February 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

demo_ivr cannot find sound files via relative paths
by Robert Hadley 11 Feb '10

11 Feb '10

Hi, It appears a recent change (possibly the new sounds_dir variable or the new ivr_menu folder?) may have broken relative sound file paths in the IVR. I built a today's trunk version and installed to the default location. Using the default conf files the demo_ivr cannot find files based on the relative paths specified in ivr_menus/demo_ivr.xml. [root@TEO-UCM-T2 conf]# cat ivr_menus/demo_ivr.xml <include>   <menu name="demo_ivr" greet-long="phrase:demo_ivr_main_menu" greet-short="phrase:demo_ivr_main_menu_short" invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav" exit-sound="voicemail/vm-goodbye.wav" 2010-02-10 10:32:15.300337 [DEBUG] switch_ivr_menu.c:414 Executing IVR menu demo_ivr 2010-02-10 10:32:15.300337 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en] 2010-02-10 10:32:15.300337 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[ivr/ivr-welcome_to_freeswitch.wav] (en:en) 2010-02-10 10:32:15.300337 [DEBUG] switch_ivr_play_say.c:1158 Codec Activated L16@8000hz 1 channels 20ms 2010-02-10 10:32:17.922147 [DEBUG] switch_ivr_play_say.c:1450 done playing file 2010-02-10 10:32:18.042149 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[ivr/ivr-this_ivr_will_let_you_test_features.wav] (en:en) 2010-02-10 10:32:18.042149 [DEBUG] switch_ivr_play_say.c:1158 Codec Activated L16@8000hz 1 channels 20ms 2010-02-10 10:32:19.962158 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 2:800 2010-02-10 10:32:19.962158 [DEBUG] switch_ivr_play_say.c:1450 done playing file 2010-02-10 10:32:20.082156 [DEBUG] switch_ivr_menu.c:329 waiting for 3/4 digits t/o 2000 2010-02-10 10:32:20.120617 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 2:400 2010-02-10 10:32:20.442158 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 2:800 2010-02-10 10:32:20.682162 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 2:800 2010-02-10 10:32:20.682162 [DEBUG] switch_ivr_menu.c:376 digits '2222' 2010-02-10 10:32:20.682162 [DEBUG] switch_ivr_menu.c:470 action regex [2222] [/^(10[01][0-9])$/] [0] 2010-02-10 10:32:20.682162 [DEBUG] switch_ivr_menu.c:560 IVR menu 'demo_ivr' caught invalid input '2222' 2010-02-10 10:32:20.682162 [ERR] mod_sndfile.c:194 Error Opening File [/usr/local/freeswitch/sounds/ivr/ivr-that_was_an_invalid_entry.wav] [System error : No such file or directory.] 2010-02-10 10:32:21.701505 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en] 2010-02-10 10:32:21.701505 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[silence_stream://1000] (en:en) 2010-02-10 10:32:21.701505 [DEBUG] switch_ivr_play_say.c:1158 Codec Activated L16@8000hz 1 channels 20ms 2010-02-10 10:32:22.700352 [DEBUG] switch_ivr_play_say.c:1450 done playing file Regards, Robert

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Skypiax latency
by Peter P GMX 11 Feb '10

11 Feb '10

Hello, I have a problem with latency and mod_skypiax Skype=>SIP is always fine (~0.3sec) SIP => Skype is always bad (~2-4 sec) I would expect that latency in both directions should be the same. Anybody has discovered this before and has a solution? The scenario is as follows: SIP-UA => FreeswitchServer=>Freeswitch/SkypIaxServer=>Skype Both freeswitch servers are in the same LAN, so latency should be low. Best regards Peter

3 3

is there a command to get a list of registered accounts?
by Paul Levin 11 Feb '10

11 Feb '10

3 3

call die after 30 seconds in lua script
by Eder Souza 11 Feb '10

11 Feb '10

Hi list im testing the script welcome.lua from http://wiki.freeswitch.org/wiki/Lua_Welcome_IVR_Example but when do a transfer my call drop in 30 seconds just in script lua!! callig one ramal (ex: 12345) direct in x-lite the call works !! When make a session:execute("transfer","12345") in the script lua the call drop after 30 seconds!! why ?? Att, Eng Eder de Souza

2 2

multiple isdn msn on an extension with a single sip account
by wessels 11 Feb '10

11 Feb '10

Hi, I'm evaluating the replacement of an ISDN pbx. One of the things I can't figure out is a suitable replacement for the msn numbers on a phone with a single sip account. The current ISDN phones can be programmed to recognize up to eight different msn numbers. The user of the phone can see and hear which msn number was called. So I've got a BRI or PRI , with multiple msn, feed that into freeswitch and simple extensions like the linksys spa922. I want the extension to ring on more than one msn but I want to be able to see which msn was originally called in the display. Can this be setup using freeswitch and a simple phone??? Any directions would be most helpful. Thank you, Wessel s

1 0

Freeswitch and G729
by Max Bridgewater 11 Feb '10

11 Feb '10

Hi, Some quick questions related to the upcomming Freeswitch G729 support: 1) When can it be tried? 2) Does it support lower bit rate extensions such as D, F, H, I, C? Thanks, Max.

2 1

can't see sip messages in the log file
by Nikolay Kondratyev 11 Feb '10

11 Feb '10

Hi all, I have a little problem with FreeSWITCH Version 1.0.5-20100209-0400 (16587M). I'd like to see SIP messages in the log file. and I tried sofia profile internal siptrace on sofia loglevel all 9 console loglevel 9 but alas i can only see what SDP is used.. The configuration is default with the exception for some new local extensions and mod_h323 compiled in. Am I missing something? Can anybody please advise how to include sip messages into the log file? (I thought that "sofia profile internal siptrace on" should be enough, but alas..) Thanks in advance, Nikolay.

4 6

Seeking Advice on SIP Phones like Aastra
by Darren C. 11 Feb '10

11 Feb '10

Tim, Many thanks for your response. I posted this message on the Dev list and all I heard was crickets. I would think a web GUI for a phone would be in demand by the FS community.... We are doing something similar to what you described. We’re developing a rather complex IVR/Switching application and it currently does all its database writes via our Web Service to an MS SQL database. We have a web site that is updated with call details via the web service’s backend database. From this web interface a user can see counts of voicemails, see call activity, play voicemails, see calls in progress, record calls, etc. It’s a specialized application so it doesn’t have every PBX feature but this is what we wanted to do with a high-end SIP Phone to replace our office PBX. Currently we have an ESI (Estech) E-Class PBX that uses normal digital phones as well as proprietary VOIP (non-SIP) phones. I think these really nice SIP phones with huge color touchscreens would be much better than even high-end proprietary digital phones + we’d get all the benefits of FS. We’ll just need to add all the basic PBX capabilities to the phone’s GUI to see how many voicemails are waiting, how many lines are in use, button for call transfer, etc. As you mentioned: “you can do a DB query to find out what voicemail messages a user has, or what calls are active, etc.” We are keeping track of all this ourselves via our web service’s backend database. I’m not sure I need to do this for everything but we are. It’s a multi-tenant system with hundreds of tenants so I’m guessing I might lose some needed relationships by querying FS but I’m going to re-visit this…I will make sure I can’t just query FS like you’re doing for some of these things…we’ve never turned on ODBC or even looked for the documentation as to what FS stores. We have to run FS in Linux for some Sangoma stuff but we’re Windows people so that is another reason we store via web services to an MS SQLbackend. I was worried I’d get one of these fancy phones and find out it doesn’t support important SIP/FS features rendering the color touchscreen useless. I’ve never owned one of these SIP phones, just used various softphones. But thanks to you I’ll get an Aastra 6739i and give it a try. I have a fulltime programmer working on this system on and off now for over a year but this has been mostly developing the IVR. We haven't made any attempt at using FS as a PBX. So based on your comments I think I’ll purchase an Aastra 6739i and develop a custom SIP Phone GUI interface with FS. If you or others would like to collaborate on an Aastra 6739i phone GUI for FS, feel free to contact me. We can try to make it extensible for other phones as well. My email is ustcor<remove this junk>poration@yah<remove this too>oo.com. Thanks, teldev

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Interesting article on OSS telephony
by Michael Collins 11 Feb '10

11 Feb '10

Check out the article on TMCnet.com, linked-to here: http://www.freeswitch.org/node/238 -Michael

1 0

How to record the call upon successful bridge
by Jingwei Yang 11 Feb '10

11 Feb '10

Hi, I'm using uuid_bridge to bridge two calls. May I know how to start recording only when the bridge succeeds? Thanks, -Jingwei

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