '07

24 Oct '07

Hello I got a couple of questions about sip and outbound gateways. I've configured a gateway in the sofia config: <gateways> <gateway name="sipgw2.thegateway.com"> <param name="realm" value="sipgw2.thegateway.com"> </gateway> </gateways> (I've replaced the real domain with a fake one) They are not using username/password but are instead limiting the access to my public ip address. 1. Do I need to configure anything more? 2. How can I see if the registration to the gateway have succeeded 3. Is it possible to debug just all proxy communication (and not all internal sip communication)? 4. I trying to call with: "pa call sofia/mydomain.com/sipgw2.thegateway.com/4670123456", is that correct? Thanks, Jonas

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getting started
by Patrick Aljord 21 Oct '07

21 Oct '07

Hey all, I'm trying freeswitch for the first time and I have a couple of questions: I have a SIP account from a provider that I use to make calls to cell phones and other, how can I use from freeswitch? Do I need to put the configurations (domain/user/password/proxy etc) in sofia.conf.xml or directory.xml and what's the difference? Once I get that configured, how can I transfer calls that are received to other numbers? When I use the console, I get plenty of messages such as: nta_agent: received garbage from udp/212.27.52.5:5060/sip is there a way to stop that? (it happened after I added my sip account to sofia.conf.xml) Is there a gui or someething to configure freeswitch? thanx in advance Pat

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Localized Numbering Plans
by Raymond Chandler 20 Oct '07

20 Oct '07

After seeing how many people responded to Brian's requests on localized say stuff, I figured I'd run this up the flagpole and see who salutes it.... I'd like to see FreeSwitch have localized numbering plans for many different areas out of the box. So, I've written a semi-complete version of a US numbering plan to spark some discussion on how others think it should be done.... and maybe get some examples written for other areas. If you notice any syntax errors, etc.... keep in mind this was written rather quickly and NOT thoroughly tested.... it's really just a (somewhat functional?) draft. ---- In Existing default_context ----  <extension name="Localized_Dialplan"> <condition field="${numbering_plan}" expression="^[[:alnum:]]+$"/> <condition field="destination_number" expression="^([[:alnum:]]+)$"> <action application="transfer" data="$1 XML ${numbering_plan}-Numbering-Plan"/> </condition> </extension> ---- in numberplans/US.conf.xml (needs to be included)---- <context name="US-Numbering-Plan">  <extension name="US_International"> <condition field="destination_number" expression="^011(\d*)$"> <action application="set" data="continue_on_fail=true"/> <action application="set" data="hangup_after_bridge=true"/> <action application="enum" data="1$1"/> <action application="bridge" data="${enum_auto_route}"/> <action application="bridge" data="sofia/$${domain}/011$1"/> </condition> </extension>  <extension name="US_LD"> <condition field="destination_number" expression="^1?([2-9]\d{2}[2-9]\d{6})$"> <action application="set" data="continue_on_fail=true"/> <action application="set" data="hangup_after_bridge=true"/> <action application="enum" data="1$1"/> <action application="bridge" data="${enum_auto_route}"/> <action application="bridge" data="sofia/$${domain}/1$1"/> </condition> </extension>  <extension name="US_Local"> <condition field="${default_area_code}" expression="\d{3}" continue="on-true"> <anti-action application="say" data="you must dial the area code to call this destination"/> </condition> <condition field="destination_number" expression="^([2-9]\d{6})$"> <action application="set" data="continue_on_fail=true"/> <action application="set" data="hangup_after_bridge=true"/> <action application="enum" data="1$1"/> <action application="bridge" data="${enum_auto_route}"/> <action application="bridge" data="sofia/$${domain}/1${default_area_code}$1"/> </condition> </extension>  <extension name="FCC_Services"> <condition field="^([4689]11)$"> <action application="bridge" data="sofia/$${domain}/$1"/> </condition> </extension> </context>

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Re: [Freeswitch-users] Tutorial on building an IVR?
by Vincent 19 Oct '07

19 Oct '07

At 10:41 18/10/2007 -0400, Brian West <brian.west(a)mac.com> wrote: >Try http://wiki.freeswitch.org/wiki/Javascript_Examples > >You write IVR's in JS or some other language. The dialplan isn't really >an IVR scripting language. Thanks guys for the info. I'll give it a shot. V.

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error during loading mod_openzap
by salzh 19 Oct '07

19 Oct '07

Hi, guys! During starting freeswitch, it reports as follows: 2007-10-19 11:11:09 [ERR] zap_io.c:1867 process_module_config() Cannot open wanpipe.conf 2007-10-19 11:11:09 [ERR] zap_io.c:1867 process_module_config() Cannot open zt.conf 2007-10-19 11:11:09 [ERR] zap_io.c:1925 zap_global_init() No modules configured! 2007-10-19 11:11:09 [ERR] mod_openzap.c:1201 mod_openzap_load() Error loading OpenZAP 2007-10-19 11:11:09 [CRIT] switch_loadable_module.c:726 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_openzap.so * both wanpipe.conf and zt.conf are already under the directory /usr/local/freeswitch/conf/, and i also add #include  #include  into freeswitch.xml

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Re: [Freeswitch-users] FeeSwitch doesn't start on windows 2003
by Hassan Firouzbakht 19 Oct '07

19 Oct '07

Thanks for the update. Is there any documented call flow for presence functionality and supported content-type? Please let me know. I am well familiar about all RFCs around SIMPLE (SIP for instant messaging and presence) and I can quickly validate the functionality with my sip client. Thanks, Hassan Date: Thu, 18 Oct 2007 21:15:28 -0400From: mike(a)jerris.comTo: freeswitch-users(a)lists.freeswitch.orgSubject: Re: [Freeswitch-users] FeeSwitch doesn't start on windows 2003I would suggest svn updating again, I just fixed a number of outstanding windows issues related to the release build. We have tested w/ xlite and it works more or less, except the mac version of xlite crashes. I think presence has been tested with polycom phones as well. On 10/18/07, Hassan Firouzbakht <hfirouz(a)hotmail.com> wrote: Thanks a lot. It works now. Do you suggest any client to test presence functionality? please let me know. I used counterPath and my sip client, it seems something is wrong. Thanks,Hassan Date: Thu, 18 Oct 2007 16:16:19 -0400From: mike(a)jerris.comTo: freeswitch-users(a)lists.freeswitch.orgSubject: Re: [Freeswitch-users] FeeSwitch doesn't start on windows 2003 Try svn update now and see if it works. There was an issue on machines with no ipv6. On 10/18/07, Michael Jerris <mike(a)jerris.com > wrote: I am seeing the same issue now.. must have just broke it. I am taking a look. On 10/18/07, Brian West < brian.west(a)mac.com > wrote: Are you using SVN as of about 10 seconds ago? if not update and try again.. Mike is in the process of revamping some of the MSVC build stuff. /b On Oct 18, 2007, at 1:18 PM, Hassan Firouzbakht wrote: Hello,I just downloaded freeSwitch and successfully compiled it with VS2005. Everytime I start the application the following message pops on screen. Please somebody tell me what is wrong in here and how it can be fixed. Thanks,Hassan WARNING: no real random source present!2007-10-18 11:11:01 [CONSOLE] switch_core.c:588 switch_core_init_and_modload() Bringing up environment.2007-10-18 11:11:01 [CONSOLE] switch_core.c:589 switch_core_init_and_modload() L oading Modules.Press any key to continue . . . Express yourself with free Messenger emoticons. Get them today! _______________________________________________ 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_______________________________________________ Freeswitch-users mailing listFreeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.free… Express yourself with free Messenger emoticons. Get them today!_______________________________________________Freeswitch-users mailing listFreeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.free… _________________________________________________________________ R U Ready for Windows Live Messenger Beta 8.5? Try it today! http://entertainment.sympatico.msn.ca/WindowsLiveMessenger

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FeeSwitch doesn't start on windows 2003
by Hassan Firouzbakht 19 Oct '07

19 Oct '07

Hello, I just downloaded freeSwitch and successfully compiled it with VS2005. Everytime I start the application the following message pops on screen. Please somebody tell me what is wrong in here and how it can be fixed. Thanks, Hassan WARNING: no real random source present!2007-10-18 11:11:01 [CONSOLE] switch_core.c:588 switch_core_init_and_modload() Bringing up environment.2007-10-18 11:11:01 [CONSOLE] switch_core.c:589 switch_core_init_and_modload() Loading Modules.Press any key to continue . . . _________________________________________________________________ Express yourself with free Messenger emoticons. Get them today! http://www.freemessengeremoticons.ca/?icid=EMENCA122

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Tutorial on building an IVR?
by Vincent 18 Oct '07

18 Oct '07

Hello I haven't found much information on how to write an Interactive Voice Response server with Freeswitch. Is there a tutorial somewhere, or is http://wiki.freeswitch.org/wiki/Getting_Started_Guide all there is at this point? Thank you V.

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