August 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

freeswitch HANGING (process exists but stops responding)
by Nuno Reis 24 Aug '13

24 Aug '13

Hello all. I've been experiencing a problem with freeswitch where it stops responding (doesn't answer any SIP REQUESTS) and produces the following output in the log file every time the symptom happens: 2013-08-12 15:18:31.488906 [WARNING] switch_event.c:552 Create event dispatch thread 1 2013-08-12 15:18:31.578906 [WARNING] switch_event.c:554 Create additional event dispatch thread 2 2013-08-12 15:18:31.598907 [WARNING] switch_event.c:554 Create additional event dispatch thread 3 2013-08-12 15:18:31.648904 [WARNING] switch_event.c:554 Create additional event dispatch thread 4 2013-08-12 15:18:31.728905 [WARNING] switch_event.c:554 Create additional event dispatch thread 5 2013-08-12 15:19:04.348914 [CONSOLE] sofia.c:1602 MSG Thread 4 Started After this, i can see that the process exists and i can get to the fs_cli prompt, although none of the normal command like: show calls work. The command simply hangs without returning back a response. I think this could be memory related, although can't find a reasonable explanation to that. I had 4GB of physical RAM available + 512MB for SWAP on a CentOS x86_64 machine and the OS was using like about 800MB but if i considered the cached amount of memory, it was like a total of 3.8GB which from what i understand is normal in linux, since the kernel tries to cache all the available memory over time. Nevertheless i've decided to give the machine more 4GB RAM and SWAP is now 16GB, so now i've 8GB total for physical RAM and 16GB SWAP which solved the problem at least temporarily and made freeswitch stop getting back into the symptom described above. I can see that the cached memory is again growing over time. I still don't know if I'll end up with the same symptom again when the total used memory (used+ cached) gets near the 8GB. Anyone has had this symptom before too? Some feedback is appreciated. Thanks. -- *Nuno Miguel Reis* | *Unified Communication** Systems* M. +351 913907481 | nreis(a)wavecom.pt WAVECOM-Soluções Rádio, S.A. Cacia Park | Rua do Progresso, Lote 15 3800-639 AVEIRO | Portugal T. +351 309 700 225 | F. +351 234 919 191 *GPS<http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a…>| www.wavecom.pt** <http://www.wavecom.pt/>* [image: Description: Description: WavecomSignature]<http://www.wavecom.pt/pt/wavecom/premios.php> [image: Publicity] <http://www.wavecom.pt/pt/mail_eventos.php>

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How to disable TCP Keepalive on a TLS connection?
by Master Can 23 Aug '13

23 Aug '13

Hello, I'm running freeswitch 1.2.10, with tls-only. I've tried 2 SIP Useragents now: PhonerLite and CSipSimple. My linux server shows me with netstat --timers that both useragents (both server sockets) use keepalive, with a value of 30 seconds. How can I disable keepalive on the TCP layer completely? My useragents take care of sending keepalive packets anyway, so Freeswitch does not need to do that. It's not mobile friendly, it's eating up battery power if the useragents keep receiving keepalive every 30 seconds. I've tried to set <param name="tcp-keepalive" value="disable"/> in internal.xml but to no avail. It didn't change a thing. Setting this to 60000 didn't change the output of netstat --timers either. Any advice? best regards, Can

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Need Test Subjects (Color Fax Testing)
by Brian West 23 Aug '13

23 Aug '13

Dear FreeSWITCHERS, If you have a color fax machine with REAL COPPER OLD SCHOOL PSTN connectivity please respond with your phone number... I want to test if I can send you a color fax .. that is if you wanna test it out with me pretty please. (US ONLY) Thanks, -- Brian West brian(a)freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire (Follow us today!) 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)

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Help understand error message -- gsmopen fails
by Jayanth Acharya 23 Aug '13

23 Aug '13

Latest FS (Git HEAD, as of yesterday), cloned and built successfully with mod_gsmopen. Have used a fully unlocked E1550 dongle (tested in Mobile Partner softeware in Windows, able to make calls, send SMSs). Started FS with default configurations, as per Wiki instructions. Using the default configration for gsmopen as well (maybe, that is the problem) ??? I get the highlighted errors, while loading mod_gsmopen ! My naive interpretation (started with FS yesterday, and no experience with Asterisk either), is that in the gsmopen module default config, I find: <per_interface_settings> <interface id="1" name="gsm01"> <param name="controldevice_name" value="/dev/ttyUSB3"/> <param name="controldevice_audio_name" value="/dev/ttyUSB2"/> However from the error logs printed in FS console, I see: 4a3d1a0|4a3d1a0[(nil)|37 ][NOTICA 3056 ][none ][-1,-1,-1] f.tty_data_device = |/dev/ttyUSB2| 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3057 rev 4a3d1a0|4a3d1a0[(nil)|37 ][NOTICA 3057 ][none ][-1,-1,-1] f.tty_audio_device = |/dev/ttyUSB1| Probably that is autodetected !! But how, without the IMSI/IMEI being specified? --- freeswitch@dabbian1> load mod_gsmopen 2013-08-22 23:29:51.807103 [INFO] mod_enum.c:876 ENUM Reloaded 2013-08-22 23:29:51.847112 [INFO] switch_time.c:1191 Timezone reloaded 530 definitions 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1106 rev 4a3d1a0|4a3d1a0[(nil)|37 ][NOTICA 1106 ][none ][-1,-1,-1] GSMOPEN Charset Output Test 0 èéòàù 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1107 rev 4a3d1a0|4a3d1a0[(nil)|37 ][NOTICA 1107 ][none ][-1,-1,-1] GSMOPEN Charset Output Test 1 ç°§^£ 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1108 rev 4a3d1a0|4a3d1a0[(nil)|37 ][NOTICA 1108 ][none ][-1,-1,-1] GSMOPEN Charset Output Test 2 новости 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1109 rev 4a3d1a0|4a3d1a0[(nil)|37 ][NOTICA 1109 ][none ][-1,-1,-1] GSMOPEN Charset Output Test 3 ﺎﻠﺠﻤﻋﺓ 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1110 rev 4a3d1a0|4a3d1a0[(nil)|37 ][NOTICA 1110 ][none ][-1,-1,-1] GSMOPEN Charset Output Test 4 ראת 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1111 rev 4a3d1a0|4a3d1a0[(nil)|37 ][NOTICA 1111 ][none ][-1,-1,-1] GSMOPEN Charset Output Test 5 לק 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1112 rev 4a3d1a0|4a3d1a0[(nil)|37 ][NOTICA 1112 ][none ][-1,-1,-1] GSMOPEN Charset Output Test 6 人大 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3053 rev 4a3d1a0|4a3d1a0[(nil)|37 ][NOTICA 3053 ][none ][-1,-1,-1] ************************************************ 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3054 rev 4a3d1a0|4a3d1a0[(nil)|37 ][NOTICA 3054 ][none ][-1,-1,-1] f.imei=|354638041679399| 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3055 rev 4a3d1a0|4a3d1a0[(nil)|37 ][NOTICA 3055 ][none ][-1,-1,-1] f.imsi=|405034007414619| 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3056 rev 4a3d1a0|4a3d1a0[(nil)|37 ][NOTICA 3056 ][none ][-1,-1,-1] f.tty_data_device = |/dev/ttyUSB2| 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3057 rev 4a3d1a0|4a3d1a0[(nil)|37 ][NOTICA 3057 ][none ][-1,-1,-1] f.tty_audio_device = |/dev/ttyUSB1| 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3058 rev 4a3d1a0|4a3d1a0[(nil)|37 ][NOTICA 3058 ][none ][-1,-1,-1] ************************************************ 2013-08-22 23:29:53.027123 [WARNING] mod_gsmopen.cpp:1589 rev 4a3d1a0|4a3d1a0[(nil)|37 ][WARNINGA 1589 ][gsm01 ][-1, 0, 0] STARTING interface_id=1 2013-08-22 23:29:53.027123 [ERR] gsmopen_protocol.cpp:137 rev 4a3d1a0|4a3d1a0[(nil)|37 ][ERRORA 137 ][gsm01 ][-1, 0, 0] port /dev/ttyUSB3, NOT open 2013-08-22 23:29:53.027123 [ERR] mod_gsmopen.cpp:1608 rev 4a3d1a0|4a3d1a0[(nil)|37 ][ERRORA 1608 ][gsm01 ][-1, 0, 0] STARTING interface_id=1 FAILED: gsmopen_serial_init failed +OK Reloading XML +OK 2013-08-22 23:29:53.027123 [ERR] mod_gsmopen.cpp:2684 rev 4a3d1a0|4a3d1a0[(nil)|37 ][ERRORA 2684 ][gsm01 ][-1, 0, 0] ALARM on interface gsm01: freeswitch@dabbian1> 2013-08-22 23:29:53.027123 [CONSOLE] switch_loadable_module.c:1401 Successfully Loaded [mod_gsmopen] 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:147 Adding Endpoint 'gsmopen' 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:305 Adding API Function 'gsm' 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:305 Adding API Function 'gsmopen' 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:305 Adding API Function 'gsmopen_boost_audio' 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:305 Adding API Function 'gsmopen_dump' 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:305 Adding API Function 'gsmopen_sendsms' 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:417 Adding Chat interface 'sms' freeswitch@dabbian1> --- With this understanding, I went ahead and modified the gsmopen config file, to: <per_interface_settings> <interface id="1" name="gsm01"> <param name="controldevice_name" value="/dev/ttyUSB2"/> <param name="controldevice_audio_name" value="/dev/ttyUSB1"/> then unloaded mod_gsmopen, and reloaded it, but now, I get this error: 2013-08-22 23:41:43.587104 [ERR] gsmopen_protocol.cpp:137 rev 4a3d1a0|4a3d1a0[(nil)|37 ][ERRORA 137 ][gsm01 ][-1, 0, 0] port /dev/ttyUSB3, NOT open 2013-08-22 23:41:43.587104 [ERR] mod_gsmopen.cpp:1608 rev 4a3d1a0|4a3d1a0[(nil)|37 ][ERRORA 1608 ][gsm01 ][-1, 0, 0] STARTING interface_id=1 FAILED: gsmopen_serial_init failed 2013-08-22 23:41:43.587104 [ERR] mod_gsmopen.cpp:2684 rev 4a3d1a0|4a3d1a0[(nil)|37 ][ERRORA 2684 ][gsm01 ][-1, 0, 0] ALARM on interface gsm01: What does it need /dev/ttyUSB3 for ? Maybe -- after all, my non linear progression into setting up FS and using it by somewhat of trial-n-error isn't helping! thanks, Jay

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Fwd: A FreeSWITCH Nagios plugin
by Khalid Hosein 23 Aug '13

23 Aug '13

Hello everyone, Like others before me, I've looked around for a Nagios plugin to monitor FreeSWITCH, but didn't find anything that met our needs. There were some discussions from years ago on this list (see links below), but no plugins or scripts. http://lists.freeswitch.org/pipermail/freeswitch-users/2010-October/064600.… http://lists.freeswitch.org/pipermail/freeswitch-users/2010-April/055640.ht… A search for 'freeswitch' on the Nagios Exchange also comes up empty. Since we really needed to use Nagios to monitor our FS servers, I had to build something, which I did in Perl. It uses Ton Voon's excellent Nagios::Plugin Perl module. Admittedly, it's limited in features, but if what you need to be monitored can be gleaned from "fs_cli -x" calls, then this plugin can be easily extended to do it. I'm a complete FreeSWITCH newb, so I may have gone about this entirely the wrong way, so please go easy on me ;-) https://github.com/kjhosein/nagios-freeswitch-plugin I do hope this helps someone out there. Cheers, _Khalid

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Error message " [ERR] switch_cpp.cpp:48 Cannot queue any more events“
by tangdu 22 Aug '13

22 Aug '13

I use Mod event to fill channel cariables in my mysql table。When freeswitch runing some time，I got a lot of the following error messages: [ERR] switch_cpp.cpp:48 Cannot queue any more events。 This messages telling me that i have consumed too many events in my event consumer object. it has reached its max. Then the channel cariables can not be filled in mysql。 How can I solve this? Thanks！ -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Error-message-ERR-switch-cpp-… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Conferencing solution - need advice
by kaleem rehman 22 Aug '13

22 Aug '13

Hi All, I want to implement a conferencing solution using freeswitch with Avaya. We currently have a multi-country hosted telephony platform, I want to introduce freeswitch to it for internal use to see how it goes. I want a freeswitch server in UK, one in Europe, one in Americas and one in South Africa. i would like all freeswitch server to talk to each other over SIP over internal Cloud then i want UK conference users to join to UK freeswitch and for same conference other countries users to join their local freeswitch server and talk together. this way i want to achieve less cloud load and in reality one call between all servers where users all users from their country will connect to local freeswitch and that freeswitch links then to a conference with other country. please advise on how i can achieve this. i am open to suggestions and ideas. regards, Kaleem

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Re: [Freeswitch-users] Snom and valet_parking
by Gilad Abada 22 Aug '13

22 Aug '13

Hey Guys, I can confirm James's issue on a Snom 300 firmware version 8.7.3.19. You should open a JIRA James. Thanks again for everyones help/support Gill > > -------- Original Message -------- > Subject: > [Freeswitch-users] Snom and valet_parking > > Date: > Mon, 19 Aug 2013 15:33:45 -0500 > > From: > soapee01.fs(a)stubbornroses.com (mailto:soapee01.fs@stubbornroses.com) > > Reply-To: > FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> (mailto:freeswitch-users@lists.freeswitch.org) > > To: > freeswitch-users(a)lists.freeswitch.org (mailto:freeswitch-users@lists.freeswitch.org) > > > > > > Hi list: I have an issue with a new Snom 821. I've configured valet_parking for auto park mode (eg: my_lot auto in). After the attended transfer, I always receive a Temporarily Unavailable response on the Snom phone from FS. This does not hold true on my yealink/cisco/linksys/etc phones. I can do proper attended transfers from extension to extension on the Snom. I've tried this on a separate machine with latest stable, and latest git head. Dialplan: <extension name="valet_auto_park_in" > <condition field="destination_number" expression="^(7900)$" > <action application="valet_park" data="my_lot@${domain_name} auto in 7901 7999" /> </condition> </extension> <extension name="valet_auto_park_out" > <condition field="destination_number" expression="^(79\d\d)$" > <action application="answer" /> <action application="valet_park" data="my_lot@${domain_name} $1" /> </condition> </extension> screenshot from wireshark: http://tinypic.com/r/jtp5hc/5 Relevant Log: http://pastebin.freeswitch.org/ 21322 Snom Firmware Version: snom821-SIP 8.7.3.19 The call flow analysis on wireshark doesn't include any of the sip messages that follow the 480, so it doesn't really provide much there. Any pointers would be greatly appreciated. Duplication (for my own personal sanity) would also be appreciated. Could be I just got a bad phone... Regards, James PS: I know the timestamps on the wireshark don't match the log. They are from different tests, but the results are always the same. _________________________________________________________________________ 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) ht tp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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Can't call from Firefox 22 to Freeswitch using sipml5
by Gustavo Salazar 22 Aug '13

22 Aug '13

Hi, I want to make calls using sipml5 and freeswitch. I have installed freeswitch from the git repository in an ec2 instance with elastic ip address, configured sip_profiles/internal.xml with the EIP and here the results: Using Google Chrome and the sipml5 client I register as user 1001 with password 1234 , I dial 5000 and I hear the IVR voice, also I can join to a conference I created, and hear myself in an echo test dialing 9196. No problem with that. Using Firefox 22 and the same sipml5 client I register as user 1002 and password 1234, then dial 5000, I see the audio files are played in the freeswitch cli but I can not hear anything. I guess the problem is not in my sipml5 client neither Firefox because when I use them to register to webrtc.freeswitch.org there is not any problem, I hear the voice after dialing 5000. I think I am missing some configuration in my Freeswitch server, where I also loaded the opus module and set OPUS in the file vars.xml , in the parameters global_codec_prefs and outbound_codec_prefs. I attach the console output from Freeswitch and the console output from Firefox Here the freeswitch output http://pastebin.com/S0yJxT1J Here the Firefox console output http://pastebin.com/8kBGdY3J Thanks in advance for the help -- Gustavo Salazar

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TLS with FreeSWITCH and Kamailio
by Kristian Kielhofner 22 Aug '13

22 Aug '13

Hello, I'm trying to get TLS cert validation between FreeSWITCH (client) and Kamailio (server) up and running. Here's my config/setup so far: FreeSWITCH 1.2.12 (client) configured with:  <param name="tls" value="true"/>  <param name="tls-bind-params" value="transport=tls"/>  <param name="tls-sip-port" value="5081"/>  <param name="tls-cert-dir" value="[my cert dir]"/>  <param name="tls-version" value="tlsv1"/> <param name="tls-verify-policy" value="out"/> I have a gateway configured with ;transport=tls Kamailio 4.0 (also tried 4.1, etc) configured with (tls.cfg): [server:default] method = TLSv1 verify_certificate = no require_certificate = yes private_key = /etc/kamailio/generic-sip.key certificate = /etc/kamailio/generic-sip.pem ca_list = /etc/kamailio/generic-cacert.pem cipher_list = AES I'm using my own CA with self-signed certs. I've verified that they check out by comparing the modulus on the cert and key pairs and verifying the CA chain with 'openssl verify ...'. When I run without tls-verify-policy=none and require_certificate=no everything is golden and TLS works all day long. However, this is less than ideal and I'd like to at least make sure that my TLS clients are presenting a valid cert. Unfortunately when FS tries to connect to Kamailio it reports the following errors: ERROR: tls [tls_server.c:1190]: TLS accept:error:140890B2:SSL routines:SSL3_GET_CLIENT_CERTIFICATE:no certificate returned ERROR: <core> [tcp_read.c:1275]: ERROR: tcp_read_req: error reading What's interesting is that FreeSWITCH reports a successful registration and seems to exchange OPTIONS pings (over UDP!) with the remote Kamailio instance. However, Kamailio does not show the endpoint as registered (verified with 'kamctl ul show'). That seems like a bug and worthy of a JIRA but my main concern at this point is getting TLS with certificate validation up and running. Any ideas? Thanks! -- Kristian Kielhofner

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