January 2014 - FreeSWITCH-dev - FreeSWITCH Mailing Lists

FreeSWITCH Friday Bug Hunt - Friday Jan 31 @ 2P Eastern
by Ken Rice 31 Jan '14

31 Jan '14

Hey Guys don¹t forget the Friday BugHunt on 888 ( sip:888@conference.freeswitch.org ) Every Friday from 2 to 4PM Eastern. Join us to help trouble shoot bugs, sort jira, and help the community with their problems live! Visit http://fs0.us/call888 for PSTN and other access methods Ken -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

1 0

gsmopen and routing incoming SMS
by Peter Villeneuve 31 Jan '14

31 Jan '14

Hi, I'm experimenting with GSMopen and I'm having trouble doing something which is likely pretty simple. I have enabled mod_sms and GSMopen in FS. I can send SIP simple messages between 2 registered peers OK. I can see that incoming GSM SMS messages arrive in FS as expected, but I can't seem to get them converted into SIP SIMPLE format and routed to the peer (1000). Once they arrive in FS, I can see in the log: 2014-01-30 20:00:54.840999 [INFO] mod_sms.c:336 Processing text message +4412398746->gsm01 in context default Chatplan: gsm01 parsing [default->basic p2p] continue=true Chatplan: gsm01 Regex (PASS) [basic p2p] to(gsm01) =~ /^(.*)$/ break=on-false Chatplan: gsm01 Action send() My chatplan has: <?xml version="1.0" encoding="utf-8"?> <include> <context name="default"> <extension name="basic p2p" continue="true"> <condition field="to" expression="^(.*)$">  <action application="send"/> </condition> </extension> </context> </include> And in my gsmopen.conf.xml: <configuration name="gsmopen.conf" description="GSMopen Configuration"> <global_settings> <param name="debug" value="8"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="hold-music" value="$${moh_uri}"/> <param name="destination" value="1000"/> </global_settings> I believe I'm doing something wrong in the chatplan. Is there any transfer action like in the regular XML dialplan? I tried creating in the default dialplan the following, hoping that it would be similar to sofia calls and transfer the incoming SMS to peer 1000, but no dice. <include> <extension name="sms_inbound"> <condition field="destination_number" expression="^(gsm01)$"> <action application="transfer" data="1000 XML default"/> </condition> </extension> </include> So, basically, how can I route the incoming SMS to peer 1000? Thanks

1 0

Transcoding g711 speex - memory leak
by Georgi Stefanov 30 Jan '14

30 Jan '14

I forgot to say: It is under Linux with 2.6.18-308.16.1.el5PAE FreeSWITCH Version 1.2.17~32bit ( 32bit)

1 0

Transcoding g711 speex - memory leak
by Georgi Stefanov 30 Jan '14

30 Jan '14

Hello All, These days I am playing with Freeswitch transcoding capabilities I succeeded to make 400+ simultaneous calls with 1500% CPU load. Freeswitch is working stable during 30-40min. The it comes the crash, but no core file found I saw in /var/log/message that Freeswitch was killed by OOM( out-of-memory) killer I repeated the test 3 times - same result Then I used the advices from http://www.oracle.com/technetwork/articles/servers-storage-dev/oom-killer-1… to handle with kernel memory overcommit strategy. OOM killer stopped killing my Freeswitch, but then when I try to make new calls I got error "MEDIA_TIMEOUT" My question is: Is it memory leak? P.S. When I do not use transcoding capabilities I do not have this "issue" Does anybody else meet similar behavior ?

2 1

ClueCon Weekly, News and Notes!!
by Ken Rice 29 Jan '14

29 Jan '14

Hey Guys, As far as the news goes, not a lot this week! Keep in mind that Tony and Crew are banging away on issues at the Annual Dev Gathering! Don¹t forget you can help them out and with a little donation to the Dev Dinner fund via the Paypal link on http://www.freeswitch.org/ Today on ClueCon Weekly, Using Jira followed by community Q&A. When 1PM EST (10AM PST) Join us at sip:888@conference.freeswitch.org or http://fs0.us/call888 for PSTN access points. -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

1 0

About "vm-mailto" para
by sparklezou 28 Jan '14

28 Jan '14

Hi FS buddies, I have reade https://wiki.freeswitch.org/wiki/Mod_voicemail#vm-mailto Multiple email addresses can be defined using a comma-separated list. When set multiple email address, such as <param name="vm-mailto" value="user101@mydomain.com,user102@mydomain.com"/> In the original send message, it shows as: To: <user101@mydomain.com,user102@mydomain.com> I think,the correct one should be: To: <user101(a)mydomain.com>,<user102(a)mydomain.com> It seems one bug, right? Thanks! 2014-01-28 sparklezou

1 0

Re: [Freeswitch-dev] [Freeswitch-users] About "origination_callee_id_name", callee ID name appears in the caller phone's display
by sparklezou 28 Jan '14

28 Jan '14

Hi Mitja and FS buddies, Thanks! Fist of all, it's the last version 1.2.18 Q1: It’s my mistake when writing email. Double checked. Two sentence is NOT work fine. <action application="set" data="origination_callee_id_name=${user_data(${dialed_extension}@${domain_name} var effective_caller_id_name)}"/> <action application="bridge" data="user/${dialed_extension}@${domain_name}"/> Q2: I think it’s the problem of Freeswith. The sip log is attached. During ring, In the FS---> Caller 183 message, Remote-Party-ID: "3366" <sip:3366@172.0.0.10>;party=calling;privacy=off;screen=no Here the Display name is NOT correct, still the phone number. It should be, Remote-Party-ID: "Jackie Chan" <sip:3366@172.0.0.10>;party=calling;privacy=off;screen=no When answered, In the FS---> Caller 200 message, Remote-Party-ID: "Jackie Chan" <sip:3366@172.0.0.10>;party=calling;privacy=off;screen=no Here Display is correct and show on the caller pone. It should be the bug. 2014-01-28 sparklezou 发件人：Mitja <unau0(a)gmx.de> 发送时间：2014-01-28 00:47 主题：Re: [Freeswitch-users] About "origination_callee_id_name", callee ID name appears in the caller phone's display 收件人："FreeSWITCH Users Help"<freeswitch-users(a)lists.freeswitch.org> 抄送： Hey, to question 1: I think in your two "sentence" example you just have closing brace too much after effective_caller_id_name. <action application="set" data="origination_callee_id_name=${user_data(${dialed_extension}@${domain_name} var effective_caller_id_name)}}"/> <action application="bridge" data="user/${dialed_extension}@${domain_name}"/> I marked the braces in colors to show you which pair belongs together. Youll see the red one has no matching opening brace. to question 2: I'm not sure but I think this is a phone depended issue and not freeswitch related. Regards, Mitja Am 27.01.2014 10:37, schrieb sparklezou: Hi All buddies, I have read the wiki, https://wiki.freeswitch.org/wiki/Variable_origination_callee_id_name And seems work now. Here still are several questions. Please help me. Thanks in advance! 1. two sentence as following is NOT work. why? <action application="set" data="origination_callee_id_name=${user_data(${dialed_extension}@${domain_name} var effective_caller_id_name)}}"/> <action application="bridge" data="user/${dialed_extension}@${domain_name}"/> one sentence as following is work. <action application="bridge" data="{origination_callee_id_name=${user_data(${dialed_extension}@${domain_name} var effective_caller_id_name)}}user/${dialed_extension}@${domain_name}"/> 2. During the ring, it's still showing the number, NOT name. During answering, it's showing the name. I hope the name is showed during ring. How to do? Thanks! 2014-01-27 sparklezou

1 0

About "origination_callee_id_name", callee ID name appears in the caller phone's display
by sparklezou 27 Jan '14

27 Jan '14

Hi All buddies, I have read the wiki, https://wiki.freeswitch.org/wiki/Variable_origination_callee_id_name And seems work now. Here still are several questions. Please help me. Thanks in advance! 1. two sentence as following is NOT work. why? <action application="set" data="origination_callee_id_name=${user_data(${dialed_extension}@${domain_name} var effective_caller_id_name)}}"/> <action application="bridge" data="user/${dialed_extension}@${domain_name}"/> one sentence as following is work. <action application="bridge" data="{origination_callee_id_name=${user_data(${dialed_extension}@${domain_name} var effective_caller_id_name)}}user/${dialed_extension}@${domain_name}"/> 2. During the ring, it's still showing the number, NOT name. During answering, it's showing the name. I hope the name is showed during ring. How to do? Thanks! 2014-01-27 sparklezou

1 0

Annual Buy The FreeSWITCH Developers Dinner Telethon is underway!
by Ken Rice 25 Jan '14

25 Jan '14

This is the one time of year all the devs get in the same place at the same time without crowds of people distracting them is rapidly approaching! This year the guys will be spending 12 hour days planning out the roadmap for the next year, working through issues in the code along with other engineering tasks that are best worked on when everyone is in the same room! And we don't want Tony feeding them all Ramen! If you could help out with the the "Dont Make them Eat Ramen Fund", Hit the Paypal link on http://freeswitch.org/ and send a few bucks! -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

1 0

Next Week!
by Brian West 25 Jan '14

25 Jan '14

FreeSWITCHers, Next week the entire team will be doing our annual engineering meet up, If you wish to donate to the dinner for devs fund please use the Paypal link on the website. Its a great way to show your appreciation and help with the week long 12+ hour days. We’ll be planning our roadmap and working thru issues on JIRA along with many other engineering tasks that make FreeSWITCH what it is today. Thanks, -- Brian West brian(a)freeswitch.org Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)

2 2