July 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

stacksize is too large: run ulimit -s 240
by Richard Klingler 02 Jul '12

02 Jul '12

Good evening Seems asterisk is not the only one haviong problems on FreeBSD (o; Starting freeswitch from the shell without "-nc" brings it up fine and operates. But when staring with "-nc" or via the rc.d script it complains about stacksize being too big: Error: stacksize 524288 is too large: run ulimit -s 240 from your shell before starting the application. auto-adjusting stack size for optimal performance... 79103 Backgrounding. It states though it runs then in the background, but it just crashes without any logs. Anyone got freeswitch 1.0.7 running on amd64 FreeBSD 9.0? thanx in advance richard

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tls ca setup
by Alexandre Fiori 02 Jul '12

02 Jul '12

Bria 3 suddenly stopped working on my mac, reporting this: All accounts failed to enable Account: test could not be enabled. Problem at server, error 503. Try again later. Nothing shows up on fs_cli, but tcpdump shows traffic. Changing sofia to loglevel 9 gives me this: tport_wakeup_pri(0x164dbd0): events IN tport_alloc_secondary(0x164dbd0): new secondary tport 0x7f39b9acce80 tport_tls_accept(0x7f39b9acce80): new connection from tls/x.x.x.x:33351/sips tls_connect(0x7f39b9acce80): events NEGOTIATING tls_connect(0x7f39b9acce80): events NEGOTIATING tls_connect(0x7f39b9acce80): TLS setup failed (error:00000001:lib(0):func(0):reason(1)) tport_close(0x7f39b9acce80): tls/x.x.x.x:33351/sips This is not a happy Canada day, where's my phone? It turns out the self-signed root CA generated by `gentls_cert setup` has expired. How I figured it out? First, on the server: # openssl x509 -noout -in /opt/freeswitch/conf/ssl/CA/cacert.pem -dates notBefore=Jun 2 01:44:26 2012 GMT notAfter=Jul 1 01:44:26 2012 GMT Second, because I opened https://my-server:5061 on Safari and got a "This certificate is not valid (expired root)". It seems the script is missing `-days`, here: http://git.freeswitch.org/git/freeswitch/tree/scripts/gentls_cert.in#n83 Manually adding it fixed the problem. -- Ship, ahoy! Hast seen the White Whale? - Cap'n Ahab

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Re: [Freeswitch-users] RTP not sent while waiting for DTMF in "read" or "play_and_get_digits"
by Peter Olsson 02 Jul '12

02 Jul '12

This should probably be enough; http://wiki.freeswitch.org/wiki/Variable_send_silence_when_idle Set it to 1400 Also, there is http://wiki.freeswitch.org/wiki/Variable_record_waste_resources, if you want to send RTP during record. /Peter -----Ursprungligt meddelande----- Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Mariusz Czulada Skickat: den 2 juli 2012 15:56 Till: FreeSWITCH Users Ämne: [Freeswitch-users] RTP not sent while waiting for DTMF in "read" or "play_and_get_digits" Hi all, I faced a problem with applications mentioned in the subject. Both of them, after playing a prompt message, start to intercept DTMF codes from the endpoint for a specified time. During this period _no_ _RTP_ packets are sent towards calling side, which causes (in my case - ALU IMS) C-BGF to diagnose end of media in channel, which in turn causes P-CSCF to terminate a call. My question: is it possible not to eforce FSW to send 'silence' RTP while waiting for user DTMF actions? TIA, Mariusz _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)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 Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4ff1a79f32767637819317!

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RTP not sent while waiting for DTMF in "read" or "play_and_get_digits"
by Mariusz Czulada 02 Jul '12

02 Jul '12

Hi all, I faced a problem with applications mentioned in the subject. Both of them, after playing a prompt message, start to intercept DTMF codes from the endpoint for a specified time. During this period _no_ _RTP_ packets are sent towards calling side, which causes (in my case - ALU IMS) C-BGF to diagnose end of media in channel, which in turn causes P-CSCF to terminate a call. My question: is it possible not to eforce FSW to send 'silence' RTP while waiting for user DTMF actions? TIA, Mariusz

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Re: [Freeswitch-users] How about a USER FORUM and kill off the mail list?
by Евгений 02 Jul '12

02 Jul '12

Hi, friends. I decided to make forum by myself. So if someone wants to join - I'll be happy :) I made it in both languages - English and Russian(my language). First of all I plan to find out what structures and threads are needed in forum. Who are interested - welcome to http://freeswitchforum.com Best regards, Evginey. С уважением Евгений. FreeSWITCH форум - http://freeswitchforum.com FreeSWITCH user-forum - http://freeswitchforum.com

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Leap second & FS using more CPU
by Avi Marcus 02 Jul '12

02 Jul '12

I was wondering why FS which usually uses only 8-10% CPU at for my 20 active calls was suddenly using 60%. I was pointed to the leap-second being a possible cause. So, as per: http://marc.info/?l=linux-kernel&m=134113577921904 I ran: $ date -s "`date`" and the CPU usage went immediately back to the norm. Not sure if it glitched the audio when doing so, though... This is just a PSA and YMMV. -Avi

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Simple SPA8800 FXO connection to FreeSwitch
by Robin Gilks 01 Jul '12

01 Jul '12

Greetings I'm a noob with FS (and VoIP generally!!) and am really struggling with getting 2 way comms between a PABX and a soft-phone using FS and an SPA8800. Calls from the PABX are OK, the FXO ports have registered with FS and when I dial the FXO extension I get the dial tone, dial the number for the soft-phone and it all works fine. The problem is connections *TO* the PABX. The SPA8800 AFAIK has a separate SIP port for each of its FXO ports. My understanding of how it should work is as follows: I have to have 4 gateways defined in a file in the sip-profiles/externals directory, each with a different "sip-port" line. I'm testing with 2 to start with: # cat cicsco_fxo.xml <include> <gateway name="fxoline3"> <param name="username" value="none"/> <param name="password" value="none"/> <param name="realm" value="172.27.1.20"/> <param name="sip-port" value="5260"/> <param name="register" value="false"/> </gateway> <gateway name="fxoline4"> <param name="username" value="none"/> <param name="password" value="none"/> <param name="realm" value="172.27.1.20"/> <param name="sip-port" value="5360"/> <param name="register" value="false"/> </gateway> I have to use mod_distributor to select one of the gateways that define the SIP ports. Thats is configured in the autoload_configs directory. # cat distributor.conf.xml <configuration name="distributor.conf" description="Distributor Configuration"> <lists> <list name="cisco_fxo" total-weight="2"> <node name="fxoline3" weight="1"/> <node name="fxoline4" weight="1"/> </list> </lists> </configuration> Finally, I use a dialplan to access this distribution of gateways when a 4 digit number starting with 8 is dialed from the soft-phone. # cat 33_ciscofox.xml <include> <extension name="pabx"> <condition field="destination_number" expression="^(8\d{3})$"> <action application="bridge" data="sofia/gateway/${distributor(cisco_fxo)}/$1"/> </condition> </extension> </include> Result expected - it all works. Result I get is: freeswitch(a)PBX.localdomain> 2012-06-29 04:13:44.287982 [NOTICE] switch_channel.c:926 New Channel sofia/internal/1002(a)172.16.164.33 [d0fc47e6-c1a0-11e1-881d-f70f560259d2] 2012-06-29 04:13:44.287982 [INFO] mod_dialplan_xml.c:485 Processing 1002 <1002>->8569 in context default 2012-06-29 04:13:44.298584 [ERR] mod_sofia.c:4492 Invalid Gateway '' 2012-06-29 04:13:44.298584 [NOTICE] mod_sofia.c:4889 Close Channel N/A [CS_NEW] 2012-06-29 04:13:44.298584 [NOTICE] switch_ivr_originate.c:2535 Cannot create outgoing channel of type [sofia] cause: [INVALID_GATEWAY] 2012-06-29 04:13:44.298584 [INFO] mod_dptools.c:2956 Originate Failed. Cause: INVALID_GATEWAY 2012-06-29 04:13:44.298584 [NOTICE] mod_dptools.c:3076 Hangup sofia/internal/1002(a)172.16.164.33 [CS_EXECUTE] [INVALID_GATEWAY] 2012-06-29 04:13:44.298584 [NOTICE] switch_core_session.c:1447 Session 11 (sofia/internal/1002(a)172.16.164.33) Ended 2012-06-29 04:13:44.298584 [NOTICE] switch_core_session.c:1449 Close Channel sofia/internal/1002(a)172.16.164.33 [CS_DESTROY] So what is invalid about the gateway? Cheers -- Robin Gilks -- ------------------------------ This email, including any attachments, is only for the intended recipient. It is subject to copyright, is confidential and may be the subject of legal or other privilege, none of which is waived or lost by reason of this transmission. If you are not an intended recipient, you may not use, disseminate, distribute or reproduce such email, any attachments, or any part thereof. If you have received a message in error, please notify the sender immediately and erase all copies of the message and any attachments. Unfortunately, we cannot warrant that the email has not been altered or corrupted during transmission nor can we guarantee that any email or any attachments are free from computer viruses or other conditions which may damage or interfere with recipient data, hardware or software. The recipient relies upon its own procedures and assumes all risk of use and of opening any attachments. ------------------------------

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limit getting reset after call state change
by Muhammad Naseer Bhatti 01 Jul '12

01 Jul '12

I am using execute_on_originate to impose limit for outgoing gateway. Limit is set to 1 here, the call is established and right after call state changes, The limit is dropped. Is this normal? 2012-06-28 10:26:35.522664 [NOTICE] switch_channel.c:926 New Channel sofia/vBilling/9198 [97924522-c0f2-11e1-b7cf-a5c0d2d3774a] 2012-06-28 10:26:35.522664 [DEBUG] mod_sofia.c:4775 (sofia/vBilling/9198) State Change CS_NEW -> CS_INIT 2012-06-28 10:26:35.522664 [DEBUG] switch_core_session.c:1229 Send signal sofia/vBilling/9198 [BREAK] EXECUTE sofia/vBilling/9198 limit(hash 10.211.55.5 10.211.55.5 1 !CALL_REJECTED) *2012-06-28 10:26:35.522664 [INFO] switch_limit.c:126 incr called: 10.211.55.5_10.211.55.5 max:1, interval:0* *2012-06-28 10:26:35.522664 [INFO] mod_hash.c:202 Usage for 10.211.55.5_10.211.55.5 is now 1/1* 2012-06-28 10:26:35.522664 [DEBUG] switch_core_state_machine.c:385 (sofia/vBilling/9198) Running State Change CS_INIT 2012-06-28 10:26:35.522664 [DEBUG] switch_core_state_machine.c:424 (sofia/vBilling/9198) State INIT 2012-06-28 10:26:35.522664 [DEBUG] mod_sofia.c:85 sofia/vBilling/9198 SOFIA INIT 2012-06-28 10:26:35.522664 [DEBUG] sofia_glue.c:2602 Local SDP: v=0 o=- 3550145379 3550145379 IN IP4 192.168.40.141 s=pjmedia c=IN IP4 192.168.40.141 t=0 0 a=X-nat:0 m=audio 4014 RTP/AVP 103 102 104 109 3 0 8 9 101 a=rtpmap:103 speex/16000 a=rtpmap:102 speex/8000 a=rtpmap:104 speex/32000 a=rtpmap:109 iLBC/8000 a=fmtp:109 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtcp:4015 IN IP4 192.168.40.141 2012-06-28 10:26:35.522664 [DEBUG] mod_sofia.c:125 (sofia/vBilling/9198) State Change CS_INIT -> CS_ROUTING *2012-06-28 10:26:35.522664 [INFO] mod_hash.c:304 Usage for 10.211.55.5_10.211.55.5 is now 0* 2012-06-28 10:26:35.522664 [DEBUG] switch_core_session.c:1229 Send signal sofia/vBilling/9198 [BREAK] 2012-06-28 10:26:35.522664 [DEBUG] switch_core_state_machine.c:424 (sofia/vBilling/9198) State INIT going to sleep 2012-06-28 10:26:35.522664 [DEBUG] switch_core_state_machine.c:385 (sofia/vBilling/9198) Running State Change CS_ROUTING 2012-06-28 10:26:35.522664 [DEBUG] switch_channel.c:1919 (sofia/vBilling/9198) Callstate Change DOWN -> RINGING 2012-06-28 10:26:35.522664 [DEBUG] switch_core_state_machine.c:433 (sofia/vBilling/9198) State ROUTING

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Strange issue setting variables for mod_xml_cdr
by Muhammad Naseer Bhatti 01 Jul '12

01 Jul '12

I have gone nuts trying to figure out what is going on. Using xml_cdr to post CDR(s) to webserver, The Bridge is done outside the lua script, and all vars are set in lua and exported. .... session:setVariable("import", "gateway") session:setVariable("import", "admin_rate_group") session:setVariable("import", "sell_rate") session:setVariable("import", "cost_rate") .... The problem is that if I set sell_rate and cost_rate both variables together, who ever is in the end disapears from the XML POST. If I change their order, for example set cost_rate first and then sell_rate, then sell_rate works fine and cost_rate goes off. Looks like some sort of variable overlapping. I am HEAD with latest/latest. freeswitch.consoleLog(FREESWITCH_CONSOLE_LOG_LEVEL, "\n cost_rate ".. cost_rate .." \n"); freeswitch.consoleLog(FREESWITCH_CONSOLE_LOG_LEVEL, "\n sell_rate ".. sell_rate .." \n"); This works fine. There is data in these two variables. I am setting up these session variables along with other variables for the bridge. log-b-leg is false in the config for xml_cdr. http://pastebin.freeswitch.org/19408 is the complete PB for the call log. http://pastebin.freeswitch.org/19409 is the complete XML CDR debug. cdr_csv shows the same "Bhatti",*"2.10000","0.19280"*,"415591","9198","default","2012-06-28 03:32:29","2012-06-28 03:32:29","2012-06-28 03:32:32","3","3","NORMAL_CLEARING","be292014-c0b8-11e1-ae74-2bc60749f960","","","","" "Bhatti",*"2.10000",""*,"415591","9198","default","2012-06-28 03:32:29","2012-06-28 03:32:29","2012-06-28 03:32:32","3","3","NORMAL_CLEARING","be2782fe-c0b8-11e1-ae6f-2bc60749f960","be292014-c0b8-11e1-ae74-2bc60749f960","","","" Sometimes it works sometimes it wont. All the variables are seen in the call log. Am I doing something wrong or we hit some sort of a bug? Thank.

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Unregister sip user via fs_cli
by Jack Nikolas 01 Jul '12

01 Jul '12

Hi Guys, i am new on freeswitch, please let me know is there any command to unregister or remove the user who is registering now using fs_cli ? thanks for help.. ----------------- Regards Jack

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