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Is external_rtp_ip=auto still valid?
by Mario G 12 Mar '11

12 Mar '11

The wiki states auto is a valid option under Sofia configuration but it is not mentioned in the vars.xml. Is this still valid? I need and way to make this work without uPNP. Thanks, Mario G

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inbound dtmf issue after upgrade
by Kevin Raison 12 Mar '11

12 Mar '11

I upgraded a freeswitch setup from 1.0.4 to 1.0.6 yesterday, and also upgraded my wanpipe drivers from 3.4.9 to 3.5.10. I am using openzap with a sangoma card. Before the upgrade, the following stanza would correctly send a caller to a mod_event_socket-based application that I have running on another machine and that caller's dtmf would flow through to the app: <extension name="Kevin_test"> <condition field="destination_number" expression="^XXXXXXXXXX$"> <action application="set" data="ringback=${us-ring}"/> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="bridge" data="sofia/internal/5000(a)prod.XXXXXXX.com"/> </condition> </extension> After the upgrade, everything but inbound dtmf works as it should, but the only way that I can get inbound dtmf to work is by adding <action application="start_dtmf"/> to the stanza. This is not ideal and I would like to go back to not using in-band dtmf if possible. In my log files, I see absolutely no dtmf being registered when a caller calls in. However, if someone calls out from a desk phone and then transfers the person that they called to sofia/internal/5000(a)prod.XXXXXXX.com, dtmf works just fine. Any ideas? Thanks! Kevin

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How to timeout att_xfer?
by Boris Kovalenko 12 Mar '11

12 Mar '11

Hello! Is there a possibility to timeout att_xfer application? For example -> A calls B, B does att_xfer to C, but C doesn't answer for a 10 seconds so B is bridged again to A (of course if B is still off hook). -- С уважением, Борис Коваленко ЗАО "Тагнет" (3435) 494991

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G.711.0
by Steve Underwood 12 Mar '11

12 Mar '11

Hi, Has anyone seen G.711.0 in real world use? The spec was published quite a while ago, but as far as I can tell there is no RFC defining the SDP and RTP details needed to deploy it, and nobody advertises that they support it in their products. Steve

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freeswitch -nc -nosql
by mazilo 12 Mar '11

12 Mar '11

It had been a while I didn't get a chance to upgrade my FS git until today. With a newly built FreeSWITCH Version 1.0.head (git-0f7682a 2011-02-25 21-06-22 +0100), at least a new problem surfaces where FS Segmentation fault as shown below: 2011-02-25 21:30:34.277865 [CONSOLE] switch_core.c:1764 _____ ______ _____ _____ ____ _ _ | ___| __ ___ ___/ ___\ \ / /_ _|_ _/ ___| | | | | |_ | '__/ _ \/ _ \___ \\ \ /\ / / | | | || | | |_| | | _|| | | __/ __/___) |\ V V / | | | || |___| _ | |_| |_| \___|\___|____/ \_/\_/ |___| |_| \____|_| |_| ************************************************************ * Anthony Minessale II, Michael Jerris, Brian West, Others * * FreeSWITCH (http://www.freeswitch.org) * * Paypal Donations Appreciated: paypal(a)freeswitch.org * * Brought to you by ClueCon http://www.cluecon.com/ * ************************************************************ 2011-02-25 21:30:34.278007 [CONSOLE] switch_core.c:1767 FreeSWITCH Version 1.0.head (git-0f7682a 2011-02-25 21-06-22 +0100) Started. Max Sessions[1000] Session Rate[30] SQL [Enabled] Segmentation fault @DockStar:/# If I add a -nosql switch, i.e. freeswitch -nosql, it runs OK but it keeps dumping the following message on the CLI: 2011-02-25 21:36:19.378416 [ERR] switch_core_sqldb.c:1721 Error Opening DB! So, whatI do I need to tweak to make the above message go away? ----- FreeSWITCH hosted on a Seagate DockStar with OpenWRT. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/freeswitch-nc-nosql-tp6066718… Sent from the freeswitch-users mailing list archive at Nabble.com.

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session:record / prevent DTMF tones from being recorded
by Matt Stockton 11 Mar '11

11 Mar '11

Hi all, I have a freeswitch lua script that is recording and I want the recording to be terminated when the user presses the '#' key. That is all working fine for me, but the recording always has the DTMF tone at the end of it. I am wondering, is there a way to instruct freeswitch to not record the DTMF tones? Any help is appreciated. For reference, here are some excerpts from my lua script: function inputHook(s, type, obj) freeswitch.consoleLog("INFO", "On input"); if (inRecording and type == "dtmf" and obj['digit'] == '#') then inRecording = false; return "break"; end end ..... session:setInputCallback("inputHook", ""); ..... session:recordFile(recordFilePath, 202, 500, 25); Thanks in advance

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Call back from voice mail creates loop
by Tim St. Pierre 11 Mar '11

11 Mar '11

Hello, I'm trying to set up the option in voicemail so that users can return a call directly from VM. In my voicemail.conf.xml profile I have specified the dialplan as XML and the context as "internal", which is what I use for calls coming from our endpoints or authenticated users. In our internal dialplan, we do some pattern matching and call an appropriate lua dialplan script to do build a set of instructions. Problem is, the lua scripts are looking at the destination_number variable to decide where to send the call. This variable is still set to "voicemail" from when the user called into their voicemail. Shouldn't the voicemail app have set that to the number we want to call back? I used the info app to look at the variables, and I didn't see anything obvious. Is destination_number the correct variable to use in a lua script that generates a dialplan? -Tim

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Internal extensions placing call on hold drops immediately
by Mario G 11 Mar '11

11 Mar '11

I don't know when this started but was working fine at one time. If a call is made between extensions and either extension places the call on hold, it states "trying to hold". I installed the git from 3/9/11 before posting here. Could this be a Sofia problem (see FREEPBX comment) or something that FS needs to "setup" the phone to work since it works fine using a different PBX? Any help appreciated! * The phones are Linksys SPA962 * If I plug the old SPA9000 in as a PBX hold works fine for the SPA9000 extensions, only FS has the issue * All connected via HP Procurve 2520G switch * Outside calls are also affected but this is still a problem even with the router disconnected * I can ping (ICMP) from FS to the phone and from anywhere to the phone FS TRACE: I started several FS traces: sip global,internal, log 9 console debug, etc. It looked like NOTHING was sent when the hold button was pressed. WIRESHARK showed these messages stacking up when the hold is pressed src 1.123.1.23 (phone) dest 1.123.1.7 (FS) Proto ICMP Destination unreachable (Port unreachable) Someone updated FREEPBX in FEB 2011 and had the same issue with SPA phones. http://www.freepbx.org/forum/freepbx/users/spa-941-942-drop-call-when-place…

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Strange Behavior with Channel Variable [session:setVariable() vs <action application="set" ... /> ]
by Sidharth Kshatriya 11 Mar '11

11 Mar '11

Dear All, I am using a little known but very useful channel variable called "playback_timeout_sec" In Lua if I do: session:setVariable("playback_timeout_sec", "10") and subsequently in a macro I have: <action function="play-file" date="/some/filename.mp3" /> Then the filename.mp3 stops playing automatically after 10 seconds. This is very useful if you want to play short excerpts of a file that could be very long. But if I do: <action application="set" date="playback_timeout_sec=10" /> <action function="play-file" date="/some/filename.mp3" /> My file does NOT stop playing after 10 seconds. This is weird given that <action application="set" .. /> is doing exactly the same thing as session:setVariable() ... right? My question 1. What could be the problem with application set? 2. Is there a way to automatically stop playback after a certain number of seconds for a file? (without doing tedious things like create a separate 10second version of an mp3 using ffmpeg etc.) Thanks, Sidharth -- Sidharth Kshatriya www.sidk.info

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Working configuration for FS behind NAT?
by GillesToo 11 Mar '11

11 Mar '11

Hello I'd like to set things up, where... - FS and local SIP extensions are on a private LAN behind a NAT firewall, - remote employees are allowed to register with Freeswitch so they're part of Freeswitch, - none of the routers support UPnP or NAT-PMP, and - to lower load on the Freeswitch server, I'd like to have RTP packets flow directly between extensions instead of going through Freeswitch, like Asterisk does with its "canreinvite=no" option. However, Freeswitch should remain in the loop somehow so that users can access PBX features during their call. Here's the picture: http://img577.imageshack.us/img577/1795/freeswitchnat.png Apparently, changes to Freeswitch made http://wiki.freeswitch.org/wiki/General_NAT_example_scenarios this page obsolete. If Freeswitch can do this, does someone have a basic, working configuration that I could use to get started? Thank you. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Working-configuration-for-FS-… Sent from the freeswitch-users mailing list archive at Nabble.com.

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