- FreeSWITCH-dev - FreeSWITCH Mailing Lists

FreeSWITCH Weekly News and Notes
by Michael Collins 04 Jun '13

04 Jun '13

Greetings all! We want to let everyone know that there will be a special announcement on Wednesday, June 19 during our weekly conference call<http://wiki.freeswitch.org/wiki/Weekly_Conference_Call>. This announcement is very important which is why we are giving everyone plenty of notice. We look forward to seeing everyone there. In the meantime we have will have a community discussion<http://wiki.freeswitch.org/wiki/FS_weekly_2013_06_05>this coming Wednesday. On our previous call <http://wiki.freeswitch.org/wiki/FS_weekly_2013_05_29> we had Martin from the VoIPMonitor.org project. We had a nice discussion about what VoIPMonitor.org can do as well as the fantastic web interface that is available commercially. If you have an idea for a FreeSWITCH conference call presentation please let us know. We are always looking for new topics and new presenters. Have a great week! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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mod_event_socket
by Georgi Stefanov 03 Jun '13

03 Jun '13

I found it. It totally my mistake. Sorry about the fake alert. I was not parsing the data right way and my code blocked it. Once again I am sorry about raising fake alert. Now I will continue having good day with this module :-) Thank you guys I owe you beers

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Freeswitch SBC register account
by truong ngoc THANH 03 Jun '13

03 Jun '13

Hi all, I have one idea in Freeswitch SBC as: + I used Freeswitch as register proxy, in Freeswitch, it forwarding register message to kamailio and receive authen status. + In this case, FS action as a UA, and kamailio as a registra server. If user authenticated, FS store user infor in internal database, if not, it pass authen status to real UA. + i have research alot and do not know how to do this. Hope every one can help me to do this, i am new in FS, and new in FS module too. Thank all in advance ThanhT

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Hey Guys Today 3PM EST Friday FreeForAll
by Ken Rice 31 May '13

31 May '13

Hey Guys don¹t forget to join us at 3PM EST (that¹s like NOON PST) for the FridayFreeForAll For those of you Voice Only, sip:888@conference.freeswitch.org as usual For those that want to join with Video, sip:888@mcu.freeswitch.org For those that have the option, try the video Bridge lets see if we can find a breaking point with load! K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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mod_event_socket
by Georgi Stefanov 30 May '13

30 May '13

Hi All there :-) Recently I have played with mod_event_socket It works great and makes me happy, but recent days I have noticed FS send to my application CHANNEL_ANSWER after I hangup the call - something like it is blocked and after I hangup it is unblocked. It doesn't do it every time - in fact it is very rare. I want to ask: Does anybody else have noticed something similar ? I have only one call which start playing about 20 files

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FreeSWITCH Conference Call Today
by Michael Collins 29 May '13

29 May '13

Hello all, Today's conference call agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2013_05_29 We have Martin from VoIPMonitor.org who will be sharing with us some information about the project, both the OSS piece and the commercial GUI front-end. Talk to you soon! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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

28 May '13

Happy Tuesday to you all! It is a short week for us here in the U.S. as yesterday was a national holiday. The first order of business: I am happy to let you know that Packt Publishing <http://www.packtpub.com> has informed us that the new FreeSWITCH 1.2 book <http://www.packtpub.com/freeswitch-1-2/book> is available as of May 24th! Congrats to the FreeSWITCH team for getting yet another book published. This is the third one we've done with Packt. Also, many thanks to all those who spent time answering questions and helping us with the technical reviewing process. With the new book completed we focus our attention on other things, not the least of which is ClueCon 2013 <http://www.cluecon.com>. Starting this week, those who've submitted talk proposals will be receiving the official acceptance of their presentations, including the day and time for the presentation. If you have not already submitted a talk proposal please do so right away as we have a limited number of available speaking slots. Our weekly conference calls have been lively with discussions on various topics suggested by those calling in. On last week's call<http://wiki.freeswitch.org/wiki/FS_weekly_2013_05_22>we had a particularly nice discussion about mod_skinny <http://wiki.freeswitch.org/wiki/Mod_skinny>. A relatively new community member, Nathan Neulinger, has really done a nice job of picking up the torch for mod_skinny. If you have a need to use Cisco phones with FreeSWITCH using SCCP<http://en.wikipedia.org/wiki/Skinny_Call_Control_Protocol>then definitely listen <http://torrents.freeswitch.org/conf_call_2013-05-22.torrent> to last week's discussion. This week <http://wiki.freeswitch.org/wiki/FS_weekly_2013_05_29> we have Martin from the VoIPMonitor.org project. We look forward to learning more about VoIPMonitor and how it can help us with keeping tabs on our VoIP servers. Have a great week! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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ANNOUNCEMENT: ClueCon Hotel Registration Available Online
by Michael Collins 22 May '13

22 May '13

The ClueCon team would like to announce that the hotel for ClueCon 2013 may now be booked online. Please visit www.ClueCon.com and click the "Book Your Room Online" button. Once you are at the Hyatt site you can enter in your check in and checkout dates and click the Search button. You will be presented with your choice of "standard king" or "standard double" rooms. Keep in mind that the standard double rooms go fast, so don't delay! Be sure to register <http://www.cluecon.com/register/> for ClueCon right away so that we can plan our various activities for when we're all in Chicago. See you in August! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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FreeSWITCH Conference Call Today
by Michael Collins 22 May '13

22 May '13

Hello all. Our community conference call is coming up shortly. The agenda page is here: http://wiki.freeswitch.org/wiki/FS_weekly_2013_05_22 We don't have a feature presentation today, but we do have several items to discuss so please join us at 1PM EDT, 10AM PDT. Thanks! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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Same headers in a SIP message
by Kevin Mathy 22 May '13

22 May '13

Hi List, I faced a problem with FS, concerning SIP messages with multiple same headers (for example, Diversion headers). FS wasn't able to manage those headers, and only the first one was taken in account by FS. After some searches in mod_sofia.c, I've made a patch, giving the ability to FS to display and manage multiple SIP Diversion headers (or other headers) You can find it here : http://jira.freeswitch.org/browse/FS-5411 But, I'm facing a new problem : FS is able to *read* some identical headers and display them as "Diversion", "Diversion-1", "Diversion-2", etc... , but it's not able to *write* some identical headers Here is an extract of my test dialplan <condition field="${sip_h_Diversion}" > expression="\+(\d+).*;reason=(\w+);privacy=(\w+);counter=(\d{1})" > break="never"> > <action application="log" data="LOG Modifying first Diversion" /> > * <action application="set" > data="sip_h_Diversion=<sip:+$1>;reason=$2;privacy=$3;counter=$4" /> > * </condition> > <condition field="${sip_h_Diversion-1}" > expression="\+(\d+).*;reason=(\w+);privacy=(\w+);counter=(\d{1})" > break="never"> > <action application="log" data="LOG Modifying second Diversion-1" > /> > * <action application="set" > data="sip_h_Diversion=<sip:+$1>;reason=$2;privacy=$3;counter=$4" /> > *</condition> You can see that I'm trying to write two Diversion headers, because what I'd like to obtain is something like that : INVITE sip:+123456789@fsbox.domain > From: .... > To : .... > ........ > Diversion: <sip:+987654321@domain.com > >;reason=unavailable;privacy=off;counter=1 > Diversion: <sip:+258147369@domain.com > >;reason=do-not-disturb;privacy=off;counter=1 But what I really obtain is : INVITE sip:+123456789@fsbox.domain > From: .... > To : .... > ........ > Diversion: <sip:+987654321@domain.com > >;reason=unavailable;privacy=off;counter=1 > *Diversion-1*: <sip:+258147369@domain.com > >;reason=do-not-disturb;privacy=off;counter=1 So, FS automatically write the second Diversion header with a "-1" at the end ... And it's really frustrating ! Moreover, I don't want to use a FS function to make every SIP headers copied automatically from A-leg to B-leg, because some of my SIP providers don't support every header, and I have to only send essential ones. Do you have any idea / question / answer to help me, please ? Perhaps my patch may be corrected, but I don't know how... Thanks a lot, * Bien cordialement, Best Regards, **Kevin MATHY* * * --

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