March 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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/

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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

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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

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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... >>

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Skype for SIP - No SILK support, yet...
by Chris Fowler 17 Mar '10

17 Mar '10

I was thrilled when mod_silk was released... but my joy was short lived when I got this reply from Skype about SILK support on their "Skype for SIP beta". "We have double checked this with Skype for SIP developers. They said that at the moment SILK codecs are not yet supported, even though we have information about this on the web site. Most probably SILK codecs will be supported when Skype for SIP goes out from beta stage." Just wanted to share to avoid frustration for other FreeSWITCHers. Chris.

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A few questions after upgrade to trunk
by Seven Du 17 Mar '10

17 Mar '10

ALL, I just upgrade FS to 16992. 1) timer warning: How's possible the 10000 microseconds? audio sounds ok. [WARNING] switch_time.c:206 Timer resolution of 10000 microseconds detected! Do you have your kernel timer set to higher than 1 kHz? You may experience audio problems. uname -a Linux xxx.idapted.com 2.6.22-14-server #1 SMP Tue Feb 12 08:27:05 UTC 2008 i686 GNU/Linux I used a script to check the interupts :~$ while :; do cat /proc/interrupts |grep LOC; sleep 1; done LOC: 14892854 14892843 14892841 14892830 LOC: 14892955 14892944 14892942 14892931 LOC: 14893056 14893045 14893043 14893032 LOC: 14893157 14893146 14893144 14893133 LOC: 14893258 14893247 14893245 14893234 LOC: 14893359 14893348 14893346 14893335 LOC: 14893460 14893449 14893447 14893436 2) run dingaling with client mode. dingaling doesn't pick up answer. I originate a call from FS, the console blocks, gtalk client ring and it looks ok when I hit answer, but no sound, FS console still blocks. a few seconds later gtalk client shows: Sorry! The voice chat with language_lesson failed because of a network problem at 5:50 PM. Please try again. Click here to report this to Google. here is a log with bgapi: http://pastebin.freeswitch.org/12434 rev 14696 works on the same server same conf. another server running 16958M works at the same time. I have no clue to check, is it related to the timer as in 1)? can some help me to take a look? let me know if you need "dl_debug on" logs. Thanks.

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RFC3578
by Timur Irmatov 17 Mar '10

17 Mar '10

Hi, Is there support for RFC3578 in FreeSWITCH? If not, is there any plans to implement it? As far as I understand from quick browsing through this RFC, it is about mapping of ISUP overlap signalling to SIP protocol. In short, remote device (softswitch for example) sends short (incomplete) number in INVITE message to freeswitch, which responds with '484 address incomplete', then softswitch send another INVITE with more digits. Freeswitch then can send another 484, and after some time (enough digits collected), proceed with usual call establishment. -- Timur Irmatov, xmpp:irmatov@jabber.ru

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MWI Question
by Aloysius Lloyd 17 Mar '10

17 Mar '10

Hi All, I delete all the voice mails in the voice mail folder for a particular extension. But the phone[AASTRA 9133i] MWI keep on blinking and and display showing the no of messages. How can I fix this issue. Thanks Lloyd

2 1

mod_cdr_csv
by Troy Anderson 16 Mar '10

16 Mar '10

What triggers mod_cdr_csv to log a call? I'm not sure if what I'm experiencing is due to my mistake or otherwise, but here's the situation. Incoming call (via PSTN or a VoIP provider) -> Extension 105 and answered 105 transfers to 115, 115 answers, 105 completes transfer The call progresses and is 115 finally hangs up. The Master.csv only gets one line added to it - the 105 to 115 call. Nothing about the incoming caller. If I knew what triggered the log entry, I could try to track the issue down in my dial plan. Thanks! -Troy

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ivr call and secret keycode
by Madovsky 16 Mar '10

16 Mar '10

Hi all, is anyone already worked on an ivr that make a call so the callee needs to enter a keycode to validate a form filled before on a website for example ? Thanks Franck

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