March 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] voicemail options
by Madovsky 16 Mar '10

16 Mar '10

I noticed that sendmail makes a segfault when FS is trying to send email and sendmail has some emails locked in folder queue.... Regards Franck

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Lua - playAndGetDigits
by Michael De Lorenzo 16 Mar '10

16 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] modules update
by Milena 16 Mar '10

16 Mar '10

Hello, if you edit modules.conf (in the source folder) and uncomment every module that you want to build, you will not need to make install every module, make current is enough. :)

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Call parking strategy/best practice in freeswitch?
by Tom Christensen 16 Mar '10

16 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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Load balancing solution using pure FreeSWITCH
by Vitalii Colosov 16 Mar '10

16 Mar '10

Hi dear FS community, After a brief analysis of 2 solutions using OpenSIPS and Kamailio ( http://www.opensips.org/html/docs/modules/devel/load_balancer.html#id226109) and ( http://wiki.freeswitch.org/wiki/SBC_Setup) I decided to implement my own solution using ONLY FreeSWITCH and ONLY 1 table of any database server (we use MySQL). It works on few of our servers right now without any issues. Feel free to reuse / improve it or provide your comments if you see something wrong with it. We have one routing server (RS) and several Media Servers (MS). FreeSWITCH is installed on each server in default configuration. --> MS_1 RS --> MS_2 --> MS_N (Failover can be implemented by adding one more routing server, but this is out of scope of current description). Routing server routes calls to media servers based on logic that you can easily define. (Generally speaking, Routing server can route part of the calls onto itself, but to simplify let's say that it will route only to Media Servers). Routing server is used to register subscribers (it will have directory files ./freeswitch/conf/directory/default/10001whatever.xml-99999whatever.xml). Media servers do NOT contain files with users (security issue? probably should be managed on dialplan stage of media servers by comparing ip addresses of requests). The whole routing can be presented by the following chain: Call arrives into Default context-->RS default dialplan-->Lua script-->PHP script to get destination MEDIA SERVER IP-->back to Lua script-->Bridge "sofia/external/" to determined MS IP address-->[Bridge to fail over media server, optionally]-->End Let's say we need to route all international calls based on less loaded server (other logic can be easily implemented in PHP script as well). *1. RS Dialplan: default.xml* <include> <extension name="route"> <condition field="destination_number" expression="^(00)(\d+)"> <action application="lua" data="route.lua $2"/> </condition> </extension> </include> *2. RS Lua script: route.lua* called_number = argv[1]; //without "00" api = freeswitch.API(); server_ip_address = api:executeString("curl http://localhost/get_server_ip.php) forwarding_session_string = "sofia/external/"..called_number.."@"..server_ip_address; session:setVariable("bypass_media", "true"); //described below!!! session:execute("bridge",forwarding_session_string); *3. PHP script: get_server_ip.php* <<< ANY PHP CODE THAT WILL QUERY COMMON DATABASE AND WILL DETERMINE LESS LOADED AT THE MOMENT SERVER>>> die($less_loaded_server_ip); All Media Servers should provide information about their current load into one common database (dedicated server or routing server can be used for database). If you use scripts you can update this table on starting a call and after call finished. Or you can create a job that will query FreeSWITCH and will update the database every, say, 10 seconds. Basically that's all (really!) I beleive it can be used for most general purpose scenarios without need of third-party tools. P.S. The only problem we had from the beginning was that RTP media traffic was passing thru both servers: RS and Media Server. But after we added the following command to the routing script, RTP media traffic started to pass only thru Media server, making RS almost free. session:setVariable("bypass_media", "true"); Best regards, vIT

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modules update
by Madovsky 16 Mar '10

16 Mar '10

Hi there, do I need to "make install" separatly modules (like distributor) everytime I upgrade from trunk (make current) ? Thanks Franck

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freeswitch to cisco gateway
by srinivasula reddy 16 Mar '10

16 Mar '10

Hi, i am using freeswitch1.0.2, farwarding call from freeswitch to cisco like just farwading the call. <action application="bridge" data= sofia/internal/${destination_number}@ciscogatewayip<sofia/internal/$%7Bdestination_number%7D@ciscogatewayip> / <data=sofia/internal/$%7Bdestination_number%7D@ciscogatewayip/>> i do not using any gateway here, just farwarding call to router, is this correct way of doing? any idea -- Srinivasula Reddy K

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answering call in irly_media mode
by Meftah Tayeb 16 Mar '10

16 Mar '10

hi all i want to setup a custom ringback tone for inbound (not outbound calls) is in irly_media mode but i still heare the normal US ringback. where is my dialplan extension: http://pastebin.freeswitch.org/12452 thanks

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Re: [Freeswitch-users] Set/Export channel variable [SOLVED]
by Michael Collins 16 Mar '10

16 Mar '10

On Mon, Mar 15, 2010 at 11:21 PM, Ahmed Ajmal <ahmed.ajmal(a)breezecom.ae>wrote: > So it seems that the FS console by default only displays the ‘a’ leg vars > which is why I couldn’t see it (and it makes sense too since gateway is on > the b leg side). I have the ESL script now display both the inbound/outbound > after which I can see the sip_gateway_name var. > Good catch! -MC

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Speex transcoding question
by Bruce Hopkins 16 Mar '10

16 Mar '10

Hi, I am having trouble getting my configuration right so that I can have a call transcoded to Speex wideband from another codec (alaw or g.722). If both phones use Speex wideband with no transcoding required by FS, the call succeeds though. My codecs are listed in vars.xml as follows: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=speex@32000 @20i,speex@16000h@20i,G722,G7221@32000h,G7221@16000h,PCMA,PCMU,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=speex@32000 @20i,speex@16000h@20i,G722,G7221@32000h,G7221@16000h,PCMA,PCMU,GSM"/> However if I make a call from a phone A using, say, g.722 to a phone using Speex wideband, the SDP in the invite from phone A to phone B does not include Speex wideband. In fact the SDP includes speex/8000 even though speex/8000 is neither enabled in vars.xml, not in either of the phones. Here is the codec listing in the SDP of the INVITE from Freeswitch to phone B (from Wireshark) : Media Attribute (a): rtpmap:9 G722/8000 Media Attribute (a): rtpmap:99 SPEEX/8000 Media Attribute (a): rtpmap:115 G7221/32000 Media Attribute (a): fmtp:115 bitrate=48000 Media Attribute (a): rtpmap:107 G7221/16000 Media Attribute (a): fmtp:107 bitrate=32000 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:3 GSM/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-16 Media Attribute (a): rtpmap:13 CN/8000 Media Attribute (a): ptime:20 Could anyone tell me what I am doing wrong please? Many thanks Bruce

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