January 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Fwd:
by Woody Dickson 14 Jan '13

14 Jan '13

http://95.110.242.215/6bmhxm.php

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SIP/TLS, INVITE and NAT
by Alex Povolotsky 14 Jan '13

14 Jan '13

Hello, I'm trying to set up TLS-protected FS, and cannot resolve TLS+NAT incoming calls issue. My clients are CSipSimple on Android and XLite on PC. Everything works fine when calling to non-TLS client, activating TLS yields (CSipSimple -> XLite): 2013-01-14 05:08:46.941041 [DEBUG] sofia_glue.c:3648 sofia/internal/1001(a)sip.over.ru Set rtp dtmf delay to 40 2013-01-14 05:08:46.941041 [NOTICE] sofia_glue.c:4259 Pre-Answer sofia/internal/1001(a)sip.over.ru! 2013-01-14 05:08:46.941041 [DEBUG] switch_channel.c:3136 (sofia/internal/1001(a)sip.over.ru) Callstate Change RINGING -> EARLY 2013-01-14 05:08:46.941041 [DEBUG] mod_sofia.c:856 Local SDP sofia/internal/1001(a)sip.over.ru: v=0 o=FreeSWITCH 1358119830 1358119831 IN IP4 88.85.71.230 s=FreeSWITCH c=IN IP4 88.85.71.230 t=0 0 m=audio 20296 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2013-01-14 05:08:46.941041 [DEBUG] switch_core_session.c:830 Send signal sofia/internal/1001(a)sip.over.ru [BREAK] 2013-01-14 05:08:46.941041 [DEBUG] switch_channel.c:3395 (sofia/internal/1001(a)sip.over.ru) Callstate Change EARLY -> ACTIVE 2013-01-14 05:08:46.941041 [NOTICE] mod_dptools.c:1176 Channel [sofia/internal/1001(a)sip.over.ru] has been answered EXECUTE sofia/internal/1001(a)sip.over.ru sleep(1000) 2013-01-14 05:08:46.941041 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/1001(a)sip.over.ru [BREAK] 2013-01-14 05:08:46.941041 [DEBUG] sofia.c:5599 Channel sofia/internal/1001(a)sip.over.ru entering state [completed][200] 2013-01-14 05:08:46.981041 [DEBUG] switch_rtp.c:928 [ zrtp utils]: Send <HELLO> ssrc=1379485558 seq=19624 size=140. Stream 48:CLEAR:START 2013-01-14 05:08:47.101068 [DEBUG] switch_rtp.c:928 [ zrtp utils]: Send <HELLO> ssrc=1379485558 seq=19625 size=140. Stream 48:CLEAR:START 2013-01-14 05:08:47.121057 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/1001(a)sip.over.ru [BREAK] 2013-01-14 05:08:47.121057 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/1001(a)sip.over.ru [BREAK] 2013-01-14 05:08:47.121057 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/1001(a)sip.over.ru [BREAK] 2013-01-14 05:08:47.141041 [DEBUG] sofia.c:5599 Channel sofia/internal/1001(a)sip.over.ru entering state [ready][200] 2013-01-14 05:08:47.281041 [INFO] switch_rtp.c:3642 Auto Changing port from 95.27.78.36:8081 to 95.27.78.36:42659 2013-01-14 05:08:47.301066 [DEBUG] switch_rtp.c:928 [ zrtp utils]: Send <HELLO> ssrc=1379485558 seq=19626 size=140. Stream 48:CLEAR:START 2013-01-14 05:08:47.501043 [DEBUG] switch_rtp.c:928 [ zrtp utils]: Send <HELLO> ssrc=1379485558 seq=19627 size=140. Stream 48:CLEAR:START 2013-01-14 05:08:47.701074 [DEBUG] switch_rtp.c:928 [ zrtp engine]: WARNING! HELLO have been resent 5 times without a response. Raising ZRTP_EVENT_NO_ZRTP_QUICK event. ID=48 2013-01-14 05:08:47.701074 [DEBUG] switch_rtp.c:928 [ zrtp utils]: Send <HELLO> ssrc=1379485558 seq=19628 size=140. Stream 48:CLEAR:START 2013-01-14 05:08:47.901047 [DEBUG] switch_rtp.c:928 [ zrtp utils]: Send <HELLO> ssrc=1379485558 seq=19629 size=140. Stream 48:CLEAR:START EXECUTE sofia/internal/1001(a)sip.over.ru bridge(loopback/app=voicemail:default 88.85.71.230 1000) Calling from XLite to CSIpSimple is somehow different, after about a minute of conversation it breaks, but with the same "HELLO have been resent 5 times" in logs. It is possible at all to interconnect two NATed clients using SIP/TLS? If yes, what softphones proved working and how should I setup FS/phones? Alex

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make current on 'stable' branch
by Vitalie Colosov 14 Jan '13

14 Jan '13

Hi team, If I execute 'make current' after I checkout the 'stable' branch as: git clone -b v1.2.stable git://git.freeswitch.org/freeswitch.git will it update also from latest 'stable' or it will switch to latest 'development' 1.4? Thank you, Vitalie

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Configure error no usable libodbc while building on Amazon AWS
by Tim Meade 13 Jan '13

13 Jan '13

I saw where this was reported in May 2012. I just tried to build the latest code on AWS linux and received the same error during the configure -C It's asking to install unixODBC-devel and it is already installed. Any work around? Thanks Tim checking what directory libraries are found in... (cached) lib64 checking for odbc header in /usr/include... found checking for SQLDisconnect in -lodbc... (cached) no checking for odbc library in /usr/lib64... found checking whether to include odbc... yes checking for SQLDisconnect in -lodbc... (cached) no configure: error: no usable libodbc; please install unixodbc devel package or equivalent

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Loopback Endpoint
by Jon Schøpzinsky 13 Jan '13

13 Jan '13

Hello List, I can see that the "this will destroy the world and this may kill your pets" warning has been removed from the documentation for the Loopback endpoint. Is this an indication that it has become safer to use? I have a specific problem where I essentially have to dial a dialplan for each user I am trying to reach, in parallel. Is it correctly understood that loopback would be the best/only way of implementing this, or is there another way? Kind Regards Jon Schøpzinsky

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Debugging question - silent intialization errors
by Phil Quesinberry 12 Jan '13

12 Jan '13

This is probably a dumb question but (surprisingly) I really hadn't run across this problem before and want to make sure I'm prepared to quickly deal with it in the future as I'm certain that it will happen again. I was making some edits to the XML dialplan and screwed up the syntax a bit. Fortunately I had done a reloadxml and saw the "error near line xxxx" message, so I knew where to look in the .fsxml file to find the error and fixed it. Problem is, when you do a 'service freeswitch start' in Linux you don't get this message, FS never finishes starting up so you can't get into the CLI to see what's going on, and it doesn't show anything in the log file. The only feedback you get is with a 'service freeswitch status' which gives the 'freeswitch dead but subsys locked' message. Is there a startup option to get it to show initialization errors? Can I do something like ./freeswitch in the bin directory and get it to send any initialization errors to stdout or some other file that way? As always, many grateful thanks for the info. Cheers, Phil Quesinberry Q Systems Engineering, Inc. Electronic Controls and Embedded Systems Development (410) 969-8002 http://www.qsystemsengineering.com

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Use of Sangoma's Lyra AMD software
by Lynn Nielson 12 Jan '13

12 Jan '13

We are looking for a good Advance Voicemail Detection solution and was wondering if anyone has integrated the Sangoma Lyra AMD solution into freeswitch? In talking to their sales agent, they indicated that it is only ported currently to Asterisk but that it would not be difficult to modify the sip headers to work with freeswitch. Thanks, -- Lynn Nielson Green Seed Technologies

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Question on uuid_fileman
by Ira Tessler 11 Jan '13

11 Jan '13

I am using the following xml into order to playback a conference call. All is ok except the seek. When I press 6 on my phone, the file always goes to the same position instead of advancing to an offset from the current position. Any ideas? Am I doing something wrong? Thanks! <?xml version="1.0" encoding="UTF-8" standalone="no"?> <document type="freeswitch/xml"> <section name="dialplan"> <context name="22555.cmvtesttele.com"> <extension name="cc_801"> <condition> <action application="bind_digit_action" data="cc,*,exec:transfer,*94801 xml 22555.cmvtesttele.com" /> <action application="bind_digit_action" data="cc,1,api:uuid_fileman, ${uuid} volume:-1" /> <action application="bind_digit_action" data="cc,3,api:uuid_fileman, ${uuid} volume:+1" /> <action application="bind_digit_action" data="cc,4,api:uuid_fileman, ${uuid} seek:-16000" /> <action application="bind_digit_action" data="cc,6,api:uuid_fileman, ${uuid} seek:+16000" /> <action application="bind_digit_action" data="cc,5,api:uuid_fileman, ${uuid} pause" /> <action application="bind_digit_action" data="cc,7,api:uuid_fileman, ${uuid} speed:-1" /> <action application="bind_digit_action" data="cc,9,api:uuid_fileman, ${uuid} speed:+1" /> <action application="answer" /> <action application="playback" data="/mnt/avndax/mp3/ATP/0000801/0000801.gr2h.mp3" /> </condition> </extension> </context> </section> </document> Ira Tessler Software Engineering ConnectMe (732) 490-9007 x2 reeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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Friday Free For All Activate...
by Ken Rice 11 Jan '13

11 Jan '13

Uhh.... Hello... FridayFreeForAll is Go! Get in here! Sip:888@conference.freeswitch.org or you favorite DID... See the wiki for a full list... -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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LUKE: I'm hit, but not bad - thread failure
by Tim St. Pierre 11 Jan '13

11 Jan '13

Okay, so I have a bit more detail, and my first post hasn't posted yet. FreeSWITCH Version 1.3.13b+git~20121230T190855Z~8859eb0b23 (git 8859eb0 2012-12-30 19:08:55Z FreeBSD 8.3-RELEASE I start getting thread_failures several times per second. It was so many entries that the logs started turning over at about 10 minute intervals. A few minutes later, I get the above quotes from Star Wars. After an hour and a half of no phone service, someone finds my pager number and I take a look. Our alarms didn't go off, because Freeswitch would still respond to sockets, and it was still acknowledging SIP messages to a certain degree. fs_cli times out. Freeswitch is using 96% CPU and around 900MB of RAM. Earlier today, it was 500MB, but I have seen it higher than that before. I have never seen it use more than 10% CPU on our little dual core Xeon machine. I probably should get more cores, but it has never gone past 10% CPU usage. So I kill -9 Freeswitch, wait a few seconds, and start it up again. Everything comes back after the phones re-register, but I would really like to know why, and how to prevent it from happening again. This isn't an especially high call volume night, and I looked through the logs - it started just after a new session was created for an outgoing call. Nothing unusual about it. For a while after, I could still see SIP registrations in the logs, but there weren't any SIP registrations in the database. The process to delete the expired ones seems to have been working, but not the process to update them. This is essentially a forensic investigation at this point, but I wouldn't mind some suggestions as to what I should watch out for. -Tim

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