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Conference no sound
by sukitha jayasinghe 17 Dec '14

17 Dec '14

Hi All, I have configured freeswitch server with conference feature enabled, When I call the conference number, i can hear hold music. when second call dial the conference number hold music stops, but no voice between legs. what would be the reason for this. Best Regards, Sukitha

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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 102, Issue 100
by sukitha jayasinghe 17 Dec '14

17 Dec '14

Other calls working fine, in conference also both sides can hear conference instruction messages. On 17 Dec 2014 19:15, <freeswitch-users-request(a)lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: Conference no sound (?talo Rossi) > 2. Re: How to play a stream other than mp3 with mod_shout. > (Aqs Younas) > 3. Re: Issue with Freeswitch Behind nat (Frederick Pruneau) > > > ---------- Forwarded message ---------- > From: "Ítalo Rossi" <italorossib(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Wed, 17 Dec 2014 09:41:08 -0300 > Subject: Re: [Freeswitch-users] Conference no sound > Is there audio when you call the other user directly, without the > conference? > > On Wed, Dec 17, 2014 at 8:30 AM, sukitha jayasinghe <sukithaj(a)gmail.com> > wrote: >> >> Hi All, >> >> I have configured freeswitch server with conference feature enabled, When >> I call the conference number, i can hear hold music. when second call dial >> the conference number hold music stops, but no voice between legs. what >> would be the reason for this. >> >> Best Regards, >> Sukitha >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> 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 >> > > > -- > Ítalo Rossi > > > ---------- Forwarded message ---------- > From: Aqs Younas <aqsyounas(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Wed, 17 Dec 2014 18:34:24 +0500 > Subject: Re: [Freeswitch-users] How to play a stream other than mp3 with > mod_shout. > Hi, Moishe Grunstein > > I am able to successfully configure mod_vlc, but when i play i stream with > vlc i hear nothing. Like dead call. > > Here is my default.xml > > <include> > <context name="default"> > <extension name="Testing"> > <condition field="destination_number" expression="^(\+1|1)?\d+$"> > <action application="answer"/> > <action application="log" data="Call in progress"/> > <action application="playback" data="vlc:// > http://s9.voscast.com:7584"/> > </condition> > </extension> > </context> > </include> > > When call is answered and i hangup the call from my softphone, it does not > actually ends on freeswitch side. > Also when i type > > freeswitch@internal> hupall > +OK hangup all channels with cause MANAGER_REQUEST > > But > freeswitch@internal> show calls count > 2 total. > > Call doesn't terminate. When i play the stream with mod_shout everything > works fine. > > My OS is Debian 7.6. > > Can you please provide my with example how to play stream with vlc. Or > anything that i am doing wrong. > > Best Regards. > > > On 15 December 2014 at 17:03, Aqs Younas <aqsyounas(a)gmail.com> wrote: >> >> Thanks for your reply. I will try that too. >> Many thanks for your valuable replies. >> >> On 15 December 2014 at 00:37, Danny Gershman <danny.gershman(a)gmail.com> >> wrote: >> >>> Also mod_rtmp lets you play from an FMS server. >>> >>> >>> On Friday, December 12, 2014, Aqs Younas <aqsyounas(a)gmail.com> wrote: >>> >>>> Hi, All >>>> >>>> How can i play a live stream other than mp3 with mod_shout or any >>>> module.? Is there any way to buffer the stream before playing it with >>>> mod_shout. >>>> >>>> >>>> Currently i have a list of streams and when i play them with mod_shout >>>> some work fine but others give (time out) error. >>>> >>>> How can i play mostly stream in freeswitch? >>>> >>>> Thanks >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting(a)freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >> > > ---------- Forwarded message ---------- > From: Frederick Pruneau <frederick(a)targointernet.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Wed, 17 Dec 2014 08:44:10 -0500 > Subject: Re: [Freeswitch-users] Issue with Freeswitch Behind nat > Sorry for this noob question but how can I see sip traffic? Is there a > specific command to show this? Is it what we find in freeswitch.log? If so, > I attached my log file in my first post. > > Thanks for you help > > 2014-12-16 15:44 GMT-05:00 Brian West <brian(a)freeswitch.org>: >> >> have you looked at the signalling? What does the sip traffic show? >> Please pastebin that. >> >> On Tue, Dec 16, 2014 at 2:37 PM, Frederick Pruneau < >> frederick(a)targointernet.com> wrote: >>> >>> Same problem... >>> >>> 2014-12-16 13:55 GMT-05:00 Brian West <brian(a)freeswitch.org>: >>> >>>> Guessing you don't have UPNP or NAT-PMP on your network, there for that >>>> won't work, >>>> >>>> ext-sip-ip=autonat:x.x.x.x >>>> ext-rtp-ip=autonat:x.x.x.x >>>> >>>> Set local-network-ac to rfc1918.auto >>>> >>>> On Tue, Dec 16, 2014 at 12:15 PM, Support Technique < >>>> support(a)targointernet.com> wrote: >>>>> >>>>> <param name="ext-rtp-ip" value="auto-nat"/> >>>>> <param name="ext-sip-ip" value="auto-nat"/> >>>>> <param name="local-network-acl" value="localnet.auto"/> >>>>> >>>>> 2014-12-16 12:25 GMT-05:00 Brian West <brian(a)freeswitch.org>: >>>>> >>>>>> On the system behind nat what do you have ext-rtp-ip, ext-sip-ip and >>>>>> local-network-acl set to? >>>>>> >>>>>> On Tue, Dec 16, 2014 at 10:41 AM, Frederick Pruneau < >>>>>> frederick(a)targointernet.com> wrote: >>>>>> >>>>>>> Hi guys, >>>>>>> >>>>>>> We have an issue with one freeswitch server behind nat. We have a >>>>>>> setup like this: >>>>>>> >>>>>>> -One master Freeswitch server >>>>>>> >>>>>>> -One freeswitch server connected to the master (Public IP) - Server A >>>>>>> >>>>>>> -One freeswitch server connected to the master (behind nat) - Server >>>>>>> B >>>>>>> >>>>>>> If server A call server B, nothing happens. There is no sound. After >>>>>>> 30 sec, it times out. We have done a tcpdump. From server A to master >>>>>>> packets are ok. From Master to server B, we have seen that there is no >>>>>>> source and no destination ports for sip invite. >>>>>>> >>>>>>> If we use our cellphone and we call server B, there is no problem. >>>>>>> >>>>>>> I have attached the failed call pcap file and freeswitch's log file >>>>>>> so you can take a look at them. >>>>>>> >>>>>>> Master = Freeswitch v1.4.13 >>>>>>> Server A = Freeswitch v.1.4.13 >>>>>>> Server B = Freeswitch v.1.4.14 (Updated to latest release since we >>>>>>> have issues with this server) >>>>>>> >>>>>>> Thanks in advance. >>>>>>> >>>>>>> PS: The failed call is from 514-448-0773. >>>>>>> >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting(a)freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://confluence.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> 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 >>>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> >>>>>> *Brian West* >>>>>> brian(a)freeswitch.org >>>>>> >>>>>> >>>>>> *Twitter: @FreeSWITCH , @briankwest* >>>>>> http://www.freeswitchbook.com >>>>>> http://www.freeswitchcookbook.com >>>>>> >>>>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>>>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting(a)freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> 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 >>>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting(a)freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> 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 >>>>> >>>> >>>> >>>> -- >>>> >>>> *Brian West* >>>> brian(a)freeswitch.org >>>> >>>> >>>> *Twitter: @FreeSWITCH , @briankwest* >>>> http://www.freeswitchbook.com >>>> http://www.freeswitchcookbook.com >>>> >>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting(a)freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting(a)freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >> >> >> -- >> >> *Brian West* >> brian(a)freeswitch.org >> >> >> *Twitter: @FreeSWITCH , @briankwest* >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> 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 >> > > > > _______________________________________________ > 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 > >

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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 102, Issue 69
by Ahmed Habiba 17 Dec '14

17 Dec '14

Thanks you really Michael. My point is as below: 1-when FS get Invite from A number the TO is as below: To: <sip:2020@71.233.135.12> 2-and when I run answer application, FS send invite to A number with TO as below: To: <sip:2020@71.233.135.12> 3-I want when I run answer application to edit the TO header and to add some prefix like 505 as below: To: <sip:5052020@71.233.135.12> is this possible, by editing sip header, and if so how can I do so? Your kind usual support will be appreciated. > > From: Michael Jerris <mike(a)jerris.com <mailto:mike@jerris.com>> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>> > Date: December 13, 2014 at 12:37:42 AM GMT+3 > Reply-To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>> > Subject: Re: [Freeswitch-users] SIP Header Modification / Manipulation > > > if you are looking to update the display of the number that is being called on the caller phone, you want to be looking at display udpates, and callee_id_name and callee_id_number params. > >> On Dec 12, 2014, at 2:39 PM, Ahmed Habiba <ahabiba(a)gmail.com <mailto:ahabiba@gmail.com>> wrote: >> >> Dears, >> >> Kindly help me on how can I modify the SIP header in the re-invite in the below scenario: >> >> >> [A Number] <———>[FreeSWitch]<———>[B Number] >> >> >> 1-A Number Call B Number (Sip Invite sent from A number) >> 2-B number is not registered. >> 3-Call logic let FreeSwitch answer the call( Freeswitch send re-invite to A number) >> >> in the part number 3 I like to do modification on the SIP headers, by for example changing the From or To. >> >> >> Your help will be appreciated. >> > >

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CallerID for E911
by Joel White 16 Dec '14

16 Dec '14

Hi everyone, I was wondering what the maximum length a caller id number can be. I need to implement appending a location ID to each extension not assigned a DID. This is for purposes of the E911 system. Does anyone have experience with this? Any help would be greatly appreciated

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How to get the action executed using bind_digit_action ?
by Keith Laaks 16 Dec '14

16 Dec '14

Hi Freeswitchers. (Reposting under new thread - previous went to the wrong place.) I am running "FreeSWITCH Version 1.5.15b+git~20141215T224714Z~0b414a8de8~64bit (git 0b414a8 2014-12-15 22:47:14Z 64bit)” - updated this morning. I have the following actions in an extension being executed: <action application="info"/> <action application="bind_digit_action" data="start_echo,##,exec:info,both,self"/> <action application="digit_action_set_realm" data="start_echo"/> <action application="delay_echo" data="10"/> For the ‘info’ action at the top, I see the normal stuff printed in the console log: EXECUTE sofia/portaprod/27878881000(a)10.10.10.10 <mailto:sofia/portaprod/27878881000@10.10.10.10> info() 2014-12-15 20:52:06.137537 [INFO] mod_dptools.c:1647 CHANNEL_DATA: Channel-State: [CS_EXECUTE] Channel-Call-State: [ACTIVE] Channel-State-Number: [4] Channel-Name: [sofia/portaprod/27878881000(a)10.17.180.204 <mailto:sofia/portaprod/27878881000@10.17.180.204>] Unique-ID: [8218a229-562d-4706-85a2-988d2bae0041] <cut cut> Then follows the ‘bind_digit_action’ and ‘digit_action_set_realm’ to bind ‘##’ to executing the ‘info’ app. The ‘echo' application at the very bottom works as per normal and I get my audio echoed back. When I send “#9” - I get this in the log: 2014-12-15 21:13:32.777537 [DEBUG] switch_rtp.c:6045 RTP RECV DTMF #:800 2014-12-15 21:13:33.637537 [DEBUG] switch_rtp.c:6045 RTP RECV DTMF 9:800 2014-12-15 21:13:33.637537 [DEBUG] mod_dptools.c:132 sofia/portaprod/27878881000(a)10.10.10.10 <mailto:sofia/portaprod/27878881000@10.10.10.10> Digit NOT match binding [#9] So all normal. But here my issue. When I send ‘##’ - I only see this in the log: 2014-12-15 21:14:58.817537 [DEBUG] switch_rtp.c:6045 RTP RECV DTMF #:800 2014-12-15 21:14:59.317537 [DEBUG] switch_rtp.c:6045 RTP RECV DTMF #:800 2014-12-15 21:14:59.317537 [DEBUG] mod_dptools.c:188 sofia/portaprod/27878881000(a)10.10.10.10 <mailto:sofia/portaprod/27878881000@10.10.10.10> Digit match binding [exec:info][both] 2014-12-15 21:14:59.317537 [DEBUG] switch_core_session.c:1188 Send signal sofia/portaprod/27878881000(a)10.10.10.10 <mailto:sofia/portaprod/27878881000@10.10.10.10> [BREAK] So its finding a match, but the app to be executed never does so. If it had, I would have seen the above ‘info’ stuff printed to the console. I actually want the call transferred to another extension when the user enters ‘##’ - but that similarly does not happen. It seems I can’t get any action to actually be executed. So what do I need to set first, such that the specified ‘action’ gets executed? Hope someone can help me out here. Looking at the code, I see that in mod_dptools.c there seems to be support for “blocking” and “non-blocking” execution modes. I figured out that to ‘activate’ the ’non-blocking mode the above action needs to change to: <action application="bind_digit_action" data="start_echo,##,exec[i]:info,both,self"/> [This sets ‘exec = 2’ such that ‘switch_core_session_execute_application’ is called - otherwise ‘switch_ivr_broadcast_in_thread’ is called.] But when I try this, as soon as I send the ##, I get a core dump. Regards Keith

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Strange xml_curl behavior. Can't see more than 7 variables on POST
by Muhammad Naseer Bhatti 16 Dec '14

16 Dec '14

I am using xml_curl to generate all config dynamically and this is getting very strange, my directory looks like this, <?xml version='1.0' encoding='UTF-8' standalone='no'?> <document type='freeswitch/xml'> <section name='directory'> <domain name='10.211.55.26'> <user id='9401404' cacheable='2000'> <params> <param name='a1-hash' value='3c66e98d4b174b42d3e17669b59165c9'/> <param name='max-registrations-per-extension' value='1'/> <param name='allow-empty-password' value='false'/> <param name='sip-force-expires' value='180'/> </params> <variables> <variable name='customer_id' value='98765'/> <variable name='effective_caller_id_name' value='Mr Bhatti'/> <variable name='effective_caller_id_number' value='9401404'/> <variable name='debug_call' value='0'/> <variable name='debug_sql' value='0'/> <variable name='user_context' value='vbilling'/> <variable name='hangup_after_bridge' value='true'/> <variable name='xbypass' value='google'/> <variable name='account_type_02' value='pre'/> <variable name='account_type_03' value='pre'/> <variable name='account_type' value='pre'/> </variables> </user> </domain> </section> </document> The directory is working fine and user is able to login. I have a bunch of variables defined here. When I look at the POST on the web server, I am only able to see first 7 variable no matter what their names are. If I change order for example account_type which is last variable, and make it on top, it shows there in the XML_REQUEST but then hangup_after_bridge goes away. I see only first 7. Are there any limitation on what channel variables can be POSTed to the web server or am I missing something here. -- Muhammad Naseer Bhatti

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Get caller IP address
by Snabel Kabiya 16 Dec '14

16 Dec '14

Hi, How can i get the caller IP from session? In sip trace I've: Call-ID: 7XHCXNSJABE2NGOBAHSXRZIGPA(a)1.2.3.4 i want to get the IP 1.2.3.4 i my Lua script, anyone knows how? Thanks, Snabel

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Originate call SIP call-id
by Doron Kruh 16 Dec '14

16 Dec '14

Hi All, How can I make sure the SIP Call-Id header sent by Freeswitch when originating a new call is in the form of: uuid@host ? Thank you, Doron

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[ERR][mod_json_cdr.c][Open of json_cdr.conf failed]
by Bilal Abbasi 16 Dec '14

16 Dec '14

Hi All, I have successfully compiled mod_json_cdr but when i go to fs_cli and load mod_json_cdr i get the below error please help. Plus there is no file named json_cdr.conf in autoconfig directory. I think FS is unable to locate that file. freeswitch@internal> load mod_json_cdr +OK Reloading XML -ERR [module load file routine returned an error] freeswitch@internal> 2014-12-16 06:37:25.377189 [INFO] switch_time.c:1369 Timezone reloaded 530 definitions 2014-12-16 06:37:25.377189 [ERR] mod_json_cdr.c:598 Open of json_cdr.conf failed 2014-12-16 06:37:25.377189 [CRIT] switch_loadable_module.c:1447 Error Loading module /usr/local/freeswitch/mod/mod_json_cdr.so **Module load routine returned an error** Regards Bilal Abbasi

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Hoot get variables in a LUA Hook Event.
by Jaime 16 Dec '14

16 Dec '14

Hello, I¹m trying to use the Hook event¹s with LUA script¹s when the channel Hangup a call, but I¹m unable to get the variables, for example uuid, ani or billed seconds for the call, I¹m using freeswitch 1.4.13 on Centos 6.6 and my configuration is: In /autoload_configs/lua.conf.xml [] <hook event="CHANNEL_HANGUP_COMPLETE" script="catch-event-cdr1.lua"/> In my catch-event-cdr1.lua script I wrote: ses = freeswitch.Session(); my_uuid = ses:getVariable("uuid"); freeswitch.consoleLog("notice"," uuid=("..my_uuid..")\n") But I¹m get the following error: 2014-12-15 23:51:46.100755 [ERR] switch_cpp.cpp:724 session is not initalized 2014-12-15 23:51:46.100755 [NOTICE] switch_cpp.cpp:1328 uuid=() Also, in the wiki doc I see a reference for the ³env² Object, (https://wiki.freeswitch.org/wiki/Mod_lua#Special_Case:_env_object) but if I call it as the example in the same lua script my result is: How I call the env variable: dat = env:serialize() freeswitch.consoleLog("INFO","Here's everything:\n" .. dat .. "\n²) And I got the following error: 2014-12-16 00:00:17.440782 [ERR] mod_lua.cpp:203 /usr/share/freeswitch/scripts/catch-event-reg6.lua:27: attempt to index global 'env' (a nil value) Your help will be appreciated. Jamie Jaimecm(a)gmail.com

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