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Configure FS using Flowroute.com
by Alex Harris 20 Apr '09

20 Apr '09

I placed the xml found at http://wiki.freeswitch.org/wiki/SIP_Provider_Examples#Flowroute.com into the file \conf\directory\default\flowroute.com.xml Should I have placed it in another location? Thank you, Alex You'll have to do this http://wiki.freeswitch.org/wiki/SIP_Provider_Examples#Flowroute.com /b On Apr 20, 2009, at 5:15 PM, Alex Harris wrote: > I am new to VoIP and FS and I was hoping that someone could help me > configure my FS server. > > I have add flowroute.com.xml to C:\Program Files (x86)\FreeSWITCH > \conf\directory\default using the example on the wiki. > > I have updated vars.xml with > > <X-PRE-PROCESS cmd="set" data="default_provider=Flowroute.com"/> > <X-PRE-PROCESS cmd="set" > data="default_provider_username=*********"/> > <X-PRE-PROCESS cmd="set" data="default_provider_password=*******"/> > <X-PRE-PROCESS cmd="set" > data="default_provider_from_domain=Flowroute.com"/> >  > <X-PRE-PROCESS cmd="set" data="default_provider_register=true"/> > <X-PRE-PROCESS cmd="set" data="default_provider_contact=5000"/> > > Right now everytime I try to dial out I get a Gateway Invalid error > and I try to dial the DID and only get a busy tone. Any help would > be much appriciate. > > Thank you, > > Alex > _______________________________________________ > 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 Brian West brian(a)freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com

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Re: [Freeswitch-users] Configure FS using Flowroute.com
by Brian West 20 Apr '09

20 Apr '09

You'll have to do this http://wiki.freeswitch.org/wiki/SIP_Provider_Examples#Flowroute.com /b On Apr 20, 2009, at 5:15 PM, Alex Harris wrote: > I am new to VoIP and FS and I was hoping that someone could help me > configure my FS server. > > I have add flowroute.com.xml to C:\Program Files (x86)\FreeSWITCH > \conf\directory\default using the example on the wiki. > > I have updated vars.xml with > > <X-PRE-PROCESS cmd="set" data="default_provider=Flowroute.com"/> > <X-PRE-PROCESS cmd="set" > data="default_provider_username=*********"/> > <X-PRE-PROCESS cmd="set" data="default_provider_password=*******"/> > <X-PRE-PROCESS cmd="set" > data="default_provider_from_domain=Flowroute.com"/> >  > <X-PRE-PROCESS cmd="set" data="default_provider_register=true"/> > <X-PRE-PROCESS cmd="set" data="default_provider_contact=5000"/> > > Right now everytime I try to dial out I get a Gateway Invalid error > and I try to dial the DID and only get a busy tone. Any help would > be much appriciate. > > Thank you, > > Alex > _______________________________________________ > 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 Brian West brian(a)freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com

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Configure FS using Flowroute.com
by Alex Harris 20 Apr '09

20 Apr '09

I am new to VoIP and FS and I was hoping that someone could help me configure my FS server. I have add flowroute.com.xml to C:\Program Files (x86)\FreeSWITCH\conf\directory\default using the example on the wiki. I have updated vars.xml with <X-PRE-PROCESS cmd="set" data="default_provider=Flowroute.com"/> <X-PRE-PROCESS cmd="set" data="default_provider_username=*********"/> <X-PRE-PROCESS cmd="set" data="default_provider_password=*******"/> <X-PRE-PROCESS cmd="set" data="default_provider_from_domain=Flowroute.com"/>  <X-PRE-PROCESS cmd="set" data="default_provider_register=true"/> <X-PRE-PROCESS cmd="set" data="default_provider_contact=5000"/> Right now everytime I try to dial out I get a Gateway Invalid error and I try to dial the DID and only get a busy tone. Any help would be much appriciate. Thank you, Alex

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Problem with inbound calls ringing for multiple extensions
by winday 20 Apr '09

20 Apr '09

I want to config a system use extension 1001-1005. When a call come in, all these five extensions will ring together. But when one picks up the call,others will stop to ring. Right now, I can config all the inbound calls ring on extension 1001. I want to add other extensions in. The configuration in public.xml is like this: <extension name="teliax">  <condition field="destination_number" expression="^winday$">    <action application="transfer" data="$1 XML default"/> </condition> </extension> And the configuration in default.xml is as following: <extension name="Inbound">   <condition field="destination_number" expression="^winday$">   <action application="ring_ready" />  <action application="set" data="call_timeout=20"/>  <action application="bridge" data="sofia/sip/1001%${domain_name},sofia/sip/1002%${domain_name},sofia/sip/1003%${domain_name},sofia/sip/1004%${domain_name},sofia/sip/1005%${domain_name},sofia/sip/1006%${domain_name}"/>  <action application="javascript" data="/usr/local/freeswitch/scripts/answermachine.js"/> </condition> </extension> I got this log from my freeswitch console: 2009-04-20 15:09:56 [NOTICE] switch_channel.c:592 switch_channel_set_name() New Channel sofia/external/3521111111(a)8.17.37.23 [d5d30fce-2dde-11de-8a55-e3aa5c861d72] 2009-04-20 15:09:56 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing Cell Phone FL->winday in context public 2009-04-20 15:09:56 [NOTICE] switch_ivr.c:1345 switch_ivr_session_transfer() Transfer sofia/external/3521111111(a)8.17.37.23 to XML[$1@default] 2009-04-20 15:09:56 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing Cell Phone FL->$1 in context default 2009-04-20 15:09:56 [NOTICE] switch_ivr.c:1345 switch_ivr_session_transfer() Transfer sofia/external/3521111111(a)8.17.37.23 to enum[$1@default] 2009-04-20 15:09:56 [INFO] switch_core_state_machine.c:129 switch_core_standard_on_routing() No Route, Aborting 2009-04-20 15:09:56 [NOTICE] switch_core_state_machine.c:130 switch_core_standard_on_routing() Hangup sofia/external/3521111111(a)8.17.37.23 [CS_ROUTING] [NO_ROUTE_DESTINATION] 2009-04-20 15:09:56 [NOTICE] switch_core_session.c:1019 switch_core_session_thread() Session 21 (sofia/external/3521111111(a)8.17.37.23) Ended 2009-04-20 15:09:56 [NOTICE] switch_core_session.c:1021 switch_core_session_thread() Close Channel sofia/external/3521111111(a)8.17.37.23 [CS_DONE] Can anyone tell me what the problem is? -- View this message in context: http://www.nabble.com/Problem-with-inbound-calls-ringing-for-multiple-exten… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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G729 settings
by Nicolas Brenner 20 Apr '09

20 Apr '09

Merely for testing/research purposes I decided to try an "open" G729 codec posted to this list a couple months ago. I tried it a couple times with "FreeSWITCH Version 1.0.trunk (11356M)". And everything worked fine, while starting FS I would get: 2009-04-20 13:29:59 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_g729] 2009-04-20 13:29:59 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G729' (G.729) 8000hz 10ms 2009-04-20 13:29:59 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G729' (G.729) 8000hz 20ms 2009-04-20 13:29:59 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G729' (G.729) 8000hz 30ms 2009-04-20 13:29:59 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G729' (G.729) 8000hz 40ms 2009-04-20 13:29:59 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G729' (G.729) 8000hz 50ms 2009-04-20 13:29:59 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G729' (G.729) 8000hz 60ms 2009-04-20 13:29:59 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G729' (G.729) 8000hz 70ms 2009-04-20 13:29:59 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G729' (G.729) 8000hz 80ms 2009-04-20 13:29:59 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G729' (G.729) 8000hz 90ms 2009-04-20 13:29:59 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G729' (G.729) 8000hz 100ms 2009-04-20 13:29:59 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G729' (G.729) 8000hz 110ms 2009-04-20 13:29:59 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G729' (G.729) 8000hz 120ms Yesterday I wanted to update to the latest revision on svn, so I did make current, and after a few problems, cleaning everything up and getting the svn source again, I built the latest rev at the time (13083). Then I ran FS, everything seemed to work alright, but when I tried to make a call using G729 I got an error saying it didn't have the right read codec. After investigating a little bit, I discovered my VoIP provider was setting G729 @ 40ms, but FS was only supporting 10ms. This is what shows while starting FS: 2009-04-20 13:33:38 [CONSOLE] switch_loadable_module.c:889 switch_loadable_module_load_file() Successfully Loaded [mod_g729] 2009-04-20 13:33:38 [NOTICE] switch_loadable_module.c:182 switch_loadable_module_process() Adding Codec 'G729' (G.729) 8000hz 10ms 2009-04-20 13:33:38 [CONSOLE] switch_loadable_module.c:889 switch_loadable_module_load_file() Successfully Loaded [mod_amr] ..... There's no support for other configurations other than 10ms. The xml config files are the same, and the G729 module code is the same too, so I guess the only thing that's changed is the code that loads the codec module. Any ideas, help or similar experiences? Thanks! Nicolas

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Re: [Freeswitch-users] Ideas for my presentation
by mszlazak＠aol.com 20 Apr '09

20 Apr '09

Start by asking Tony what "pain" he was experiencing that made him create FreeSWITCH in the first place. What's posted about FS on Wikipedia does not go into it that much. I'm not an IT guy but I can _imagine_ how life _might_ be easier with FS. *Using FS requires less maintenance, like it crashes less. So on my days off I don't get as many calls to come to the office. I have more time with friends and family since using FS. There are less complaints from my supervisors and customers using the service and costs the company less, etc. * Get into more specific "pains" like,? "When it does crash it's quicker and easier to fix ..." * When it runs, it runs well because ... * I can upgrade and extend it easily ... I'm assuming that you pretty much got a handle on developer needs, but you get the idea. Very general pain issues about their lives then get more specific stuff. On the "vision thing", well I doubt FS is "the future of telephony" but that's because I doubt that voip is "the future of telephony." -----Original Message-----From: Diego Viola <diego.viola(a)gmail.com> To: mszlazak(a)aol.com Sent: Mon, 20 Apr 2009 4:40 am Subject: Re: [Freeswitch-users] Ideas for my presentation Hey guys, Any ideas on what I could show for the demo? I would like to show something really impressive that will make people go "omg, wow". I was thinking in showing a stress testing with sipp or something, and a real call on the background... to show how fast FS is, or to show how much load can handle. But I think I could just say that... I'd like to do something more impressive, any ideas? Diego On Sun, Apr 19, 2009 at 11:55 PM, Diego Viola <diego.viola(a)gmail.com> wrote: Yep, I'm also separating now everything that is FreeSWITCH and the business side into different sections. So I can talk about FreeSWITCH entirely first and then the business. Thanks everyone :) Diego On Sun, Apr 19, 2009 at 11:43 PM, <mszlazak(a)aol.com> wrote: The slides look nice but it's hard for me to judge much without their accompanying dialog. -----Original Message----- From: Diego Viola <diego.viola(a)gmail.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Sun, 19 Apr 2009 5:46 pm Subject: Re: [Freeswitch-users] Ideas for my presentation Ok, here is my current draft for the presentation. http://voip.richapplabs.com/freeswitch/freeswitch.pdf Enjoy :D PS: If you have any suggestions or feedback keep them coming, long life FreeSWITCH! Thanks, Diego On Sun, Apr 19, 2009 at 3:31 PM, bakko <asannucci(a)gmail.com> wrote: Me too :) ? we want slides! we want slides! we want slides! _______________________________________________ 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 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 Why pay full price? Check out this month's deals on the new AOL Shopping.

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Freeswitch Core dump error
by Baskar 20 Apr '09

20 Apr '09

Hi, I have installed Freeswitch 1.0.2 in CentOS 5.2 with audio code. For past one month i did not get any error in the Freeswitch.Today i get Four times segmentation Fault (core dump). I have paste the error in this path http://pastebin.freeswitch.org/8640. Can any one assest me to find what is the actuall error in the Freeswitch. This will help to find what is the error. Thanks in advance. -- Warm Regards, N.Baskar

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upper registration in FS?
by xbipin 20 Apr '09

20 Apr '09

i have been trying to do this but seem to have been lost, actually i want to use freeswitch as a session border controller so basically all the clients that try to register to FS will actually be authenticated by voipswitch but i want FS to be in between and proxy everything and with any of its advanced options turned off so as to simply work as topology hiding, proxy, SBC only. -- View this message in context: http://www.nabble.com/upper-registration-in-FS--tp22764757p22764757.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Skypiax as a windows service
by UV 19 Apr '09

19 Apr '09

Great work on Skypiax, Giovanni. We've tested it in our lab for sometime and it works very well. Unfortunately, when we tried deploying it on a production environment (running Win2K3 server farm), we ran into a barrier: 1. FS is running as terminal server console application (to be easily maintained remotely by RDP) 2. This is because Win2K does not allow RDP to access system console (session /userid 0) 3. Skype does not work on terminal server due to a well known disappearing audio drivers problem, therefore it has to run either as a console or a service (both on session 0). 4. FS can run well as a windows service 5. Skypiax seem to load as service, but it can't find the skype client and exit with the following error: 2009-04-13 20:54:14 [ERR] mod_skypiax.c:990 load_config() rev 13006M[00000000|37 ][ERRORA 990 ][skype_user ][-1, 0, 0] Failed to connect to a SKYPE API for interface_id=1, no SKYPE client running, please (re)start Skype client. Skypiax exiting This situation prevents me to run skypiax in production. I understand from the wiki page that windows service is not done yet - so I presume this is a predicted outcome. Any idea when and if this is planned to be implemented? Keep up the good work! Cheers, UV

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Event sockets and IPC
by David Knell 18 Apr '09

18 Apr '09

Hi - Consider a situation where I've two call legs up, each talking to a controlling process using an outbound event socket connection from FS. They each know the other's UUID. I'd like to send an arbitrary message from one to the other. I was going to build some sort of IPC thing, but it struck me that the elegant way to do it would be via FS' event mechanism - that way the messages come in to the processes in exactly the same way as, for example, DTMF indications. Problem I'm having is working out how to (and, indeed, whether one can) use sendevent to send a CUSTOM event to a specific UUID, and I was wondering if anyone would be kind enough to put me out of my misery one way or the other - or have I missed another mechanism for doing this? Cheers -- Dave

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