- FreeSWITCH-users - FreeSWITCH Mailing Lists

Vulnerability Testing
by Jerry Allen 04 Nov '14

04 Nov '14

I believe I have my system "ready for use" Are there any exploit testing tools that test for the ability to make calls etc. off of a system? I have tested with sipvicious and sipsak but I am not positive that I am testing correctly. I would like a real hack test. The system has been "wide open" for a few weeks, fine-tuned fail2ban as well as the addition of a firewall that protects my FS machine (hopefully) and there have been no outbound calls made that I did not make there have been hundreds of various hackers/scanners that fail2ban blocked etc. I am not exactly sure where to go to make sure it is not a big bomb waiting to be set off. Jerry Allen

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XML DialPlan help
by Sean Devoy 04 Nov '14

04 Nov '14

I swear I have read this, but now that I need it I can't find it. My user wants to dial international. I want to require them to do SOME extra step other than dialing 0113010... to call the Netherlands. Oddly enough, it is cheaper than US termination - go figure. Ideally, I would like them dial some extension like 777, hear a new dialtone and then accept the international number. I might even like to play a message that says "All charges for International are their responsibility and not part of their current package" before the dialtone. Ideas? Thanks again, Sean

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Freeswitch Week in Revew (Master Branch) September 28th-October 5th
by Ken Rice 04 Nov '14

04 Nov '14

New Post on freeswitch.org from kathleen check it out at http://ift.tt/1E2VPVl Freeswitch Week in Revew (Master Branch) September 28th-October 5th Hello, again. This week in the FreeSWITCH master branch we had 48 commits. We had a few new features added, like, tab completion for api commands in mod_gsmopen, a new feature to filter the SDP on bypass_media calls to remove or limit codecs, and a way to globally disable system commands by setting global var disable_system_api_commands=true. The following bugs were squashed: d619017 FS-6757 FS-6713 FS-6868 FS-6863 FS-6858 5 bugs one typo. From commit 1b612fecb6e8db11da9b15c5522b87e7b642423d [Jira: http://ift.tt/1uqVaxG;] da51603 FS-6757 FS-6713 FS-6868 FS-6863 FS-6858 #resolve #comment 5 bugs one typo. From commit 1b612fecb6e8db11da9b15c5522b87e7b642423d [Jira: http://ift.tt/1sh2sks] fbe857e FS-6644 Fix ptime from known broken endpoints on re-invite [Jira: http://ift.tt/1E2VGkM] 0150c86 FS-6854 Patch for SIP lockout on WebRTC in mod_verto [Jira: http://ift.tt/1uqV8WN] 644b41f FS-6874 Fix for multipart invite format in mod_sofia [Jira: http://ift.tt/1uqV9d2] 35aeae0 FS-6822 #comment The code in question appears to have been added by me (18f20e24). I think this patch is the correct solution. [Jira: http://ift.tt/1uqV9d6] 8db31f9 Fix some recovery issues with dynamic payloads ae5d865 FS-6884 #comment Fix for simple warnings in mod_sofia and mod_conference [Jira: http://ift.tt/1E2VGBa] 01bf422 FS-6888 Fix regression from refactoring new feature [Jira: http://ift.tt/1uqVaxL] bde2e2d FS-6889 Fix for Reinvite Codec Error when using bypass_media_resume_on_hold and bypass_media_after_hold in mod_sofia [Jira: http://ift.tt/1E2VIsW] b2ae5f4 Fixed a few bugs in the new features in mod_sofia afd6875 FS-6781 Adding a check to ensure consistency in mod_valet_parking [Jira: http://ift.tt/1uqVaxP] New features that were added: a94fbe8 Add tab completion for api commands in mod_gsmopen 92a66fb Improve adaptive jitter buffer ascending check in switch_stfu 24084ad Add new feature to filter the SDP on bypass_media calls to remove or limit codecs. 8e408e9 FS-6865 #resolve add XMPP priority to dingaling [Jira: http://ift.tt/1E2VIJi] 789e148 FS-6880 Improvements to filtering invites in the core. [Jira: http://ift.tt/1uqVaxR] b3d7191 FS-6870 vs2010 and vs2012 fix [Jira: http://ift.tt/1E2VGBl] 91ffe17 Use OPUS_APPLICATION_VOIP always to get FEC and filtering in mod_opus 9e91753 FS-6886 Addition rtp_disable_hold when bypass_media is active and addition of ignoring unhold in mod_sofia [Jira: http://ift.tt/1uqVaxT] 6bfc05b FS-6887 Add always_auto_adjust parameter [Jira: http://ift.tt/1E2VGRD] 43733a6 FS-6886 #comment addition of ignoring unhold as well to mod_sofia [Jira: http://ift.tt/1uqVaxT] 10a3fa5 Add bypass_media_resume_on_hold and bypass_media_after_hold variables to be set to true to enable these functions on a per channel basis 0d1f5d0 Add way to globally disable system commands by setting global var disable_system_api_commands=true Improvements in cross platform build supports: 812f911 Fix reference to libressl in makefile on OpenBSD a39db86 FS-6870 #vs2012 and vs2010 make download of openssl dependent [Jira: http://ift.tt/1E2VGBl] d17f14e Make sure to pass along appropriate configure flags to sub-configure’s when cross compiling d52cb33 Fix trivial vs2010 build errors The complete list of commits can be found here: 2014_9_28-2014_10_5

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mod_verto unwanted profile shutdown
by Balázs Sándor 04 Nov '14

04 Nov '14

Hi all. I'm testing mod_verto. After a while (5-10 minutes) it always disconnecting. I do nothing with FS (its sleeping) It seems to be normal shutdown 2014-11-03 08:50:30.495346 [INFO] mod_verto.c:4130 Secure key and cert specified 2014-11-03 08:50:30.495370 [INFO] mod_verto.c:4218 mine Bound to 0.0.0.0:8081 2014-11-03 08:50:30.495377 [INFO] mod_verto.c:4218 mine Bound to 0.0.0.0:8082 -- 2014-11-03 09:00:29.211502 [INFO] mod_verto.c:4524 profile mine shutdown, Waiting for 0 threads 2014-11-03 09:00:29.211502 [INFO] mod_verto.c:4535 mine Thread ending I attached a detailed log Best regards, Balázs SÁNDOR Software Developer Virtual Call Center MUNICH <http://virtual-call-center-software.de/> | BUDAPEST <http://virtual-call-center-software.hu/> | WARSAW <http://virtual-call-center-software.pl/> Phone: +44 (0) 863 801 69 Web: www.virtual-call-center.eu <https://www.linkedin.com/profile/view?id=215233036>

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Embedded Freeswitch
by Guillermo Ruiz Camauer 04 Nov '14

04 Nov '14

Can anyone point me to any examples / source code of embedde Freeswitch? I have looked at the documentation, and the different books, but can find now examples of this. Guillermo -- Guillermo Ruiz Camauer

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To decypt webrtc audio
by Varghese Paul 04 Nov '14

04 Nov '14

Hi all, We are using freeswitch for handling webrtc calls and the negotiation in DTLS mode. We are looking for a tool to decrypt the traffic between the webrtc client and freeswitch server. I know there is already a tool pcaputils from PJSIP for decrypting the SRTP traffic. Since we are using in DTLS mode we can't use this. Can any one suggest any tool/method to decrypt the webrtc audio? Regards Varghese Paul

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Good News Everyone! We can finally use HTML5, Oh yeah we already were!
by Ken Rice 04 Nov '14

04 Nov '14

New Post on freeswitch.org from anthm check it out at http://ift.tt/1xVsqdv Good News Everyone! We can finally use HTML5, Oh yeah we already were! After almost 15 years, we can finally get started on HTML6! Maybe it’ll be done in time for FreeSWITCH 3.0 http://ift.tt/1zGJkBV

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Software/Hardware interoperability
by Mario Gasparoni Jr 03 Nov '14

03 Nov '14

Hello guys, I am trying to use FreeSWITCH as a conference server for group calls. I'm using mod_conference, and I know it works fine with multiple SIP clients (for testing purposes I use Ekiga and Jitsi on both Windows and Linux). Audio is mixed individually for each user, and video is switched based on the active talker, and the same video is sent to all clients. The problem shows up when I join the group call using a Polycom hardware. If I configure FreeSWITCH to support H.263 only, it works fine, my Polycom can display the video sent by FreeSWITCH, including when the active talker is itself. The other SIP clients work fine as well. When I configure FreeSWITCH to support H.264 only, the video switching only works when Polycom is the active talker, otherwise all the users just see the last video frame sent by Polycom. If I do a peer-to-peer call between the Polycom and a SIP client, both configured to use H.264, everything works just fine, which points me that the problem could be related to the transition between viewing a video stream with a given resolution and viewing another video stream with a different resolution, or something related to the non-continuity of video packets timestamp. Is there anybody out there that have experienced similar issues? Do you guys have any suggestion or anything that could help me to make it work properly? Thanks in advance. -- Att. Mário Gasparoni. www.inf.ufrgs.br/~mcgjunior

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[Lua Script] Conditional Remapping of Hangup Cause
by Massimo Varriale 03 Nov '14

03 Nov '14

Hi, I'm creating a Lua script for managing different call scenario. At the moment I'm stuck with a problem regarding the interoperability between Freeswitch and some potential customer. The problem regards the hangup cause that Freeswitch is sending to their softswitch, to allow a their automatic rerouting based on some Hangup Cause. I got a request to send different Hangup Cause, so the Customer switch can automatically reroute call to another carrier. Here is the explaination The scenario is: All the endpoints are using a Public IP and no NAT is involved in these steps. CUSTOMER ------------------> MY_FREESWITCH -----------------> REMOTE_CARRIER What happen now: The customer can send calls. On My freeswitch I will route calls based on Dialled Prefix to a Remote Carrier. In my Lua script there is a conditional IF to check for remote disconnect cause, however it seems that the Release Cause is already sent (CALL_REJECTED) when I'm checking with my IF and so, even if the cause has been already sent it's sent again but what "win" is the first hangup sent. What should be: If the remote carrier will send me a CALL_REJECTED (isdn 21) I need to send to the remote customer another Hangup Cause for example: NORMAL_CIRCUIT_CONGESTION (isdn 34). How to implement that? I can create this remapping using a Diaplan XML configuration, but as I'm using a Lua script to manage the call, I found that is a Lua task to deal with the Hangup Cause, the dialplan rules are ignored. The "telephony" part of the Lua script is working, what I'm trying to change is the release cause to make my customer happy as with a different release cause he can route traffic correctly and I will not loose my reputation :) Below there is my Lua script (just the routing part) Thank you so much! Cheers Max > function string.starts(String,Start) > return string.sub(String,1,string.len(Start))==Start > end > > session:execute("set","hangup_after_bridge=true") > called_number = session:getVariable("destination_number") > > > while session:answered()==false and session:ready()==true do > > if string.starts(called_number, "393") then > -- ITALY MOBILE > route_dial_string = "sofia/gateway/REMOTE_CARRIER/" ..called_number > else > -- OTHER DESTINATIONS > route_dial_string = "sofia/gateway/OTHER_CARRIER/" ..called_number > end > > session:execute("bridge",""..route_dial_string.."") > > > -- ATTEMPT TO REMAP RELEASE CAUSE > if ( session:getVariable("last_bridge_hangup_cause") == "CALL_REJECTED" ) then > session:hangup("NORMAL_CIRCUIT_CONGESTION"); > end > > > if session:ready()==false then > session:hangup(); > > > end > > > session:hangup(); > freeswitch.consoleLog("info", "That's all folks!");

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Outbound Caller ID in a multi-Department environment
by Joel White 03 Nov '14

03 Nov '14

I am having an issue setting up CID for individual extensions I am running out of a PostgreSQL DB and have created a Users Table Using Lua I am dynamically generating the Directory I have a column in the table for CID, but have been unsuccessful in implementing CID for outbound calling I want the extension to display internally, but to add CID when calling outside of the system How can I implement this? And can it be done using Lua to dynamically lookup in the DB for each users CID?

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