March 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

What is the best way implement the follow me for a extension
by Aloysius Lloyd 18 Mar '10

18 Mar '10

Hi All, I would like to setup follow me dialplan for individual extensions. Some extensions need a follow-me and some not. If the the follow-me not reach any live person the call should go to the FreeSWTICH voicemail.I have the extensions range from 201 -209. Also I am using the following Default dial plan to call between extensions. <extension name="Local_Extension"> <condition field="destination_number" expression="^(20[0-9])$"> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/>  <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/> <action application="set" data="ringback=${us-ring}"/> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="set" data="call_timeout=30"/>  <action application="set" data="hangup_after_bridge=true"/>  <action application="set" data="continue_on_fail=true"/> <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/>  <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> <action application="bridge" data="user/${dialed_extension}@ ${domain_name},${followme}"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/> </condition> </extension> I use the wiki samples to make a follow me dial plan for extension 201. Here is the dial plan. <extension name="201_followme"> <condition field="destination_number" expression="201"> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="ignore_early_media=true"/> <action application="set" data="call_timeout=10"/> <action application="bridge" data="sofia/$${domain}/201"/> <action application="set" data="call_timeout=13"/> <action application="bridge" data="sofia/gateway/voicepulse/14161235555" /> <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="default $${domain} 201"/> </condition> </extension> Call to extension 201 , there are two dialplan matching the request. Either Local_Extension or follwme_201. Which order freeswtich select? Thank you for your help. Thanks, Lloyd

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Re : how to dial out through a local sip endpoint - mv372 mobile > gateway adapter (rentmycoder rentmycoder)
by Steven Brown 18 Mar '10

18 Mar '10

Hi, You can do it with the Portech registered on FS as a regular endpoint then bridge to it as below or similar <extension name="gsm"> <condition field="destination_number" expression="^9(.*)$"> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="sofia/internal/1005^$1@${domain}"> where the Portech registers as 1005 and I prefix the number I wish to dial with 9 or alternatively just bridge directly to it without having to register <action application="bridge" data="sofia/default/$1@192.168.1.3:5060"/> Where 192.168.1.3 is the Portech In both cases you need to configure the Portech Lan to Mobile Table as below Item URL Call Num 0 192.168.1.2 # where 192.168.1.2 is my FS box Steve > Message: 5 > Date: Thu, 18 Mar 2010 13:21:09 +0100 > From: rentmycoder rentmycoder <rentmycoder(a)gmail.com> > Subject: [Freeswitch-users] how to dial out through a local sip > endpoint - mv372 mobile gateway adapter > To: freeswitch-users(a)lists.freeswitch.org > Message-ID: > <50e456911003180521je008a2dme94fcc6901a04b99(a)mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > Hi guys, > Freeswitch rocks! > I've registered an Portech mv-372 mobile adapter to FS as an internal > user as userid/extension=5000. > How to dial out through this endpint? > > In asterisk I've used Dial(sip/1234567@5000) where 1234567 is the > phone no to call... > > In FS: bridge(sophia/internal/1234567@5000) > mod_dptools: normal_temporery_failure > > any ideas how to do the bridge? > or should I setup the adapter as a gateway profile? > > Just to make sure the adapter is ok i've tested it using asterisk and > it worked and was set up like this: > http://www.voip-info.org/wiki/view/Setup+MV-370+GSM+Gateway+with+Asterisk > > > > ------------------------------ > > Message: 6 > Date: Thu, 18 Mar 2010 08:22:10 -0400 > From: "Todd" <phunk0000(a)hotmail.com> > Subject: [Freeswitch-users] FS Best practices > To: "FreeSWITCH-users@lists. freeswitch. org " > <FreeSWITCH-users(a)lists.freeswitch.org> > Message-ID: <BLU110-DS11B84F16B12464595A3B07A72B0(a)phx.gbl> > Content-Type: text/plain; charset="us-ascii" > > Hey List, still kinda new to this whole deal, looking for some advice. We > are building a website that needs to interact with FS. Not looking for > specific instructions just opinions about the best practices for doing > things like click to dial from a web page, calling CDR records to web > applications to display customers stats and general web integration. Does > this all need to be done using scripting languages like Lua or Perl, or can > you directly integrate FS with a website in some other manner. Again, just > getting into all this, but having a blast learning FS, config and power > rock! Anyhow, some places or things I should start looking into would be > greatly appreciated. Thanks List! > >

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how to dial out through a local sip endpoint - mv372 mobile gateway adapter
by rentmycoder rentmycoder 18 Mar '10

18 Mar '10

Hi guys, Freeswitch rocks! I've registered an Portech mv-372 mobile adapter to FS as an internal user as userid/extension=5000. How to dial out through this endpint? In asterisk I've used Dial(sip/1234567@5000) where 1234567 is the phone no to call... In FS: bridge(sophia/internal/1234567@5000) mod_dptools: normal_temporery_failure any ideas how to do the bridge? or should I setup the adapter as a gateway profile? Just to make sure the adapter is ok i've tested it using asterisk and it worked and was set up like this: http://www.voip-info.org/wiki/view/Setup+MV-370+GSM+Gateway+with+Asterisk

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Fscomm unable to detect devices
by Rob Dyck 18 Mar '10

18 Mar '10

16987 on Fedora I realize fscomm is an immature project. I get an error message when it tries to detect my devices ( pci sound card, usb handset, usb camera with microphone ). Is this the expected result? Detection works in Twinkle and Ekiga. Second question - is video support on the roadmap?

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Release Cause in response message
by Tihomir Culjaga 18 Mar '10

18 Mar '10

hello folks, anyone knows how to pass a Release cause/ Disconnect Reason when firing respond from a dialplan ... e.g.: Ive tried with sip_h_Reason=Q.850;cause=27 but it is not working :) <extension name="parseResult"> <condition field="destination_number" expression="^parseResult$"/> <condition field="${componentStatus}" expression="^1$" break="on-true"> <action application="log" data="INFO ######################## CALL FAILED REASON 1' ##########\n"/> <action application="set" data="sip_h_Reason=Q.850;cause=16"/> <action application="respond" data="480 Temporarily Unavailable"/> </condition> </extension> any hint?

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Re: [Freeswitch-users] Skype for SIP - No SILK support, yet...
by Madovsky 18 Mar '10

18 Mar '10

But Brian, are you ready to answer to 40 millions users if the bug is not resolved ? :D ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Wednesday, March 17, 2010 12:44 PM Subject: Re: [Freeswitch-users] Skype for SIP - No SILK support, yet... I wish also for you guys... ----- Original Message ----- From: Brian West To: freeswitch-users(a)lists.freeswitch.org Sent: Wednesday, March 17, 2010 12:33 PM Subject: Re: [Freeswitch-users] Skype for SIP - No SILK support, yet... Wish our FreeSWITCH users would do the same. Seems we are close to a release and a flood of bugs come in. :P /b On Mar 17, 2010, at 11:29 AM, Madovsky wrote: I think when you have millions of users they need more attention before to release anything. I know a guy who works for google chrome project he said every bug there are 40 millions people who are waiting to test new version... so it's better to be sure that the bug is well corrected... ---------------------------------------------------------------------------- _______________________________________________ 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

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Re: [Freeswitch-users] freeswitch to cisco gateway
by Madovsky 18 Mar '10

18 Mar '10

Also Reddy, FS is protected (correct me if I'm wrong) under MPL licence http://www.mozilla.org/MPL/MPL-1.1.html in the article 3, paragraph 2: 3.2. Availability of Source Code. Any Modification which You create or to which You contribute must be made available in Source Code form under the terms of this License either on the same media as an Executable version or via an accepted Electronic Distribution Mechanism to anyone to whom you made an Executable version available; and if made available via Electronic Distribution Mechanism, must remain available for at least twelve (12) months after the date it initially became available, or at least six (6) months after a subsequent version of that particular Modification has been made available to such recipients. You are responsible for ensuring that the Source Code version remains available even if the Electronic Distribution Mechanism is maintained by a third party. so my opinion is you are infringing the Licence Regards Franck ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Tuesday, March 16, 2010 12:00 PM Subject: Re: [Freeswitch-users] freeswitch to cisco gateway so share your customized version to the community and maybe we ll be glad to help you.... this is the principle of open source ----- Original Message ----- From: srinivasula reddy To: freeswitch-users Sent: Tuesday, March 16, 2010 11:49 AM Subject: Re: [Freeswitch-users] freeswitch to cisco gateway we started customizing Freeswitch1.0.2 2yrs back, so now we are unable to use new versions. srinivas On Mon, Mar 15, 2010 at 9:06 PM, srinivasula reddy <srinivas.ksvreddy(a)gmail.com> wrote: Hi, i am using freeswitch1.0.2, farwarding call from freeswitch to cisco like just farwading the call. <action application="bridge" data=sofia/internal/${destination_number}@ciscogatewayip/> i do not using any gateway here, just farwarding call to router, is this correct way of doing? any idea -- Srinivasula Reddy K -- Srinivasula Reddy K ---------------------------------------------------------------------------- _______________________________________________ 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

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How long did the phone ring?
by Fraser Redmond 18 Mar '10

18 Mar '10

I'm converting a call-center app from Asterisk to FreeSwitch (using xml and javascript dialplans) and I think I've worked out how to do nearly everything, except for tracking one important metric: How long the phone rang before an agent picked it up. I'm not even entirely sure what to call it in FreeSwitch lingo - so that would be a good starting point :-) (Media-duration maybe?) I can work out the length of ringing for calls that didn't connect, by logging when the bridge is attempted, and after it fails and comparing the time. But if the person being called picks up, then all I know from that is the total length of the call including the ringing/media time, not the media time and connected time as 2 separate values The info application gives several useful looking variables like: Caller-Channel-Progress-Media-Time Caller-Channel-Transfer-Time ...but they're still set to zero. I thought I might be able to subtract Caller-Channel-Created-Time from Caller-Channel-Answered-Time to get the milliseconds of how long the phone rang for - but there's no reliable correlation between the difference and how long the phone rang. In asterisk I achieved this by logging when I was about to make a call, then running a macro when the Dial() connected, like so: exten => s,n,Macro(Dial_Attempted) exten => s,n,Dial(SIP/1001,20,M(Dial_Succeeded)) exten => s,n,Macro(Dial_Failed) ;Only gets here if the Dial didn't connect. Is there any way to do this in FreeSwitch? Cheers, Fraser

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Re: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA Support [Resolved]
by Gabriel Kuri 18 Mar '10

18 Mar '10

We resolved the issue with Broadsoft SCA and inbound phone calls as it relates to this thread. If you're using a domain name rather than IP address for $${domain} in vars.conf, you need to set the sip_invite_domain on the bridge app in the dialplan, otherwise the INVITEs from FreeSWITCH end up being from the IP address, which screws up SCA. For example, we set sip_invite_domain=$${domain} on the bridge app for local extensions in the dialplan ... <action application="bridge" data="{sip_invite_domain=$${domain}}user/${dialed_extension}@${domain_name}"/> Shouldn't this be part of the default config in SVN ? Cheers, Gabe On Mon, Mar 1, 2010 at 11:01 PM, Brian West <brian(a)freeswitch.org> wrote: > You have to watch phones that might do the switch on you... like resolve the IP then do the subscribe incorrectly... that could happen. > > Verify the subscribe packet... > > /b > > On Mar 1, 2010, at 11:49 PM, Gabriel Kuri wrote: > >> I've torn my hair out for hours and I still can't get it to work with >> the domain name set everywhere <sigh> ... >> >> I can get it to work if I hardcode the IP address on the phones and in >> all the config files, but I can't get the domain name to work on >> incoming calls, I'm at a total loss. If I had a mismatch somewhere, >> I'd think it wouldn't work on outgoing calls either, but that seems to >> work fine. >> >> Any other ideas? Might it have anything to do with the incoming call >> coming in from the PSTN and going through the external profile? >> >> Cheers, >> Gabe > > > _______________________________________________ > 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 >

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[roblem with reloadxml command
by false 18 Mar '10

18 Mar '10

Hi List i am new to freeswitch, i do the reloadxml in Freeswitch console and get the error: freeswitch@internal> reloadxml 2010-03-17 15:55:22.694226 [ERR] switch_xml.c:1297 Couldnt open /usr/local/freeswitch/conf/dialplan/internal/*.xml (No such file or directory) +OK [Success] Please help Ha` __________________________________________________________ Thử dùng Yahoo! Messenger 10 Beta mới. Nay với tính năng gọi video, xem cập nhật và nhiều tính năng khác! Xem thêm tại http://vn.messenger.yahoo.com/

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