January 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

continue_on_fail does not works with limit?
by Muhammad Naseer Bhatti 29 Jan '13

29 Jan '13

I am using limit_execute to bridge a call. <action application="set" data="continue_on_fail=false"/> <action application="limit_execute" data="hash outbound first_vb03_local 1 bridge sofia/internal/19198@10.211.55.18:5080" /> <action application="limit_execute" data="hash outbound second_vb05_local 1 bridge sofia/internal/19198@10.211.55.20:5090" /> My continue_on_fail is set to false since I don't want to try the next gateway if the call is not tried because there is 1 call already first gateway. Since the call is never tried with the first gateway because of limit, there is no FAILURE from the gateway for continue_on_fail to react. Is there a way to make limit work with continue_on_fail? Or is there a better way to send multiple gateways in the dial plan where I only want to try the next gateway in the list if first gateway's limit it reached? Thanks, -- Muhammad Naseer Bhatti

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segfault in libfreeswitch.so.1.0.0
by Mimiko 29 Jan '13

29 Jan '13

Hello. Randomly I get this in kern.log: kernel: freeswitch[3136]: segfault at 8d00000081 ip 00007fbe75c90130 sp 00007fbe705c66d0 error 4 in libfreeswitch.so.1.0.0[7fbe75b68000+277000] Its so random that I can't find why. Usually this happen once a day at random time. There are days when this does not happen. echo 1 > /proc/sys/fs/suid_dumpable ulimit -c unlimited As per wiki, core files can not be located. Ulimit is enabled before starting freeswitch. But as FS is started under user from basedir directory, and that user does not have write permission to this directory, therefore no core files are created. The FS version is: FreeSWITCH Version 1.3.8b+git~20121130T003427Z~b35ac6b3bd (git b35ac6b 2012-11-30 00:34:27Z) Maybe something was changed from that version. In jira changelog I didn't see any related to this. How to track this? Is there a variable to set for core files directory dump? Thank you. -- Mimiko desu.

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[mod_sofia] Is the SIP Presence (SIMPLE) functionality limited to 2 states on FreeSWITCH?
by kkgp20＠gmail.com 29 Jan '13

29 Jan '13

Good morning, I am trying to understand and make working the SIP Presence functionality (SIMPLE) with the FreeSWITCH server. I have noticed that FreeSWITCH seems to support only two states: registered and not registered (online, offline). I have installed two SIP Clients: PortGO and Zoiper Communicator, both have to functionality to set a user status like: busy, away, in a call, on the lunch etc. By the Wireshark software I have seen that changing the user status on PortGO does not cause to send any SIP NOTIFY or PUBLISH messages - which seems to be a bug, but the Zoiper was doing it. So then I have noticed that when I have changed the user state on one of the clients, the second client all the time was receiving the NOTIFY message from FreeSWITCH with "Available" status (no matter which state I have chosen, despite offline). Only in case when I have switched to offline, the client have received "Not registered" message. My question is if it is possible to configure the FreeSWITCH server to support other states? It means any text which is going inside the <dm:note/> XML tag? Because now it seems that this tag is ignored and FreeSWITCH is sending the Available. Thank you for help.

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Direcotry Status with mod_xml_curl
by mehroz 29 Jan '13

29 Jan '13

Hi, I have been working with asterisk previously, recently came over Freeswitch platform. I dig up for the similar fucntionality of asteriks-realtime and found MOD_XML_CURL as the nearest match for dynamic directory(extension) creation. One good thing about asterisk-realtime was, it was compact in a single table and direct conversation with Asterisk and DB is possible , whereas in Freeswitch its through Webserver and curl requests. Watever, i accepted that! but one more thing i am missing is the status of SIP Registration (registered or not). Again, in asterisk realtime i can see the RTP port assigned to extension when it registers, this way i can see the current status reading the DB , and port number. Now the question is , how can i see this with mod_xml_curl ? and also, if any one can help me with the above mentioned grudges and crushes! -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Direcotry-Status-with-mod-xml… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Remove inband DTMF on record_session
by Tamas Jalsovszky 29 Jan '13

29 Jan '13

Hello, Is there a way to to remove inbad DTMF from the recorded wav file (conditionaly)? If not, how could be that done the easyiest way and a way it could get into FS git? As FS has a great DTMF detector and it has probably all the stuff doing such a feature, I think it would be much better than using an external tool for removing DTMF from finished wav file - and it would consume more I/O probably. Regards, Jalsot

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Help with originating calls
by Gary Foreman 29 Jan '13

29 Jan '13

Hi, I'm very new to Freeswitch and having a bit of difficulty with the originate command. Essentially I'm using the CLI to originate a call however the process seems to be that the originating extension is first called, then once answered the actual call takes place. My application will take care of whether the agent is available so I want to skip the first part and have the extension dial without having to be called first, this is the way the Avaya systems I have experience with work. I've looked at the sip_auto_answer option however my sip client doesn't support auto answer. I'm just not sure on the correct way of achieving what I'm looking to do, maybe some kind of bridge would be better? Thanks in advance for your help

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Re: [Freeswitch-users] Webrtc
by Sanjay Soni 29 Jan '13

29 Jan '13

Thanks. I also had this license thing in mind and didnt like this sipml business! May be i should have framed my question differently. So, what i want to know is how the webrtc solution will look like with fs ? Because frankly there is no information about it anywhere except some of the mailing list old emails which also doesnt give this information. Anthony Minessale <anthony.minessale(a)gmail.com> wrote: I would be careful with sipml5. All of the code related to it is GPLv3 which is the new version of the GPL that goes viral over a socket. So if you make a product from this code and sell it as a service, you will be bound to the GPL license and forced to surrender your platform. See the Web Services section in http://neptune.billgatliff.com/GPL3.html Also the project has changed its goals towards making a converter to change webrtc based connections into regular SIP for people who do not wish to implement the webrtc profile....... On Mon, Jan 28, 2013 at 7:57 AM, soni4m <sanjay.sanjaysoni(a)gmail.com<mailto:sanjay.sanjaysoni@gmail.com>> wrote: Hi Anthony, Can you please explain how the overall solution will look like once this webrtc support in FS effort completes ? Will it be the way asterisk is used in many of these demos (One example here <http://highsecurity.blogspot.in/2012/12/webrtc-and-asterisk-11-using-sipml5…> ) In a sense that sipml5 (SIP stack in the browser) is necessarily there on the browser side. My interest here is to understand the topology here how FS will help connecting the legacy SIP devices with browser and what will come into the picture in between. Thanks -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Webrtc-tp7580875p7586790.html Sent from the freeswitch-users mailing list archive at Nabble.com. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com<mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net<http://irc.freenode.net> #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org<mailto:sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org<mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900

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Need Help On mod_xml_curl
by johnthan123 29 Jan '13

29 Jan '13

HI All, Can You Please any one help me To Write Directory and Dialplan and In XML_CURL Like Below #################CODE######################## #!/usr/bin/perl use strict; use warnings; use Data::Dump qw(dump); use CGI; my $q = CGI->new; my $section = $q->param('section'); if ($section eq 'directory'){ $XML_STRING2 = ' <?xml version="1.0" encoding="UTF-8" standalone="no"?> <document type="freeswitch/xml"> <section name="directory"> <domain name="192.168.4.32"> <user id="1001"> <params> <param name="password" value="1001"/> </params> <variables> <variable name="accountcode" value="1001"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="1001"/> <variable name="nibble_rate" value="0.05"/> <variable name="nibble_account" value="1"/> <variable name="effective_caller_id_number" value="1001"/> </variables> </user> </domain> </section> '; } elsif ( $section eq 'dialplan' ) { $XML_STRING = ' <section name="dialplan" description="example_curl_dialplan"> <context name="public"> <extension name="curl_test"> <condition field="destination_number" expression="^(80\d{2})$"> <action application="set" data="dialed_extension=$1"/> <action application="set" data="nibble_account=${accountcode}"/> <action application="bridge" data="sofia/gateway/peer_asterisk/$1"/> </condition> </extension> </context> </section> </document> '; } ###########################END############################# This is Working for Me But i cant able to take the Database values into XML file Thanks In Advance Thanks J :: T

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Webrtc
by Tamas Jalsovszky 28 Jan '13

28 Jan '13

Hello, Is there any plan to support webrtc in FS? Right I've found this post about asterisk: http://blogs.digium.com/2012/05/23/asterisk-11-webrtc/ T.

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FreeSWITCH Weekly News and Notes
by Michael Collins 28 Jan '13

28 Jan '13

Hello all! I wanted to start off by giving you all some good news: Our friends at Newegg <http://www.newegg.com> have successfully slain a Patent Troll named Soverain. This NPE (non-producing entity) acquired some questionable patents and started suing just about anybody with an online shopping cart on their website. To make a long story short: Newegg fought them in court instead of settling and in the process got all three patents invalidated! More details available at the Ars Technica article<http://arstechnica.com/tech-policy/2013/01/how-newegg-crushed-the-shopping-…>. In any event, those of us who do software for a living, especially in the US, are well aware of the dangerous patent situation we have. Hattip to Newegg for doing us all a favor. More good news! We have released a few more ClueCon 2012 videos: * Open Source is revamping telecom. You want in. Learn from our wins and misses.<http://www.cluecon.com/presentation/open-source-revamping-telecom-you-want-…>- Darren Schreiber * HOMER in the cloud. <http://www.cluecon.com/presentation/homer-cloud/> - Alexandr Dubovikov * OSTAG - Who we are, what we do, and why.<http://www.cluecon.com/presentation/ostag-who-what-why/>- Raymond Chandler I will be uploading the slides for the talks shortly. In other ClueCon news we are gearing up for the 2013 event. We look forward to hearing from prospective speakers, sponsors, and attendees. As a reminder, we are at the same hotel <http://www.cluecon.com/hotels/> this year and the event runs from August 6th through the 8th. Stay tuned for more information. We have an annual tradition that is coming up next week: buy the developers dinner! Quite simply, we invite the community to throw some money into the hat and then the developers use it to buy dinner. The easiest way to donate is to click the Donate button on the main FreeSWITCH website<http://www.freeswitch.org>. (Right-hand side, under the ClueCon 2013 logo.) Do a Paypal transaction, mark it person (a gift is fine) and in the message to the recipient mention that the money is for the developer dinner. We appreciate the community doing this again for us this year. Thank you! We'll talk to you this Wednesday on our weekly conference call<http://wiki.freeswitch.org/wiki/FS_weekly_2013_01_30>. Last week was an open discussion<http://wiki.freeswitch.org/wiki/FS_weekly_2013_01_23>. This week I would will be doing an event socket and ESL review. More people are using these great tools and we are seeing some of the same questions being asked so we want to make sure that we get this information out there for everyone to use. Thanks again for being a great community and we will talk to you again next week. -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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