June 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

how to configuring nibblebill module to only billing every amount of heartbeat?
by Samira Mh 14 Jun '12

14 Jun '12

hi, in "nibblebill" module of freeswitch, billing will only occur at the every 60 second by set like:( <action application="nibblebill" data="heartbeat 60"/>) but while making call and call is answered the minimmun amount of cash is reduced and also billing is done every 60 seconds, (in addition of billing every 60 seconds the minimmum amount of cash also reduced at the first time the call is answered) is it possible to set somethings so that don't reduce (billing) the minimum amount of cash at the same time call is answered and only billing every 60 second? thanks so much

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FSClient UA error
by Jack 14 Jun '12

14 Jun '12

I am getting this error in my log when I try to run FSClient on windows xp: 2012-06-06 09:59:36.500000 [NOTICE] sofia.c:4892 Started Profile softphone [sofia_reg_softphone] 2012-06-06 09:59:36.500000 [DEBUG] mod_sofia.c:5471 Waiting for profiles to start 2012-06-06 09:59:36.546875 [ERR] sofia.c:2099 Error Creating SIP UA for profile: softphone I also get a dialog box that says: Warning the master sofia profile was not able to load and the phone will most likely _not_work,..... I have checked the port and opened the firewall 5060 and tried reloading the sofia profile by editing the settings and saving and I still get the same. Thanks, Jack

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lcr sql variable
by FERNANDO VILLARROEL 14 Jun '12

14 Jun '12

Hi all. I need setup some variable like foo to my cdr from lcr : <param name="export_fields" value="tarifa"/> <param name="custom_sql" value="select a.tarifa as tarifa ...... Returning Dialstring [lcr_carrier=carrier1,lcr_rate=0.15000,nibble_account=1004,nibble_rate=0.15000,lowbal_amt=0,nobal_amt=0,tarifa=1]user/1006 My dialplan: <extension name="Outbound to PSTN 11 Digits"> <condition field="destination_number" expression="^(10[01][0-9])$"> <action application="set" data="hangup_after_bridge=false"/> <action application="set" data="continue_on_fail=true"/> <action application="lcr" data="$1 pg_prefix2"/> <action application="set" data="tarifa=${tarifa}"/> <action application="bridge" data="${lcr_auto_route}"/> </condition> </extension> But the log show : mod_dptools.c:1294 sofia/internal/1004(a)192.168.1.108 SET [tarifa]=[UNDEF] So how i can get the value of the variable "tarifa" and set for my CDR? Regards.

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NAT Router Recommendation For Large no of IP Phones Behind the Router
by Lloyd Aloysius 13 Jun '12

13 Jun '12

Hi All: Currently I use pfSense for NAT implementations. But I found there are few bugs in the firmware cause Presence not working reliably. What are best routers out there with good support for SIP. I am looking for a NAT router that support 20-30 phones . Any good recommendations? Thanks Lloyd

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Audio Connection Delay with SCA
by Sean Devoy 13 Jun '12

13 Jun '12

Hi, I finally got Shared Call Appearance working to 4 NAT connected phones!!!!! Now the users say when they answer that line, they must wait a second or so to start talking (no big deal), but they all claim it takes nearly 5 seconds to be able to hear the caller. They answer the standard "Thank You for calling XYZ, how can I help you today" which the caller DOES HEAR, but they cannot hear his response for the first 5 seconds and ALWAYS have to ask them to repeat it. In the logs, I see what I believe is them picking up extension 299: 2012-06-08 11:44:15.115249 [NOTICE] switch_channel.c:926 New Channel sofia/external/sip:299@69.251.170.6:5063 . And then the RPT finally connecting: 2012-06-08 11:44:20.665094 [DEBUG] sofia_glue.c:3248 AUDIO RTP [sofia/external/sip:299@69.251.170.6:5064] 10.10.40.185 port 23500 -> 69.251.170.6 port 16498 codec: 0 ms: 20 Note the 5.5 second delay! Granted there are 100+ lines between those if you want to see them, I will post them. After that audio is fine. Any thoughts would be appreciated. Sean

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Polycom caller ID problem on 4.0.2
by DJB International 13 Jun '12

13 Jun '12

I've recently updated my Polycom 650 phone to use 4.0.2. I am not sure what broke the caller id for received calls, it kept showing as Unknown Party; however, the weird thing is that the Missed Calls show the correct caller id. Anyone experienced this issue. Please advise. FreeSWITCH Version 1.2.0-rc2 Thank you.

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problem on playinback " .wav "files on freeswitch using asterisk with Codec G729 ?
by Samira Mh 13 Jun '12

13 Jun '12

hi guys, as you know on freeswitch the codec G729 only use fro pass through, so we couldn't play .wav file using codec G729, so i have installed asterisk to everytime i want to play file with that codec play them on asterisk using gateway on freeswitch , but in asterisk when i issue the command (exten => 999,1,playback(hello-world)) the file play successfully using freeswitch that is connect to asterisk but i don't want to answer channel at this situation so issue the command (exten => 999,1,playback(hello-world,noanswer) or (exten => 999,1,background(hello-world,n) -- n used to don't answer channel )) i don't hear the sound of .wav file on the debug of sip on asterisk the file is played and command was executed but i coudn't hear sound notice:because of using nibble_account on freeswitch i don't want to answer channel to reduce the cash plz help thanks

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Re: [Freeswitch-users] how to use codec g729 on freeswitch ?
by Peter Olsson 13 Jun '12

13 Jun '12

The passthrough codec for G.729 is built by default, just make sure to load the module mod_g729. If you need transcoding, or use IVR's, voicemail etc, you must purchase commercial licenses for G729. /Peter Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Samira Mh Skickat: den 11 juni 2012 10:16 Till: Free SWITCH Users Help Ämne: [Freeswitch-users] how to use codec g729 on freeswitch ? hi guys, i am going to make call using codec g729, because the carrier support only codec g729 is it possible to get free that? in freeswitch implement only g729 passthrough . but how can i get it ? !DSPAM:4fd5a7d032762000267653!

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Moving up to better PC - Multi-tenant died
by Sean Devoy 13 Jun '12

13 Jun '12

HI, I have a nice working multi-tenant configuration working on my development box - P4 3.0Ghz 4GB RAM Raid 1 Centos 5.7 Freeswitch (make current). I built a nice clean Centos 5.7 install on a Dell Power Edge 2850 Dual Xeon 3.0 Ghs 4GB RAM and current Freeswitch. I saved the entire "/usr/local/freeswitch/conf" tree from new server. I copied the entire "/usr/local/freeswitch/conf" tree from devel box to new server. I shutdown development box. I set new server to same IP as old Devel boix. Rebooted. I have a couple of issues: 1. Freeswitch does not startup at boot. I tried to follow the wiki. I have a freeswitch file in /;etc/init.d but no startup! 2. Only my primary domain (tenant) users can login. All of my secondary domain phones have proxy=nnn(a)fs_mbri.bizfocused.com but the errors on FS_CLI say it can't find nnn(a)fs_bfis.bizfocused.com (MY PRIMARY DOMAIN). It is though freeswitch thinks these registrations are coming to the primary domain even though they are going to a secondry with the same IP address. So, how do I get FS to start on boot up in CENTOS 5.7? I swear I tried what's in the wiki. Or maybe I should say how do I tell wait is failing? What am I missing in the multi-tenant domain definition (that is not in the /conf folder)? Thanks in advance. Sean

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Domain name lost in SIP registration (w/ SIP TRACE)
by Sean Devoy 13 Jun '12

13 Jun '12

HI All, In the pastebin: <http://pastebin.com/KyGXMTP0> http://pastebin.com/KyGXMTP0 is a sip trace. The phone is on the lan and uses proxy: fs_test.bizfocused.com which is 10.10.40.189. In the SIP messages you can see the domain name "fs_test.bizfocused.com". I have a directory that includes <domain name="fs_test.bizfocused.com"> and a user for this subscription. But the SIP response is "SIP/2.0 401 Unauthorized" and the message displayed in FS_CLI is: [WARNING] sofia_reg.c:2376 Can't find user [202(a)10.10.40.189] You must define a domain called '10.10.40.189' in your directory and add a user with the id="202" attribute and you must configure your device to use the proper domain in its authentication credentials. Why isn't it using the credentials in the domain "fs_test.bizfocused.com"? Thanks, Sean

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