September 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Busy week for the FreeSWITCH Crew...
by Ken Rice 21 Sep '12

21 Sep '12

Hey Guys, The FreeSWITCH Crew has had a very busy week. Things from this week. Security issue in libsofia fixed. The problem was a specifically crafted Route header on an invite could cause a segfault leading to a denial of service situation. For those of you that have your FreeSWITCH boxes exposed to the internet, its a good idea to update at this time. This has been resolved in both the master development branch and the stable branch Along with resolving this issue and updating the stable branch, we have released FreeSWITCH 1.2.3. Source is available in all the normal locations. Have a great weekend guys! -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

1 0

bridge_terminate_key alternatives?
by Emrah 21 Sep '12

21 Sep '12

Hello listers, bridge_terminate_key is good for what I am trying to do, except the limitation to only 1 key. Is there a way so to be able to terminate a bridge session with a DTMF sequence? E.g.: *#. As a fallback solution, I think I could put in a feature extension that kills the UUID of the leg I need to hangup. Is that the route you would recommend? Cheers, Emrah

2 3

sofia recover does not recover calls on hold
by Vik Killa 21 Sep '12

21 Sep '12

Hi, I've been testing 'sofia recover' the past few days using two freeswitch servers and a floating IP. It works great in general, but I did notice a problem with any call that is holding during the crash and recovery. Basically, any call that is on hold comes back after 'sofia recover' but when trying to take the call off hold the call is dropped. I'm using the following parameters on both FS instances in switch.conf.xml: <param name="switchname" value="fs01"/> <param name="core-db-dsn" value="dsn:name:password"/> <param name="core-recovery-db-dsn" value="dsn:name:password"/> And in sofia config: <param name="track-calls" value="true"/> <param name="odbc-dsn" value="dsn:name:password"/> As I said before, sofia recovery works except for the calls holding...just wondering if this is a known issue before I open up a jira ticket. Thanks! -Vik

2 1

recording's channels shifted
by Tóth Róbert 21 Sep '12

21 Sep '12

No, the callers do not notice anything, they hear each other without problem. One party is on the local network where FS is, and other is outside. But as they do not notice this problem, i think the problem is at the recording. I think. >/ /This is more likely an issue with the media streams than the recording >/ /itself. Do the callers notice anything? Is one or both callers outside the >/ /local network where the FS server reside? Just looking for places where >/ /interesting/weird things might happen. >/ / >/ /-MC >/ / >/ /On Thu, Sep 20, 2012 at 6:40 AM, Tóth Róbert <trob at freemail.hu <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>> wrote: >/ / >/ />/ Hi />/ />/ />/ />/ Why left and right channels' voice shifted from each other in the recorded />/ />/ mp3? />/ />/ />/ />/ I record the sessions' voice thios way: /> >/ <action application="set" data="execute_on_answer=record_session /> >/ $${base_dir}/Record/${strftime(%Y%m%d-%H%M%S)}.mp3"/> /> >/ When i listen back the mp3, at begining the two chennels are synchonized, /> >/ but after some minutes there will be a shifting between the channels. /> >/ /> >/ Why? /> >/ /> >/ thanks, /> >/ Robert/

3 3

Segmentation Fault while running configure
by Muhammad Shahzad 21 Sep '12

21 Sep '12

Hi, I am running Ubuntu 12.04 64-bit in KVM and getting segmentation fault while running configure script of latest stable Freeswitch v1.2. Here are lines just before crash. /configure --enable-64 --enable-core-odbc-support --enable-optimization --with-pic ... checking whether getpwnam_r and getpwuid_r are posix like... yes checking whether getpwnam_r and getpwuid_r are posix _draft_ like... no configure: creating ./config.status ./configure.gnu: line 3: 20996 Segmentation fault (core dumped) $srcpath/configure "$@" --disable-shared --with-pic configure: error: ./configure.gnu failed for libs/libedit Any idea what's wrong here. I did tried to run configure without any parameters but same result. Thank you. -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk(a)hotmail.com Email: shaheryarkh(a)googlemail.com

4 5

Speak vs. StreamFile
by Malay Thakershi 21 Sep '12

21 Sep '12

Hello, I am working on optimizing call flow code. I use Cepstral as TTS from FS managed module (C# .net code). I suspect using Session.Speak is slowing down the call experience (prompt delay) when there are # of concurrent calls. My question: does it help with performance if I create WAV file for each static prompt in advance and use Session.StreamFile instead of Session.Speak? Any help is appreciated. Malay

5 5

Asking a Question (a.k.a. Gabe must be getting old)
by Gabriel Gunderson 21 Sep '12

21 Sep '12

This should be required reading before posting questions to the list: http://www.catb.org/esr/faqs/smart-questions.html#intro Special thanks to those people on the list who take special care to ask *smart* questions and those who provide valuable answers. You know who you are ;) Gabe

7 13

UK English Prompt Set Recording
by Campbell Steven 20 Sep '12

20 Sep '12

Hi en_UK FreeSWITCH users, I'm trying to get a full prompt set (every prompt currently recorded by Callie including the ztrp prompts etc..) organised. The talent would be female and prompts would be recorded through GM Voices and therefore easy to get matching custom recordings for your applications into the future. The prompt set would then be given to the FreeSWITCH project for distribution as per the existing prompts. Anyone who is willing to help me out with funding this or has any questions please contact me off-list as I'd really like to get this sorted. Every little bit counts here so even if you just want to drop a few $ in it all adds up. Thanks Campbell

6 10

Convert to e.164
by ljjimenez＠gmail.com 20 Sep '12

20 Sep '12

Hello list, I have a provider who needs to receive the call prefixed with a "+" in the "From" and "To" headers, is there any dial plan function for this or how is the best practice to modify the "From" and "To" headers to prefix the + ? Thanks in advance Luis Jiménez UDR

1 0

extra headers to event message to locate user directory for voicemail
by Daniel-Constantin Mierla 20 Sep '12

20 Sep '12

Hello, I was trying to build user directory file dynamically via a lua script at a dial-in event to listen to/leave a voice message, looking to pass some values from particular SIP headers. But the session object seems to be unavailable at that moment - next are the relevant config snippets, the dialplan: <extension name="vbox"> <condition field="destination_number" expression="^vb-(.+)$"> <action application="answer"/> <action application="voicemail" data="default ${domain_name} $1"/> </condition> </extension> and lua.conf: <param name="xml-handler-script" value="/usr/local/freeswitch/scripts/dp.lua"/> <param name="xml-handler-bindings" value="directory"/> The event message is like the one from: http://wiki.freeswitch.org/wiki/Mod_lua/Serving_Configuration#when_being_ca… The Core-UUID is different that channel UUID, so I could not find any API commands that could be used to get channel variables. Do I miss any setting/command that should get me access to channel variables or SIP headers inside the dp.lua? My quick workaround was to patch to be able to add extra headers to the internal event message (most of the time diving in the sources is the fast way to get things done :-) ). I am attaching the patch for the case there is no other solution and someone else is interested - by now it adds extra headers only when checking voicemail or recording a new message, but should be straightforward to add to other places across voicemail module if it has to (I can make the patch if one points other types of voicemail events that can benefit of extra headers). The patch adds a new channel variable, locate_event_extra_headers, that can be set to a prefix of the channel variable names that have to be passed as extra headers in the internal event message fired to locate user directory. An example is in the patch comments. If there is no solution to pass channel variables/SIP headers to the lua scripts used for generating user directories, would it be a good feature to add to devel branch? My particular need comes from using Kamailio as the SIP server and when user is offline, the call is directed to FreeSwitch for voicemail service. Also users can dial in to listed to messages. I want to pass voicemail box id, pincode and email via SIP headers, as I have them already loaded in Kamailio config script, avoiding another database lookup from freeswitch lua script. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

1 0