- FreeSWITCH-users - FreeSWITCH Mailing Lists

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?

1 1

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

3 3

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

5 5

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

6 14

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 >

2 1

[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/

2 1

Newbie question on freeswitch
by false 17 Mar '10

17 Mar '10

HI list i am new to Freeswitch, i am successfully install freeswitchand i use the default configi use 1000 and 1001 to register successfull and make call between 2 endpoints fine after that, i create a new 8000.xml file which in the same location as 1000.xmlfreeswitch:/usr/local/freeswitch/conf/directory# cat default/8000.xml<include> <user id="8000"> <params> <param name="password" value="$${default_password}"/> <param name="vm-password" value="1000"/> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="8000"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Extension 8000"/> <variable name="effective_caller_id_number" value="8000"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> <variable name="callgroup" value="techsupport"/> </variables> </user></include> the 8000 is registered successfull but i cannot use the 1000 call to 8000 please helpHa` Cảm giác an toàn hơn khi online. Nâng cấp lên Internet Explorer 8 mới, an toàn hơn,tối ưu hóa cho Yahoo!. Hoàn toàn miễn phí. Tải IE8 tại đây! http://downloads.yahoo.com/vn/internetexplorer/

4 3

Call parking question/best practice advice
by Tom Christensen 17 Mar '10

17 Mar '10

Hello, Pretty new, but so far very happy with freeswitch! The setup and config is really nice, straightforward, and I appreciate the design of the system. I've worked with Asterisk pretty regularly for the last 4-5 years, and am really looking for something more stable, and I think I've found it here! One issue I've run into, and I'm not sure if its just that my use case is abnormal, hence the email. I've worked with PBX systems for a number of years (11 now...), and in all of the systems I've worked with, call parking was kind of a "roaming" feature. By this I mean, you think party X (the intended recipient of the call) is in the building, but they are not at their desk. So, you park the call, and then page, or by some other means attempt to locate party X and tell them "get to the nearest phone and pick up 5901". If you can't locate party X or you can't find them in time, the call should time out back to the parking party (receptionist) to be handled appropriately (message taken, sent to voicemail, transferred to party X's boss) This scenario requires a few things: 1) the receptionist/person who actually receives the call needs to be able to put the call in a specific place (extension) that doesn't change 2) Party X (the intended recipient) needs to be able to access that specific place (extension) from any phone inside the company 3) Order must be preserved (IE, the mod_fifo call park in freeswitch is inadequate, because if the receptionist parks 2 people on 5900, one for party X and one for party Y, in that order, but party Y gets to a phone first, they will be connected to party X's caller) 4) The receptionist has to be protected from or provided with sufficient information to avoid connecting 2 customers with each other (IE the valet_park in freeswitch is inadequate, because if she parks a caller for X on 6001, then parks a caller for Y on 6002, then a caller for Z calls in, how can she know if 6001 is free yet? valet_park doesn't support BLA/SLA so what can she do? What if she just forgets and sends that caller to 6001? Then the 2 customers are bridged together and thats a big no-no) Also, I haven't seen a way to make valet_park timeout? But maybe I'm wrong there... The closest I can get to implementing the above scenario is by using multiple mod_fifo queues (one for each parking spot), and SLA/BLA on the receptionist phone to "notify" her that someone is already parked in a certain queue. Unfortunately, there is no enforcement of this policy by the system, and if she fat fingers her transfer, she will double up a queue, creating the problem indicated in 3 above. The other shortcoming of this setup is that without a large phone (IE, with 10-15 SLA/BLA slots) it becomes impossible to appropriately handle call park. If the receptionist is away from her desk and attempts to answer a call, she can't tell which parking spots are free at all on a regular 2 line phone. Also, regular users are prohibited from using call park as they cannot know which slots are free (the need for this arises often in certain industries, such as law, client is speaking with attorney, attorney finishes and says "let me connect you with my assistant to schedule an appointment and handle x, y, and z for you.", paralegal is not at desk, atty needs to park call and page his assistant, maybe they are in the copy room, or the library doing research...) The asterisk parking lot seems to solve this problem quite elegantly (although the stability issues of asterisk undermine it). It is simple to use (receptionist and everyone else only needs to remember 1 extension to park any call), The callers/parking spots are maintained by the system to prevent double parking, and each caller sits at a specific extension until they are picked up or time out. So, the question boils down to how do people using freeswitch view/use call park? Is there a "better" roaming feature that I just don't know about? If it is just something that hasn't been implemented/thought about this way yet, any idea what it would take to get a call parking feature that satisfies these requirements into freeswitch? Thanks for your time, sorry for the long email, hopefully it explains well enough what I'm trying to accomplish. -Tom

5 9

Lua playAndGetDigits
by Michael De Lorenzo 17 Mar '10

17 Mar '10

I'm having some trouble getting consistent results with playAndGetDigits, sometimes the digits are record and other times they're not. It seemed to have something to do with how quickly the digits were pressed, but now that no longer seems to be the case. I'm trying to have users press a confirmation code, in this case "1111" although I'd like to make it accept "1" or "11" or "111" or "1111" if possible. Can anyone point me in the right direction? session = freeswitch.Session("{ignore_early_media=true}sofia/gateway/" .. provider .. "/1" .. number_to_call); while(session:ready()) do session:answer(); session:setInputCallback("onInput","true"); --playbackMessage(check_message); digits = session:playAndGetDigits(4, 4, 1, 5000, "", get_digits_message, "", "[1]"); freeswitch.consoleLog("info", digits .. "\n"); freeswitch.consoleLog("info", "Did this check out? " .. (check_message == true and "yes" or "no" .. "\n")); -- we got what we're looking for, so we can end this if(check_success) then freeswitch.consoleLog("info", "A positive response was received from this check.\n"); break; else -- not a positive response -- if we reached our max attempts, we're finished move on if(attempts == max_attempts) then freeswitch.consoleLog("info", "We've reached our maximum attempts for this number.\n"); break; end end end

4 12

Re: [Freeswitch-users] how to access session variables after hangup???
by rentmycoder rentmycoder 17 Mar '10

17 Mar '10

> Pass them as arguments to your luarun hangup hook. And what about variables are not available before the call setup (like uuid), only during the call progress or after the hangup??? I would need to access the originate_disposition variable... >> 'luarun hangup.lua ${originate_disposition}' >> 2010-03-14 18:25:08.122511 [CRIT] switch_channel.c:759 Invalid data (${api_hangup_hook} contains a variable) any other hint??? > ---------- Forwarded message ---------- > From: Brian West <brian(a)freeswitch.org> > To: freeswitch-users(a)lists.freeswitch.org > Date: Sun, 14 Mar 2010 13:00:10 -0500 > Subject: Re: [Freeswitch-users] how to access session variables after hangup??? > Pass them as arguments to your luarun hangup hook. > > /b > > On Mar 14, 2010, at 12:40 PM, rentmycoder rentmycoder wrote: > >> Hi guys, >> Freeswitch rocks! >> except the documentation:( >> I'm not able to figure out how to access session variables after hangup... >> I try to dial out on a gateway and detect it the called party hangs up >> before bridge... >> >> bgapi originate >> {ignore_early_media=true,continue_on_fail=true,bypass_media=false,hangup_after_bridge=true,originate_timeout=10,api_hangup_hook='luarun >> hangup.lua ${uuid}'}sofia/gateway/phoneno &park() >> >> I cannot access the variables using: >> 1. script arguments: >> 'luarun hangup.lua ${uuid}'} >> 2010-03-14 18:25:08.122511 [CRIT] switch_channel.c:759 Invalid data >> (${api_hangup_hook} contains a variable) >> >> 2. session:getVariable("originate_disposition") >> 2010-03-14 18:31:37.042756 [ERR] mod_lua.cpp:182 hangup.lua:2: attempt >> to call global 'getVar' (a nil value) >> >> 3. params:getHeader("variable_sip_req_uri") >> 2010-03-14 18:36:19.973939 [ERR] mod_lua.cpp:182 hangup.lua:3: attempt >> to index global 'params' (a nil value) >> >> I have asked this before on the dev list too, but Anthony didn't gave >> me a clean answer here: >> http://lists.freeswitch.org/pipermail/freeswitch-dev/2009-July/002491.html >> >> "In the script you can access the data from the env event which is >> automatically created for you and it >> contains all the channel variables from that departed channel." >> >> Ok, but how????? >> >> Please help... >>

3 2