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ClueCon 2011 - Assistance w/ posting videos
by Michael Collins 19 Aug '11

19 Aug '11

Hello all! I am looking for a volunteer to assist me with some Drupal/Web work on cluecon.com. I would like to be ready to post the videos as they become available and to do that I need to do some layout work on the cluecon site. Anyone with any kind of CMS/Drupal/Web dev skills would be a perfect candidate to help. If you're in a position to assist please contact me off list. Thanks! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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FreeSWITCH Janitorial Items
by Michael Collins 19 Aug '11

19 Aug '11

Hello all! We have need of your skills with a few sub-projects. Sorry, nothing glamorous, just some behind-the-scenes work. First off, I need to know if anyone out there has experience with UberCart for Drupal. If so, please contact me off list so we can discuss the items I need help with. Secondly, I have a housekeeping task. We have some new "standard vocabulary" files for Callie US English. We need to add them to phrase_en.xml. However, there are quite a few of them and some of them are the same recording with a different inflection. For example, the inflection on the words "one" and "message" are different in these phrases: "To leave a message, press 1." "Press 1 to leave a message." I have a list of all the file names and what's in them but there's a bit of work that needs to be done to document the inflection for each phrase. We also need to settle on a naming convention. If you are in a position to assist with this please contact me off list and I will give you more information. Thanks! -Michael

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Fetching user-variables
by Christian Benke 19 Aug '11

19 Aug '11

Hi! I'm working on a redirect script and ran into a problem with toll_allow: A tries to reach B but B has redirected to C B is allowed to make external calls(via the user-specific-variable "toll_allow"), but A is not(Could be a external caller or a directory-user with restrictions). B's redirection will not work as she intended, as the dialplan is restricted through toll_allow. B is responsible for the external call of A to C, therefore her accountcode and her toll_allow should apply - unfortunately the dialplan doesn't know that the call has been redirected by B. Right now i don't know how to solve this with FreeSWITCH. Is there a way to fetch user-specific-variables of B, like "toll_allow", in a call that was not directly initiated by B? Best regards Christian

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Newbie - external originating leg has no sound when calling
by Charles Bujold 19 Aug '11

19 Aug '11

Weird situation, we have an external location which if they originate the call we cannot hear them, but if we originate the call to them everything works fine. The configuration we have is Sense as firewall that sends and has all stipulated Freeswitch ports open and forwards incoming ports to the Freeswitch server (Ubuntu/Freeswitch/fusionpbx) Internal to internal call works Internal call to the external location works Both parties can talk without issues. External location to Internal location: we do not hear them, but they can hear us. External location call to external location : the originating caller cannot be heard, the second leg works perfectly. (This suggest that it is not a firewall issue but a configuration issue.) External location calls to the voicemail and can record without problems a recording. The problem seems to be limited to the first leg of the call, in that if it originates from an external location the first leg cannot be heard. Any suggestions on how to fix would be appreciated. Thanks! Cjb

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Freetdm bridge to SIP DTMF problem
by Jakub Ouhrabka 19 Aug '11

19 Aug '11

Hi, we've changed OpenZAP to FreeTDM sometime ago and from that time we're experiencing problems with DTMF. We're bridging incoming ISDN30 calls from Sangoma A108DE card + libri + FreeTDM to SIP outgoing calls. Freeswitch is recognizing DTMF correctly but DTMF is not send to outgoing SIP call - it's not present in RTP stream trace (we've setup RFC2833 DTMF signalling). When bridging the call back to ISDN30 DTMF works correctly. We're using latest versions of all software as of 1 month ago. Below are attached two snippets of logs: first is bridging back to ISDN30 where DTMF is sent to called party. Second is bridging to SIP where DTMF is not sent to the called party. We've tried it with both <action application="start_dtmf" /> on and off with no success. Any pointers how to investigate the issue? Thanks, Jakub Log snippets: ISDN30 to ISDN30 bridge - DTMF ok 2011-08-18 12:31:45.950158 [DEBUG] ftmod_wanpipe.c:1545 [s1c11][1:11] read wanpipe event 3 2011-08-18 12:31:45.950158 [DEBUG] ftmod_wanpipe.c:1415 [s1c11][1:11] Queuing wanpipe DTMF: 5 2011-08-18 12:31:45.950158 [DEBUG] ftdm_io.c:3504 [s1c11][1:11] Queuing DTMF 5 (debug = 0) 2011-08-18 12:31:45.950158 [DEBUG] mod_freetdm.c:733 Queuing DTMF [5] in channel FreeTDM/1:11/XXXXXXXXX 2011-08-18 12:31:45.970623 [DEBUG] switch_ivr_bridge.c:391 Send signal FreeTDM/1:5/XXXXXXXXX [BREAK] 2011-08-18 12:31:45.991084 [DEBUG] ftdm_io.c:3694 [s1c5][1:5] Generating DTMF [5] ISDN30 to SIP bridge - DTMF not forwarded 2011-08-18 12:34:05.287480 [DEBUG] ftmod_wanpipe.c:1545 [s1c29][1:29] read wanpipe event 3 2011-08-18 12:34:05.607708 [DEBUG] ftmod_wanpipe.c:1545 [s1c29][1:29] read wanpipe event 3 2011-08-18 12:34:05.607708 [DEBUG] ftmod_wanpipe.c:1415 [s1c29][1:29] Queuing wanpipe DTMF: 6 2011-08-18 12:34:05.607708 [DEBUG] ftdm_io.c:3504 [s1c29][1:29] Queuing DTMF 6 (debug = 0) 2011-08-18 12:34:05.607708 [DEBUG] ftmod_libpri.c:1586 -- Caught Event span 1 18 (KEYPAD_DIGIT) 2011-08-18 12:34:05.628171 [DEBUG] mod_freetdm.c:733 Queuing DTMF [6] in channel FreeTDM/1:29/XXXXXXXXXXXXXXX 2011-08-18 12:34:05.628171 [DEBUG] switch_ivr_bridge.c:391 Send signal sofia/external/XXXXXXX@XXXXXXX:5060 [BREAK]

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INFO: Wiki update
by Michael Collins 19 Aug '11

19 Aug '11

Hello all, We have been experimenting with some access controls on the wiki. If you get a dialog box asking for a password then use the same credentials as you would for pastebin.freeswitch.org. You should only see the dialog box when logging in or editing pages. Just searching the wiki does not cause the dialog to show up. We are trying to cut down on the spammers. Let me know if you run into any weirdness. Thanks, MC

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Cannot transfer incoming call
by Tomasz Hyziak 19 Aug '11

19 Aug '11

Hi Transfers from local to local user work without problem, but when a person call in via sip provider (selects call type via ivr and he is put into fifo) and somebody answers this call, he is unable to transfer this call to another person (via *1/*3/*4). When he presses *, he is disconnected. Any ideas why ? 2011-08-19 14:55:17.475054 [DEBUG] switch_rtp.c:3303 RTP RECV DTMF *:800 2011-08-19 14:55:17.475054 [DEBUG] switch_channel.c:2775 (sofia/external/693xxxxxx(a)trunk.dialog.pl) Callstate Change ACTIVE -> HANGUP -- pozdrawiam - Tomasz Hyziak

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play
by Sam 19 Aug '11

19 Aug '11

Hello, What could be the reason of the below error , i am trying to pass command from the console . I see the file on the system located /usr/local/freeswitch/sounds/en/us/callie/ivr-call.wav . freeswitch@internal> conference conference_1 play ivr-call.wav (play) File: ivr-call.wav not found. 2011-08-19 05:22:39.455963 [ERR] mod_sndfile.c:194 Error Opening File [/usr/local/freeswitch/sounds/en/us/callie/ivr-call.wav] [System error : No such file or directory.] freeswitch@internal> freeswitch@internal> conference conference_1 play ivr-call (play) File: ivr-call not found. 2011-08-19 05:28:00.259488 [ERR] switch_core_file.c:112 Unknown file Format [/usr/local/freeswitch/sounds/en/us/callie/ivr-call] Regards Sam

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Best tabletop conference phone
by Luis F Urrea 19 Aug '11

19 Aug '11

Hi all, After using a Snom Meeting Point for 9 months that suddenly went dead while connected to PoE which in turn is properly connected to UPS my customer is extremely upset and frustrated to that experience. He is asking me for a quote on the best possible tabletop conference phone that we can use with FS since he is not even interested in processing the Meeting Point warranty. Please please please,your input is appreciated!!! Regards, Luis

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having problems building mono 2.10
by covici＠ccs.covici.com 18 Aug '11

18 Aug '11

Hi. I am using gentoo, so I have to build mono by compiling and when I try to do this I get the following error: make[8]: Entering directory `/var/tmp/portage/dev-lang/mono-2.10.2-r1/work/mono-2.10.2/mcs/tools/gacutil' MCS [basic] gacutil.exe Inconsistency detected by ld.so: dl-deps.c: 622: _dl_map_object_deps: Assertion `nlist > 1' failed! make[8]: *** [../../class/lib/basic/gacutil.exe] Error 127 Any assistance on this would be appreciated. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici(a)ccs.covici.com

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