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Return from lua two values.
by Mimiko 20 Feb '13

20 Feb '13

Hello. If using <action application="set" data="MY_VAR=${lua(helloworld.lua $1)}" /> I get one value from lua to MY_VAR. Can I get two values to two variables in one call? Thanks. -- Mimiko desu.

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Re: [Freeswitch-users] Reading incoming skype chat messges
by David Matthews 20 Feb '13

20 Feb '13

<param name="report_incoming_chatmessages" value="false"/> in autoload_configs/skypopen.conf.xml was the problem, fixed by changing that to "true" -- David Matthews mail(a)dmatthews.org

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Problems with sip provider
by Johan Helsingius 20 Feb '13

20 Feb '13

I am using 3 different SIP providers. 2 of them work just fine, but I have never managed to get the third one, telfort (NL) to work with Freeswitch. Here is my gateway definition: <gateway name="telfort"> <param name="username" value="020NNNNNNN"/> <param name="auth-username" value="020NNNNNNN"/> <param name="from-user" value="020NNNNNNN"/> <param name="realm" value="tel.telefoniedienst.nl"/> <param name="from-domain" value="tel.telefoniedienst.nl"/> <param name="proxy" value="sip.telefoniedienst.nl"/> <param name="password" value="XXXXXXX"/> </gateway> sofia status shows: external::telfort gateway sip:020NNNNNNN@sip.telefoniedienst.nl REGED but when I try to dial out on that provider, I get: 2013-02-17 18:04:19.216205 [NOTICE] sofia.c:6385 Hangup sofia/external/06MMMMMMMM [CS_EXCHANGE_MEDIA] [CALL_REJECTED] The strange thing is that everything works OK when I configure the parameters into the ADSL modem/gateway - a ZyXEL P-2601HN-F1: SIP Server Address: sip.telefoniedienst.nl Port: 5060 REGISTER Server Address: sip.telefoniedienst.nl Port: 5060 SIP Service Domain: tel.telefoniedienst.nl Port: 5060 RTP Port Range: 16384-32767 Enable Early Media Account Number: 020NNNNNNN Username: 020NNNNNNN Password: XXXXXXX Compression Type: G.711ALaw, G.729, G.722 Active G.168 Echo Cancellation so somehow I am not mapping that information properly into the Freeswitch conf... Julf

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Pre-loading Voicemail prompts and names
by Steven Schoch 20 Feb '13

20 Feb '13

We're switching from an old phone system to Freeswitch, so I converted everyone's voicemail greetings and recorded names using sox, and dropped the WAV files to greeting_1.wav and recorded_name.wav in their respective voicemail directories. The voicemail module didn't automatically recognize these (even on restart), but I was able to set everyone's greeting by connecting to their voice mailbox from their extension and selecting greeting 1. I don't know how to do this to make it take their recorded name. I want the directory module to play their name instead of spelling it. How to I tell Freeswitch that there are recorded_name.wav files sitting there ready to be used? -- Steve

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valet_parking - aastra phones
by Lloyd Aloysius 19 Feb '13

19 Feb '13

Hi All: I have a problem with the valet parking and Aastra phones. There is no issue with Polycom 1. Use A Dial User B 2. User B answer the call 3. User B attn xfer to the parking lot 6000 4. User B hear the lot number and complete the transfer 5. *User A also hear the Lot Number . Is there anyway to avoid the announcement for the User A.* http://pastebin.freeswitch.org/20610 The 6000 Extension execute two times, by User B and User A I try with the blind xfer. User A hear the announcement. Any help is appreciated. Thanks Lloyd

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Benchmark and making session and bridging virtual devices....
by Sayyed Mohammad Emami Razavi 19 Feb '13

19 Feb '13

How to test server capacity to load many freeswitch sessions with ready voice recording and ... features? I want to make at least 2000 virtual sessions(1000 concurrent calls). How to do this? Does you have any idea?

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CHANNEL_DATA variables
by Archana Venugopan 19 Feb '13

19 Feb '13

Hi, I could see that I have been getting the below variables in logs when I dial my voicemail number. But when I make internal or external calls, I don't see these variables in logs. 910672e4-6ec8-11e2-8027-7dea0d27034c 2013-02-04 12:44:08.027128 [INFO] mod_dptools.c:1421 CHANNEL_DATA: Channel-State: [CS_EXECUTE] Channel-Call-State: [RINGING] Channel-State-Number: [4] Channel-Name: [sofia/internal/367(a)vectone.uk01.net] Unique-ID: [910672e4-6ec8-11e2-8027-7dea0d27034c] Call-Direction: [inbound] Presence-Call-Direction: [inbound] Channel-HIT-Dialplan: [true] Channel-Presence-ID: [367(a)vectone.uk01.net] Channel-Call-UUID: [910672e4-6ec8-11e2-8027-7dea0d27034c] Answer-State: [ringing] Caller-Direction: [inbound] Caller-Username: [367] Caller-Dialplan: [XML] Caller-Caller-ID-Name: [Test] Caller-Caller-ID-Number: [367] Caller-Network-Addr: [192.168.5.129] Can anyone please tell me how am I to get these variables if i make internal or external calls as well as like my voicemail call? Thanks Regards, Archana

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Running sox from within lua
by Alex Lake 19 Feb '13

19 Feb '13

Having a spot of bother with this bit of lua: result = os.execute(string.format([[sox --norm %s %s.new]], file_to_normalise, file_to_normalise)) It's returning -1. I believe freeswitch is running as root and that the sox file is in that location and is runnable. Any ideas?

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multiple inbound trunks
by shayne.alone＠gmail.com 19 Feb '13

19 Feb '13

Hi all; I need to run a switch will handle calls from 30 different provides ( some of them have more that IPaddress as originator ) to two gate. 30 SIP providers =====> FS ==> 2 SIP gate cos there is no registration, first way is to open them in acl.conf.xml. but I like to know if there any other way ( for incoming trunks without registration ) to open / accept there calls and distinct them before dial plan to route them better and of course easily log parsing! simply management is there any idea? tnx

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How SIP clients able to establish a call with out Freeswitch Server?
by Muqtheear S 19 Feb '13

19 Feb '13

1. I stopped Freeswitch server. User Agent client 1 was able to call User Agent client 2. 2. What configurations I need to change , so that Sip clients can use Freeswitch server.? Thanks & Regards, Muqtheear.S

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