- FreeSWITCH-users - FreeSWITCH Mailing Lists

valet_park timeout
by Bob Smith 13 Jan '12

13 Jan '12

Hello All, I found an old thread from October 2010 entitled "valet_park timeout and spot announcement" (http://lists.freeswitch.org/pipermail/freeswitch-users/2010-October/064143.…) I also found this tread from March 2011 entitled "valet parking improvements" (http://lists.freeswitch.org/pipermail/freeswitch-dev/2011-March/004778.html) Could anyone tell me whether : 1/ Timeouts have made it into the tree ? As far as I can tell they have not ? 2/ Are there perhaps now better way to achieve the same goal ? (e.g. looking at the console, I see a variable generated called valet_ticket ?) Basically I don't want to run into the same problems Jeremy did, i.e : "If the call is transferred somewhere else first or picked up from the first park on a different extension and then parked again, it doesn't recall the second parker." Thanks Bob

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Dial by IP
by Brian Foster 13 Jan '12

13 Jan '12

I have some phones on my network that are capable of dialing a SIP URI. It seems by default FreeSWITCH doesn't allow this, as it just picks up everything before the @ and puts it as the destination number. Is there a way to allow this? What would be needed in the dialplan?

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Re: [Freeswitch-users] script vs scripts directories, what's the difference ? (found it !.... 023818bc)
by Bob Smith 12 Jan '12

12 Jan '12

* Marc Olivier Chouinard * 023818bc… on master * Diff * Annotation * -1 * +1 more FS-302 Error in initial patch.. default scripts folder was missing the s at the end

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Improving mod_conference for H.264 video // bounty?
by Roland Hänel 12 Jan '12

12 Jan '12

Dear all, We're looking for a way to improve the user experience with FreeSwitch's mod_conference when video (H.264) is involved. After some investigation in mod_26x's source code it seems that the creator actually had the idea to switch video (based on who has the floor) only on H.26x keyframes, but this was not fully implemented (i.e. it doesn't really work well). Maybe someone from the FreeSwitch guys could advise me about the way how to put up a bounty for improvements in this area. I found some boundy lists in the wiki, other things in Jira, and I'm not sure where to post my whishes (and maybe also a hint what amount of $'s seems appropriate here). Basically, that's what we need: - switch video source only on H.264 keyframes to avoid artifacts when the stream switches to a new floor member - feature to throttle floor changes in conference (i.e. floor is only changed after some time of activity, to prevent very frequent video stream changes) - investigation whether the codec parameter negotiation can be influenced/improved for H.264 passthru, e.g. to allow for specific profiles in SDP a=fmtp parameters Greetings, Roland

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Conference Anonymous BBB
by Torstein Knutsen 12 Jan '12

12 Jan '12

Hi I am thinkering with FreeSWITCH and BigBlueButton conferences. I have on server running FS, and the BBB server is connected via EventSocket. Now, my challenge is regarding to Private numbers. I can't seem to get the conference to respect privacy. I set (in dialplan) <action application="set" data="caller_id_number=0000"/> <action application="conference" data="11111111@webconf"/> BUT : While monitoring the Event socket on the CUSTOM Actoin:add-member, the caller_id number is back to the original. This cases my setup to leak Privacy marked numbers. I'm not running (FreeSWITCH Version 1.0.head (git-8fde25c 2011-10-20 13-15-58 -0500)) Is there any way around this ? Thank you ! Torstein

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script vs scripts directories, what's the difference ?
by Bob Smith 12 Jan '12

12 Jan '12

My freeswitch base directory contains the following directories : bin conf db grammar htdocs include lib log mod recordin run script scripts sounds storage What's the difference between "script" and "scripts" ? Seems a bit silly to me to have such confusingly similar names in the same parent directory !

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Multi domains issue
by Miha Zoubek 12 Jan '12

12 Jan '12

Hi, plase help me out why I am getting: 2012-01-11 15:42:56.746075 [WARNING] sofia_reg.c:1359 SIP auth failure (REGISTER) on sofia profile 'internal' for [0123456@fs1.test.si@fs1.xxxx.si] from ip xxx.xxx.xxx.xxx I do not know how to fix this or what I am doing wrong. $${domian} I chaged in fs1.test.si http://pastebin.freeswitch.org/18117 In user auth for phone I put 0123456(a)fs1.test.si, Thanks! Br, Miha -- Best regards / Lep Pozdrav Miha Zoubek Softnet d.o.o.

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Freeswitch CDR Packet counters
by Wesley Akio 12 Jan '12

12 Jan '12

Hi all, I'm trying to pinpoint the origin of a call quality problem. The XML CDR has some information on it but I'm having some trouble on finding documentation for some of the fields. <rtp_audio_in_packet_count>6801</rtp_audio_in_packet_count> <rtp_audio_in_media_packet_count>6528</rtp_audio_in_media_packet_count> <rtp_audio_in_skip_packet_count>37</rtp_audio_in_skip_packet_count> <rtp_audio_in_jb_packet_count>0</rtp_audio_in_jb_packet_count> <rtp_audio_in_dtmf_packet_count>0</rtp_audio_in_dtmf_packet_count> <rtp_audio_in_cng_packet_count>0</rtp_audio_in_cng_packet_count> <rtp_audio_in_flush_packet_count>273</rtp_audio_in_flush_packet_count> Most of this fields are self explanatory(I hope to be right on this one), flush_packet_count is the one I cant figure out with certainty. Best, Wesley Akio

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mod_curl holding call
by Oz Mortimer 12 Jan '12

12 Jan '12

Hi Mailing list, I am having an issue with curl where the app execution doesn't return where there is an issue resolving domain, even if the caller has hungup. The billable duration in the CDRS along with the end time stamp both reflect the script end time rather than call end time. Here is an example to force the issue. Edit /etc/resolve and set a name server to 9.9.9.9 Create a simple lua script session:answer() session:execute("curl","http://www.google.com") When calling this script via softphone and hanging up during the CURL, I see 2012-01-12 17:29:23.070105 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/testing@myip [BREAK] 2012-01-12 17:29:23.070105 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/testing@myip [BREAK] 2012-01-12 17:29:23.070105 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/testing@myip [BREAK] 2012-01-12 17:29:23.070105 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/testing@myip [BREAK] And then nothing, until I assume the CURL call ends (including the Callstate Change ACTIVE -> HANGUP) If I then look at the cdrs, they also show billable duration inclusive of the curl timeout even though the call ended some seconds before. I tried setting a hangup hook in the lua but this too doesn't get called when the call terminated. It seems like the state machine for the call gets stuck until curl completes. I've just downloaded the git head and the issue is present there too. Obvious question, shouldn't billable duration and end stamp reflect the actual call? Or is there something I need to do that I've currently missed. Is this issue specific to curl, or could it happen in other modules? Any pointers? Many Thanks Oz.

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FreeSWITCH in a University Environment
by Gabriel Kuri 12 Jan '12

12 Jan '12

Does anyone know of any Universities running FreeSWITCH, or other open source VoIP for that matter, for local call routing between handsets? We're looking at replacing our old Avaya system or upgrade it, and the forklift upgrade from Avaya is ridiculously expensive (no surprise). We'd like to replace our Avaya system with a combination of OpenSIPS and FreeSWITCH and some Cisco routers for external PSTN access, but it's going to be a tough sell to our CIO, unless we can show someone else has done it already. Any pointers to other Universities would be great. Cheers, Gabe

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