- FreeSWITCH-users - FreeSWITCH Mailing Lists

IMPORTANT NEWS: blacklist.pl script not working because of infiltrated.net expiration
by Michael Collins 19 Nov '10

19 Nov '10

Hello all, It seems that infiltrated.net expired yesterday. This is the source of the data for our blacklist.pl script. The script pulls two files: http://www.infiltrated.net/voipabuse/addresses.txt http://www.infiltrated.net/voipabuse/netblocks.txt It uses these two files for building a better ACL that rejects known bad guys. Since these files are not currently available the script will receive bogus information and pollute the ACL. If you have run the blacklist.plscript prior to the failure and your system is working properly then please make a backup copy of your freeswitch.xml.fsxml file. This will contain the proper ACL nodes. DO NOT RESTART FREESWITCH OR RELOAD XML BEFORE BACKING UP THIS FILE! If you do then you will lose the information. I am researching this right now and I will update you as soon as I have some information. In the meantime if any of you have a freeswitch.xml.fsxml file with the blacklist info please zip and email me offlist a copy of that file and I will see if I can't extract the IPs and create a temporary workaround. I will also be updating the blacklist.pl script to do some error checking so that it doesn't pollute the ACL. Last item: if you have inside knowledge of what's happening with the VoIP Abuse project please contact me off list. Thanks, Michael S Collins

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transfer_ringback heard by both parties (during att_xfer)
by Fraser Redmond 19 Nov '10

19 Nov '10

I've been exploring att_xfer the last couple of days and am really happy with it, works really well, and it's been interesting looking back at old mailing list emails to see how it's evolved over the years. One thing that isn't quite how I'd expected it to be is that if I set transfer_ringback=$${hold_music} both the original parties hear it, instead of just the person who is being transferred. For example (if we have lines A, B, C: A calls in to B, then B transfers to C and hangs up leaving A & C talking) then during the transfer A hears the hold_music (which is what I'd expect) but when C's phone is ringing B hears it as hold_music instead of a 'normal' ringback. Reading through the wiki, it looked like setting one of these vars before doing the transfer might work, but no change: <action application="set" data="ringback=${us-ring}"/> <action application="set" data="ignore_early_media=true"/> Anything else I should try? Or is this intended behaviour, or a bug? Cheers, Fraser

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dialplan to share skype instances
by Jian Ren 19 Nov '10

19 Nov '10

Hi, Currently I am assigning one instance to each extension like below: <extension name="us.skypeout"> <condition field="username" expression="^10(\d{2})$"> <action application="set" data="skype_instance=$1"/> </condition> <condition field="destination_number" expression="^(\d{11})$"> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="skypopen/interface${skype_instance}/$ </condition> </extension> Are there any other ways like to get unused instances from a pool? I guess it could be something like condition instance1 available? bridge instance1 condition instance2 available? bridge instance2 ... then there could be much more extensions to share skype out. Could anyone help? Thanks! Jian

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mod_dingaling for Google Voice - delay issues?
by Avi Marcus 19 Nov '10

19 Nov '10

Hi, has anyone else experienced about 2.5 second delays when using mod_dingaling for google voice? I used my gizmo callback for google voice, though the same freeswitch server to an echo test and noticed about 1/2 a second delay. (I am in Israel, that could explain it.) But when using mod_dingaling, it's 2.5 seconds, so it's not a generic google voice issue. Either it's a mod_dingaling issue, or a google jingle issue. Any ideas on how to fix this? Thanks, Avi

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sofia: call does not exist
by Christian Löschenkohl 19 Nov '10

19 Nov '10

hello the sofia stack responds to an outgoing invite with this message. the problem seems to be the following behavior from the endpoint (ip pbx) - 1st invite To: <sip:0123456789@sipserver> - server sends Proxy Authentication Required + nonce - pbx sends To: <sip:0123456789@sipserver>;tag=Sp1m3ayr748rH so the two sip dialoges do not match and the server respons with "Call Does Not Exist". is there a way to handle this (maybe broken) behavior from this sip client. possibly per user with an NDLB config option br -- Ing. Christian Löschenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl(a)xpirio.com

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Receiving calls from incoming DID's
by Neil J. Macvicar 19 Nov '10

19 Nov '10

Hello, I have recently compiled and installed freeswitch, and it is running with no problems. I can make and receive calls via my defined gateway with my provider, VoipTalk. My problem lies in that I have several DIDs with Voiptalk, and I have the option of routing them directly to my IP address as a SIP call. When I do this and call one of the numbers, not much happens. So I put sofia into debug sip mode, and observed the same message repeating :- ------------------------------------------------------------------------ send 866 bytes to udp/[217.14.138.126]:5060 at 22:22:15.236893: ------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required 217.14.138.126 being one of my providers' machines. What i'd like to know is how can I get freeswitch to stop asking for authentication to my provider, without putting a blacket "allow" in the domain ACLS. Voiptalk have a large number of IP addresses that they use to send the outbound calls, so listing them all would be a chore. Thanks in advance, --Neil.

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intercept call with pin
by budi wibowo 19 Nov '10

19 Nov '10

hi any idea to intercept outgoing call with pin? before or after press dialed digit, ivr will be played and ask for pin. i already googling but most of pin usage is for conference. thx and regards budi

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IVR menu failure/timeout action
by Raymond Chan 19 Nov '10

19 Nov '10

Hi all, I am experiencing an IVR problem. Except the top menu, all other sub menu will not hang up after reached max invalid input or timeout limit. It will go back up upper menu. I want it to load the exit-sound and then hang the call when it reached max invalid input or timeout limit. Do you tell how to configure? <menu name="m800_ivr_submenu-m_ch" greet-long="C:/FreeSWITCH/recordings/800-ivr-2nd-m.wav" greet-short="C:/FreeSWITCH/recordings/greet-short-m.wav" invalid-sound="ivr/ivr-that_was_an_invalid_entry-m.wav" exit-sound="voicemail/vm-goodbye-m.wav" timeout="5000" max-failures="2"> <entry action="menu-exec-app" digits="1" param="playback C:/FreeSWITCH/recordings/800-ivr-announcement-m_ch.wav"/> <entry action="menu-exec-app" digits="2" param="lua endcall-m.lua"/> </menu> Thanks Raymond

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error while loading mod_spidermonkey
by ovvenkat 19 Nov '10

19 Nov '10

Hi , I am getting following error while loading mod_spidermonkey freeswitch@internal> load mod_spidermonkey +OK Reloading XML -ERR [module load file routine returned an error] 2010-11-19 13:32:04.754650 [INFO] mod_pocketsphinx.c:482 PocketSphinx Reloaded freeswitch@internal> 2010-11-19 13:32:04.754650 [INFO] mod_enum.c:808 ENUM Reloaded 2010-11-19 13:32:04.754650 [INFO] switch_time.c:950 Timezone reloaded 530 definitions 2010-11-19 13:32:04.755651 [CRIT] switch_loadable_module.c:928 Error Loading module /usr/local/freeswitch/mod/mod_spidermonkey.so **/usr/lib/libnssutil3.so: undefined symbol: PR_GetDirectorySeparator** -- Regards Venkatesan OV.

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problem with voicemail
by joy this 19 Nov '10

19 Nov '10

Dear all: I am confused with voicemail for several days. I hope someone could help me please. I would like to send a voicemail with audio attachment by my application. My FS is Version 1.0.head, on Windows XP. The following are my configurations: switch.conf.xml: <param name="mailer-app" value="C:\FreeSWITCH\Mailer.exe"/> <param name="mailer-app-args" value="MyArgs -t"/> \directory\default\1001.xml <param name="vm-password" value="1001"/> <param name="vm-mailto" value="thisjoy(a)gmail.com" /> <param name="vm-email-all-messages" value="true"/> <param name="vm-attach-file" value="true" /> Other settings are as default. I register with 1000 and call 1001 (not registered) for sending a voicemail. When I left a message, my Mailer.exe only receives the arguments: -f 1001(a)192.168.1.1 MyArgs –t thisjoy(a)gmail.com. FS shows "The process tried to write to a nonexistent pipe", and the mail actually sends to me without the attachment (I have test my Mailer.exe, it could attach a file without FS). The audio file exists in "C:\FreeSWITCH\storage\voicemail\default\192.168.1.1\1001". I saw “2010-11-12 18:50:18.171875 [DEBUG] switch_utils.c:631 Emailed file [C:\DOCUME~1\123\LOCALS~1\Temp\mail.12895590120029] to [thisjoy(a)gmail.xn--com]-jb7a in the log file, but the file “mail.12895590120029” never appears. I found that the directory “C:\DOCUME~1\123\LOCALS~1\Temp” is as the sqlite temporary path in the “freeswitch\libs\sqlite\www\pragma.tcl”, although I do not know if this causes the mail failure. Could anyone teach me how to attach the audio file with the mail? Sincerely yours, thisjoy.

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