October 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Emulate and script emulation of users calling in/receiving calls, transferring calls &etc
by Alec Taylor 15 Oct '11

15 Oct '11

Good evening, If asterisk or freeswitch would be taught in a classroom environment, is there someway to emulate and script emulation of users calling in/receiving calls, transferring calls &etc? The plan is to have each student setup there own Asterisk or FreeSwitch box, and measure handling efficiency, and communicate between the servers (by transferring calls from server to server). Thanks for all suggestions, Alec Taylor

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Change sound someone hears when they ring me
by Ryan How 15 Oct '11

15 Oct '11

Hi, I've set a freeswitch box with bluebox (with a bit of hassle! but it got there). I can't find where to start looking regarding changing the sound an external person hears when they call me. usually from the standard landline when dialing a number I get a ring-ring ring-ring, but when someone dials my number (which is a voip DID) it has a long riiiiiiiiinnnnngggggg . . . . . riiiiiiiiinngggggg. Is there any way I can change that? My asterisk box and other voip phones haven't done it, so I figure it must be able to be changed. Thanks, Ryan

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Custom rinback tone
by Cesar Bermudez 14 Oct '11

14 Oct '11

hi fs users and devs. mi ringback tone its 450/1200, 0/4900 ( 1.2 on 4.9 off ) how can modify this: <action application="set" data="ringback=%(2000, 4000, 440.0, 480.0)"/> to have the same ringing tone. Best regards.

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Freeswitch equilavent to Asterisk BackgroundDetect
by Jon Schøpzinsky 14 Oct '11

14 Oct '11

Hello List. Ive been trying to find a function equal to what BackgroundDetect does in Asterisk. Anybody got an idea on how I could implement the functionality in Freeswitch? I use it in IVR services, to detect when a caller interrupts the currently playing prompt. Med venlig hilsen/Kind Regards Jon Leren Schøpzinsky

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Run a js program as soon as someone registers with FS
by Yungwei Chen 14 Oct '11

14 Oct '11

Hi, I am wondering if it's possible to run a js program as soon as someone registers with a freeswitch. Thanks.

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Error while using origiante command -- "Cannot Blind Transfer 1 Legged Calls"
by Papineni, Suneel 14 Oct '11

14 Oct '11

Hi, I have downloaded latest version of Freeswitch (from GIT) and compiled. It went through and created Release version (on a Windows system). Using X-lite, I am able to register users 1001 and 1003 and able to make a call between two users, but when I try to originate a call to one of extensions from freeswitch console, I am getting an error saying "Cannot Blind Transfer 1 Legged Calls". Following command is used for originating call. "originate sofia/internal/1003(a)172.22.200.300<mailto:sofia/internal/1003@172.22.200.300> 9999" Could someone help me why it is giving this error and how to resolve this. Thanks & Regards Suneel Here is sofia status and logs freeswitch@UK-SPapineni-7> sofia status Name Type Data State ================================================================================================= 172.22.200.300 alias internal ALIASED internal profile sip:mod_sofia@172.22.200.300:5060 RUNNING (0) internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) external profile sip:mod_sofia@172.22.200.300:5080 RUNNING (0) external::example.com gateway sip:joeuser@example.com NOREG ================================================================================================= 3 profiles 1 alias Console Log..... freeswitch@UK-SPapineni-7> originate sofia/internal/1003(a)172.22.200.300 9999 2011-10-14 16:44:24.894231 [NOTICE] switch_channel.c:911 New Channel sofia/internal/1003(a)172.22.200.300 [739a9189-d72b-4092-9c97-04af9d577e40] 2011-10-14 16:44:24.914233 [NOTICE] switch_channel.c:911 New Channel sofia/internal/0000000000(a)172.22.200.300 [56968324-cacc-41e8-834e-224716e666cf] 2011-10-14 16:44:24.914233 [INFO] mod_dialplan_xml.c:336 Processing <0000000000>->1003 in context public 2011-10-14 16:44:24.914233 [ERR] sofia.c:6591 Cannot Blind Transfer 1 Legged calls 2011-10-14 16:44:25.014243 [NOTICE] switch_core_state_machine.c:226 sofia/internal/0000000000(a)172.22.200.300 has executed the last dialplan instruction, hanging up. -ERR NO_USER_RESPONSE 2011-10-14 16:44:25.014243 [NOTICE] switch_core_state_machine.c:228 Hangup sofia/internal/0000000000(a)172.22.200.300 [CS_EXECUTE] [NORMAL_CLEARING] 2011-10-14 16:44:25.014243 [NOTICE] sofia.c:5926 Hangup sofia/internal/1003(a)172.22.200.300 [CS_CONSUME_MEDIA] [NO_USER_RESPONSE] 2011-10-14 16:44:25.014243 [NOTICE] switch_core_session.c:1369 Session 27 (sofia/internal/0000000000(a)172.22.200.300) Ended 2011-10-14 16:44:25.014243 [NOTICE] switch_core_session.c:1371 Close Channel sofia/internal/0000000000(a)172.22.200.300 [CS_DESTROY] 2011-10-14 16:44:25.034245 [NOTICE] switch_core_session.c:1369 Session 26 (sofia/internal/1003(a)172.22.200.300) Ended 2011-10-14 16:44:25.034245 [NOTICE] switch_core_session.c:1371 Close Channel sofia/internal/1003(a)172.22.200.300 [CS_DESTROY]

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Re: [Freeswitch-users] There is no voice in Two party call
by chrisbware＠interfree.it 14 Oct '11

14 Oct '11

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Re: [Freeswitch-users] conference with pin code and esl
by Madovsky 14 Oct '11

14 Oct '11

I just found http://jira.freeswitch.org/browse/FS-1565 but closed for incomplete I wonder why before the pin code was working also for the moderator but not now. Thanks ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Friday, October 14, 2011 11:48 AM Subject: Re: conference with pin code and esl mmhmm ok I didn't see this line in log 2011-10-14 11:28:27.714649 [DEBUG] mod_conference.c:6102 Moderator PIN found! but I didn't set any pin code in conference.conf.xml ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Friday, October 14, 2011 11:34 AM Subject: conference with pin code and esl I'm running a conference in esl so in the log EXECUTE sofia/internal/123456789(a)kiki.hehe socket(127.0.0.1:9090 async full) 2011-10-14 11:28:15.534653 [DEBUG] switch_core_session.c:1009 Send signal sofia/internal/123456789(a)kiki.hehe [BREAK] 2011-10-14 11:28:15.534653 [DEBUG] switch_ivr.c:577 sofia/internal/123456789(a)kiki.hehe Command Execute conference(conf_123456789@default+2222+flags{moderator,endconf}) EXECUTE sofia/internal/123456789(a)kiki.hehe conference(conf_123456789@default+2222+flags{moderator,endconf}) but if the caller don't type any pin code ther conference is created. this behaviour exist since I updated to this git FreeSWITCH Version 1.0.head (git-fa3fa30 2011-10-07 11-20-36 -0500) Must I change anything ? Thanks

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conference with pin code and esl
by Madovsky 14 Oct '11

14 Oct '11

I'm running a conference in esl so in the log EXECUTE sofia/internal/123456789(a)kiki.hehe socket(127.0.0.1:9090 async full) 2011-10-14 11:28:15.534653 [DEBUG] switch_core_session.c:1009 Send signal sofia/internal/123456789(a)kiki.hehe [BREAK] 2011-10-14 11:28:15.534653 [DEBUG] switch_ivr.c:577 sofia/internal/123456789(a)kiki.hehe Command Execute conference(conf_123456789@default+2222+flags{moderator,endconf}) EXECUTE sofia/internal/123456789(a)kiki.hehe conference(conf_123456789@default+2222+flags{moderator,endconf}) but if the caller don't type any pin code ther conference is created. this behaviour exist since I updated to this git FreeSWITCH Version 1.0.head (git-fa3fa30 2011-10-07 11-20-36 -0500) Must I change anything ? Thanks

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esl functions in c
by adam harris 14 Oct '11

14 Oct '11

Where can I find out what esl functions are availible in C

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