September 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Fwd: [asterisk-users] VoIP Abuse to Twitter (real time VoIP Abuse)
by vip killa 22 Sep '11

22 Sep '11

---------- Forwarded message ---------- From: Robert Huddleston <rhuddleston(a)gmail.com> Date: Thu, Sep 22, 2011 at 11:10 AM Subject: Re: [asterisk-users] VoIP Abuse to Twitter (real time VoIP Abuse) To: Asterisk Users Mailing List - Non-Commercial Discussion < asterisk-users(a)lists.digium.com> Sounds like a great idea.. Hopefully the page/account never gets hacked and bad IP’s published.. I could see a great hack of **** 127.0.0.1 **** 192.168.0.0/16 **** 10.0.0.0/8 **** getting up there somehow and next thing you know – BAM!**** ** ** But I haven’t RTFM – I’m guessing there is probably a white list that supersedes the naughty list.**** ** ** ** ** *From:* asterisk-users-bounces(a)lists.digium.com [mailto: asterisk-users-bounces(a)lists.digium.com] *On Behalf Of *vip killa *Sent:* Thursday, September 22, 2011 11:06 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] VoIP Abuse to Twitter (real time VoIP Abuse) **** ** ** very cool!**** On Thu, Sep 22, 2011 at 10:37 AM, J. Oquendo <asterisk(a)tormenting.net> wrote:**** Apologies for cross posting but some of us aren't on the other list (vice/versa) and thought both groups would benefit. For those familiar with the VoIP Abuse Project, no need to explain the gist of this. I got tired of parsing through the alerts (lists) I receive via email daily. They're long and sometimes I don't have the time to post them all. So for now, posting VoIP Abuse addresses straight to Twitter. So, anyone trying to compromise a pbx, is now autoposted on an hourly basis to Twitter. Still working on pulling, have about 4 machines linked up now, will mop em up during the week. http://twitter.com/#!/voipabuse Now, you can concoct a quick script off of it, e.g.: links -dump "http://twitter.com/voipabuse"|awk '/attacker/{print "iptables -A INPUT -s "$2" -j DROP"| "sort -u"}' Will get a quickie soon from my Acme's, nCites, etc. when I have time. For those NOT familiar with it, please Google it as I don't feel like typing anymore ;) (sorry) -- =+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ J. Oquendo SGFA, SGFE, C|EH, CNDA, CHFI, OSCP, CPT, RWSP, GREM "It takes 20 years to build a reputation and five minutes to ruin it. If you think about that, you'll do things differently." - Warren Buffett 42B0 5A53 6505 6638 44BB 3943 2BF7 D83F 210A 95AF http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x2BF7D83F210A95AF -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users**** ** ** -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

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Multiple apps with Execute_on_answer
by Chris Martineau 22 Sep '11

22 Sep '11

Hi, How do you execute multiple applications on answer? I need to execute a sched_hangup and a sched_broadcast on answer so that the timers start from answer. However cannot find any combination which allows me to put multiple apps in the same execute on answer action. If I just create multiple action entries it only seems to action the last one entered? So have... application='set' data='execute_on_answer=sched_hangup +XX allotted_timeout' if I add... application='set' data='execute_on_answer=sched_broadcast +XX /path/file' only the broadcast triggers? Many thanks again Chris

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Terminate early media playback on answer
by Chris Martineau 22 Sep '11

22 Sep '11

Hi, I am using a playback on early media to provide a welcome message and would like to terminate the playback on answer. I have tried application='set' data='execute_on_answer=break all' but this doesn't seem to do anything even though the log shows it being executed on answer. Currently if the far end answers quickly they get silence until the playback has finished and then they get through. Any ideas on how I can do this? Many thanks again Chris

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how to redirect $${sounds_dir} ?
by jesse 22 Sep '11

22 Sep '11

I would like to pint sounds_dir to a different location, but it seems freeswitch doesn't support --sounds_dir as command line arguments. please let me know how to re-define sounds_dir? thanks!

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FreeTDM kickstart help ERRORS inline
by Luis F Urrea 21 Sep '11

21 Sep '11

This dAHDI stuff is like voodoo! I know this has to be configuration issues but sh****t after a few years in the telecom industry it is really hard to find a more awful syntax for configuring such a stupid device as an FXO port. I am trying to get a Xorcom Astribank going with FreeTDM. So far after a bunch of firmware and driver BS (come on there is probably nothing more buggy in a freking kernel than the USB layer, how can a telecom device be using a USB driver!!!!!!!). I get the following output from #lsdahdi ### Span 1: XBUS-00/XPD-00 "Xorcom XPD #00/00: FXO" (MASTER) 1 FXO RED 2 FXO RED 3 FXO RED 4 FXO RED 5 FXO RED 6 FXO RED 7 FXO RED 8 FXO RED ### Span 2: XBUS-00/XPD-10 "Xorcom XPD #00/10: FXO" 9 FXO RED 10 FXO RED 11 FXO RED 12 FXO RED 13 FXO RED 14 FXO RED 15 FXO RED 16 FXO (battery) ### Span 3: XBUS-00/XPD-20 "Xorcom XPD #00/20: FXO" 17 FXO RED 18 FXO (battery) 19 FXO (battery) 20 FXO (battery) 21 FXO (battery) 22 FXO (battery) 23 FXO (battery) 24 FXO (battery) But when loading mod_freetdm I am getting: 2011-09-19 15:44:03.149105 [NOTICE] ftdm_io.c:5723 Modules configured: 1 2011-09-19 15:44:03.149105 [DEBUG] ftdm_config.c:80 Configuration file is /usr/local/freeswitch/conf/freetdm.conf. 2011-09-19 15:44:03.149105 [DEBUG] ftdm_io.c:4633 Reading FreeTDM configuration file 2011-09-19 15:44:03.149105 [DEBUG] ftdm_io.c:4649 found config for span 2011-09-19 15:44:03.149105 [NOTICE] ftmod_zt.c:1323 Using DAHDI control device 2011-09-19 15:44:03.149105 [INFO] ftdm_io.c:4963 Loading IO from /usr/local/freeswitch/mod/ftmod_zt.so [zt] 2011-09-19 15:44:03.149105 [DEBUG] ftdm_config.c:80 Configuration file is /usr/local/freeswitch/conf/zt.conf. 2011-09-19 15:44:03.149105 [INFO] ftmod_zt.c:585 Setting rxgain val to 0.000000 2011-09-19 15:44:03.149105 [INFO] ftmod_zt.c:593 Setting txgain val to 0.000000 2011-09-19 15:44:03.149105 [INFO] ftdm_io.c:787 Auto-loaded I/O module 'zt' 2011-09-19 15:44:03.149105 [DEBUG] ftdm_io.c:4670 created span 1 (SPAN1) of type zt 2011-09-19 15:44:03.149105 [DEBUG] ftdm_io.c:4688 span 1 [fxo-channel]=[17-24] 2011-09-19 15:44:03.149105 [DEBUG] ftdm_io.c:4718 setting trunk type to 'FXO' start(KEWL) 2011-09-19 15:44:03.149105 [ERR] ftmod_zt.c:288 failure configuring device /dev/dahdi/channel chan 17 fd 48 (No such device or address) 2011-09-19 15:44:03.149105 [ERR] ftmod_zt.c:288 failure configuring device /dev/dahdi/channel chan 18 fd 48 (No such device or address) 2011-09-19 15:44:03.149105 [ERR] ftmod_zt.c:288 failure configuring device /dev/dahdi/channel chan 19 fd 48 (No such device or address) 2011-09-19 15:44:03.149105 [ERR] ftmod_zt.c:288 failure configuring device /dev/dahdi/channel chan 20 fd 48 (No such device or address) 2011-09-19 15:44:03.149105 [ERR] ftmod_zt.c:288 failure configuring device /dev/dahdi/channel chan 21 fd 48 (No such device or address) 2011-09-19 15:44:03.149105 [ERR] ftmod_zt.c:288 failure configuring device /dev/dahdi/channel chan 22 fd 48 (No such device or address) 2011-09-19 15:44:03.149105 [ERR] ftmod_zt.c:288 failure configuring device /dev/dahdi/channel chan 23 fd 48 (No such device or address) 2011-09-19 15:44:03.149105 [ERR] ftmod_zt.c:288 failure configuring device /dev/dahdi/channel chan 24 fd 48 (No such device or address) 2011-09-19 15:44:03.149105 [ERR] ftdm_io.c:4579 1:Failed to configure span 2011-09-19 15:44:03.149105 [INFO] ftdm_io.c:4885 Configured 0 channel(s) 2011-09-19 15:44:03.149105 [ERR] ftdm_io.c:5733 FreeTDM global configuration failed! 2011-09-19 15:44:03.149105 [ERR] mod_freetdm.c:4805 Error configuring FreeTDM 2011-09-19 15:44:03.149105 [INFO] switch_time.c:1028 Timezone reloaded 530 definitions 2011-09-19 15:44:03.149105 [CRIT] switch_loadable_module.c:1019 Error Loading module /usr/local/freeswitch/mod/mod_freetdm.so **Module load routine returned an error** 2011-09-19 15:44:03.149105 [NOTICE] ftdm_sched.c:147 Main scheduling thread going out ... Thanks in advance and sorry for the rant.

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vBilling Beta Program!!
by Muhammad Naseer Bhatti 21 Sep '11

21 Sep '11

Hello everyone, As promised we have opened beta testing program for *vBilling*. An open source billing platform for FreeSWITCH. You are invited to login, take a look and play with it. Email us with your comments, let us know what improvements can be made. Following are the details for the program: ======================================== vBilling User Panel: *http://demo.vbilling.org/* User Login: *demouser* User Password: *P@ssw0rd* vBilling Admin Panel: *http://demo.vbilling.org/admin/* Admin Login: *admin* Admin Password: *P@ssw0rd* We have configured a LIVE gateway for you. You can send your *SIP* calls to *demo.digitallinx.com* port *5060* and call any toll free (800 and 888 only) number in US. Make calls, login to *vBilling* and see how the billing works. ======================================== Most of the features are working. Some of the them mentioned on the site are still is development so be patient :) For product features and more details, visit our website at *http://www.vbilling.org/* <http://www.vbilling.org/>and us know what do you think about it. Regards, Muhammad Naseer CEO vBilling/Digital Linx

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change sip_to_user in dialplan
by Weigel, Stefan 21 Sep '11

21 Sep '11

Hello all, I'm trying to change some incoming calls 'sip_to_uri' (it's a rerouting of our ITSP) so it matches our main number: 0891234567 -> 08998765 I'm checking the sip_to_uri and then trying to export/set/set_profile_var to the new number. But in debug trace it's still comparing against the original sip_to_uri. Is there a way to change as soon it's hitting the dialplan ? Thanks and best regards Stefan

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Inaccurate variable_record_ms
by Adam Kelloway 21 Sep '11

21 Sep '11

Hi there, I've noticed that the variable_record_ms and variable_record_seconds values are often inaccurate. Is there any explanation for this? The example below shows that while the RECORD_START and RECORD_STOP events were about 4 seconds apart, the variables claim that the file is 7 seconds long. See the soxi output at the end, which shows that the length of the file is 00:00:03.50, thus more in line with what the event timestamps suggest. RECV EVENT Event-Name: RECORD_START Core-UUID: 235bab4c-36c6-48af-87d7-0f30a1140d52 *Event-Date-GMT: Wed, 21 Sep 2011 14:52:24 GMT* Event-Date-Timestamp: 1316616744483384 Event-Calling-File: switch_ivr_play_say.c Event-Calling-Function: switch_ivr_record_file Event-Calling-Line-Number: 615 Unique-ID: fa39d249-c9da-4072-aebe-4b606e661e5f Caller-Profile-Created-Time: 1316616744483384 Caller-Channel-Created-Time: 1316616740683382 Caller-Channel-Answered-Time: 1316616740683382 *variable_current_application_data: /tmp/fa39d249-c9da-4072-aebe-4b606e661e5f.wav 120 200 2* variable_current_application: record Record-File-Path: /tmp/fa39d249-c9da-4072-aebe-4b606e661e5f.wav RECV EVENT Event-Name: RECORD_STOP Core-UUID: 235bab4c-36c6-48af-87d7-0f30a1140d52 *Event-Date-GMT: Wed, 21 Sep 2011 14:52:28 GMT* Event-Date-Timestamp: 1316616748083413 Event-Calling-File: switch_ivr_play_say.c Event-Calling-Function: switch_ivr_record_file Event-Calling-Line-Number: 790 Unique-ID: fa39d249-c9da-4072-aebe-4b606e661e5f Caller-Profile-Created-Time: 1316616744483384 Caller-Channel-Created-Time: 1316616740683382 Caller-Channel-Answered-Time: 1316616740683382 *variable_current_application_data: /tmp/fa39d249-c9da-4072-aebe-4b606e661e5f.wav 120 200 2* variable_current_application: record *variable_record_seconds: 7* *variable_record_ms: 7000* variable_record_samples: 56000 Record-File-Path: /tmp/fa39d249-c9da-4072-aebe-4b606e661e5f.wav root@dev:/usr/local/freeswitch# soxi /tmp/fa39d249-c9da-4072-aebe-4b606e661e5f.wav Input File : '/tmp/fa39d249-c9da-4072-aebe-4b606e661e5f.wav' Channels : 1 Sample Rate : 8000 Precision : 16-bit Duration : *00:00:03.50* = 28000 samples ~ 262.5 CDDA sectors File Size : 56.0k Bit Rate : 128k Sample Encoding: 16-bit Signed Integer PCM

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Missing mod_blacklist.conf file?
by mazilo 21 Sep '11

21 Sep '11

I took a look at and studied the applications/mod_blacklist.c file which was recently committed to FS git source. I found a reference to access mod_blacklist.conf file on line #116 of the applications/mod_blacklist.c file; however, I searched through my local FS source directories and could not find the mod_blacklist.conf file. So, anyone here who has used mod_blacklist can point to me where I can get a copy of mod_blacklist.conf file? ----- FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Missing-mod-blacklist-conf-fi… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Goto Voicemail When Network Down
by Dan Lan 21 Sep '11

21 Sep '11

Hi, I install the Freeswitch without much modification. I have an extension, and with incoming DID transfer to that extension. Some Test Scenarios 1. When the extension is registered, the incoming call will go to VM after 30 secs, if no one answer the call (This is normal) 2. When the extension is not registered, the incoming call will go directly to VM 3. However, if the endpoint is registered, but for some reason, the endpoint's network is down. The incoming call will just go to blank scilience. after about 60 secs then I can hear the busy signal. >From the packet capture, I can see FS try to reach the end point by sending INVITE during the 60 secs, but because the end point is not responding, so the FS keep sending INVITE. I am just wondering, are there anyway to break the loop? see, if the FS send INVITE without response for some certain time, then go to Voice Mail without the caller waiting and dont know what to do? Dan

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