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Nibblebill multiple sql requests
by Ximo Peral 25 Sep '15

25 Sep '15

I am using FS with nibblebill.I see that SQL queries are duplicated. I copied the debug log: ^[[m^[[33m2015-09-25 13:19:39.124073 [DEBUG] switch_core_state_machine.c:40 sofia/external/***** Standard INIT ^[[m^[[33m2015-09-25 13:19:39.124073 [DEBUG] switch_core_state_machine.c:48 (sofia/external/*****) State Change CS_INIT -> CS_ROUTING ^[[m^[[33m2015-09-25 13:19:39.124073 [DEBUG] switch_core_state_machine.c:516 (sofia/external/*****) State INIT going to sleep ^[[m^[[33m2015-09-25 13:19:39.124073 [DEBUG] switch_core_state_machine.c:473 (sofia/external/*****) Running State Change CS_ROUTING ^[[m^[[33m2015-09-25 13:19:39.124073 [DEBUG] sofia.c:6715 Channel sofia/external/***** entering state [calling][0] ^[[m^[[33m2015-09-25 13:19:39.124073 [DEBUG] switch_core_state_machine.c:532 (sofia/external/*****) State ROUTING ^[[m^[[33m2015-09-25 13:19:39.124073 [DEBUG] mod_sofia.c:141 sofia/external/***** SOFIA ROUTING ^[[m^[[33m2015-09-25 13:19:39.124073 [DEBUG] switch_ivr_originate.c:67 (sofia/external/*****) State Change CS_ROUTING -> CS_CONSUME_MEDIA ^[[m^[[33m2015-09-25 13:19:39.124073 [DEBUG] mod_nibblebill.c:506 Attempting to bill at $0.09139 per minute to account 16 ^[[m^[[33m2015-09-25 13:19:39.124073 [DEBUG] mod_nibblebill.c:518 Not billing 16 - call is not in answered state ^[[m^[[33m2015-09-25 13:19:39.124073 [DEBUG] mod_nibblebill.c:422 Doing lookup query [SELECT customer_balance AS nibble_balance FROM company WHERE id='16'] ^[[m^[[33m2015-09-25 13:19:39.124073 [DEBUG] mod_nibblebill.c:430 Retrieved current balance for account 16 (balance = 0.000000) ^[[m^[[33m2015-09-25 13:19:39.124073 [DEBUG] mod_nibblebill.c:523 Comparing 0.000000 to hangup balance of -50.000000, taking into account minimum charge of 0.000000 ^[[m^[[33m2015-09-25 13:19:39.124073 [DEBUG] mod_nibblebill.c:422 Doing lookup query [SELECT customer_balance AS nibble_balance FROM company WHERE id='16'] ^[[m^[[33m2015-09-25 13:19:39.124073 [DEBUG] mod_nibblebill.c:430 Retrieved current balance for account 16 (balance = 0.000000) ^[[m^[[33m2015-09-25 13:19:39.124073 [DEBUG] switch_core_state_machine.c:532 (sofia/external/*****) State ROUTING going to sleep ^[[m^[[33m2015-09-25 13:19:39.124073 [DEBUG] switch_core_state_machine.c:473 (sofia/external/*****) Running State Change CS_CONSUME_MEDIA ^[[m^[[33m2015-09-25 13:19:39.124073 [DEBUG] switch_core_state_machine.c:551 (sofia/external/*****) State CONSUME_MEDIA ^[[m^[[33m2015-09-25 13:19:39.124073 [DEBUG] mod_nibblebill.c:506 Attempting to bill at $0.09139 per minute to account 16 ^[[m^[[33m2015-09-25 13:19:39.124073 [DEBUG] mod_nibblebill.c:518 Not billing 16 - call is not in answered state ^[[m^[[33m2015-09-25 13:19:39.124073 [DEBUG] mod_nibblebill.c:422 Doing lookup query [SELECT customer_balance AS nibble_balance FROM company WHERE id='16'] ^[[m^[[33m2015-09-25 13:19:39.124073 [DEBUG] mod_nibblebill.c:430 Retrieved current balance for account 16 (balance = 0.000000) ^[[m^[[33m2015-09-25 13:19:39.124073 [DEBUG] mod_nibblebill.c:523 Comparing 0.000000 to hangup balance of -50.000000, taking into account minimum charge of 0.000000 ^[[m^[[33m2015-09-25 13:19:39.124073 [DEBUG] mod_nibblebill.c:422 Doing lookup query [SELECT customer_balance AS nibble_balance FROM company WHERE id='16'] ^[[m^[[33m2015-09-25 13:19:39.124073 [DEBUG] mod_nibblebill.c:430 Retrieved current balance for account 16 (balance = 0.000000) ^[[m^[[35m2015-09-25 13:19:39.124073 [WARNING] switch_core_session.c:1579 sofia/external/***** using scheduler due to bypass media or media is not established. ^[[m^[[33m2015-09-25 13:19:39.124073 [DEBUG] switch_scheduler.c:249 Added task 249745 switch_core_session_sched_heartbeat (4190b143-adef-4b34-aeed-1fbf592af187) to run at 144318117 9 ^[[m^[[33m2015-09-25 13:19:39.124073 [DEBUG] switch_core_state_machine.c:551 (sofia/external/*****) State CONSUME_MEDIA going to sleep ^[[m^[[31m2015-09-25 13:19:39.124073 [ALERT] switch_core_state_machine.c:590 sofia/external/***** session thread sleep state: CS_CONSUME_MEDIA! ^[[m^[[33m2015-09-25 13:19:39.124073 [DEBUG] switch_scheduler.c:144 Deleting task 249744 switch_core_session_sched_heartbeat (d9372f20-2d25-47c5-affa-238322f57100) In the same second, 4 requests! As I see it after a "CS_ROUTING" or "CS_CONSUME_MEDIA". But many queries... Any ideas?

1 0

No audio after bridging outbound calls
by Govind Rajan M 25 Sep '15

25 Sep '15

Hi, I was testing the freeswitch with the gateway and when I make two outgoing calls and bridge them together, I am not able to hear the audio. But as soon as I played something on any one of the channel, both the parties can be able to here the audio. Did anyone face this kind of issue? Can you please help me to solve the issue? My machine is running on AWS behind NAT and i have done all the configuration that is specified in the freeswitch wiki page. P.S : when I make one outgoing to sip user and another outgoing call via SIP gateway everything works fine(both the parties able to hear the voice). -- Thanks, Govind

3 2

[NOTICE] : SSL Updates to FreeSWITCH.org and related Web Sites
by Ken Rice 24 Sep '15

24 Sep '15

As part of our routing updates and upgrades to the FreeSWITCH infrastructure we have made some changes to SSL Configurations to tighten up the security. As of today only TLS1.2 is supported and we have dropped support for several cipher suites that are known problematic or due to advancements in computational power have been rendered obsolete. What this means for you: If you are running up to date browsers or software that uses HTTPS to communicate with FreeSWITCH.org and its related web properties you will not be affected. However if you are still using older versions of web browsers there is a possibility that would will not be able to access the websites. Known such as Android 4.2.X or earlier, IE 10 or earlier, Java 7u25 or earlier and Safari 6 versions and earlier will not work. Thanks! The FreeSWITCH Team

1 0

t38 version
by Bruce Lefko 24 Sep '15

24 Sep '15

Which version of T38 does the latest freeswitch/spandsp support?

4 3

ICE SDP Params
by Serge Yuriev 24 Sep '15

24 Sep '15

Hi, I’m trying to disable any mentions about WebRTC/DTLS in SDP. My PBX can’t understand them and drops calls. I use pre_answer app and get such SDP towards my PBX send 1698 bytes to udp/[10.0.64.63]:6060 at 20:02:39.595098: ------------------------------------------------------------------------ SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.0.64.63:6060;branch=z9hG4bK300267909a88 From: "IT, ????? ??????" <sip:12550@10.0.64.63>;tag=357758~27154efa-6325-45a2-9e47-67e5d9302ebc-249340814 To: <sip:17000@10.0.64.100>;tag=73ccUeacZSQ7c Call-ID: bb282700-5f91a0af-24648-3f40000a(a)10.0.64.63 CSeq: 101 INVITE Contact: <sip:mod_sofia@10.0.64.100:6060> RSeq: 145774175 User-Agent: FreeSWITCH-mod_sofia/1.6.0+git~20150903T203652Z~6762f14140~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, PRACK, NOTIFY Require: 100rel Supported: precondition, 100rel, timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 925 v=0 o=FreeSWITCH 1442395963 1442395964 IN IP4 10.0.64.100 s=FreeSWITCH c=IN IP4 10.0.64.100 t=0 0 a=msid-semantic: WMS wp2SlBEpzCNtEIOknB2WO45EEsyGM2B8 m=audio 26996 RTP/SAVPF 18 9 8 0 3 101 13 a=rtpmap:18 G729/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=rtpmap:13 CN/8000 a=fingerprint:sha-256 09:0E:1F:D3:0D:FD:F9:54:AA:FF:AD:7F:CC:0A:85:94:D6:16:38:9F:94:43:3F:78:AC:E8:FC:8B:2B:0C:83:CA a=setup:actpass a=rtcp-mux a=rtcp:26996 IN IP4 10.0.64.100 a=ssrc:1443086239 cname:huRtVlIlOmEjUjt8 a=ssrc:1443086239 msid:wp2SlBEpzCNtEIOknB2WO45EEsyGM2B8 a0 a=ssrc:1443086239 mslabel:wp2SlBEpzCNtEIOknB2WO45EEsyGM2B8 a=ssrc:1443086239 label:wp2SlBEpzCNtEIOknB2WO45EEsyGM2B8a0 a=ice-ufrag:LmUqgPUL8iiKbJJj a=ice-pwd:3qCPCb0hnvq8FJ6gVQw2vhEu a=candidate:7296319604 1 udp 659136 10.0.64.100 26996 typ host generation 0 a=ptime:20 ———————————————————————————————————— If I use webrtc_enable_dtls=false, FS don’t send any codecs send 932 bytes to udp/[10.0.64.63]:6060 at 18:25:17.885114: ------------------------------------------------------------------------ SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.0.64.63:6060;branch=z9hG4bK311ba4379c394 From: "IT, ????? ??????" <sip:12550@10.0.64.63>;tag=364113~27154efa-6325-45a2-9e47-67e5d9302ebc-249450749 To: <sip:17000@10.0.64.100>;tag=FcQj96g8709vS Call-ID: 4b777200-5fa1db5d-24eb6-3f40000a(a)10.0.64.63 CSeq: 101 INVITE Contact: <sip:mod_sofia@10.0.64.100:6060> RSeq: 485241663 User-Agent: FreeSWITCH-mod_sofia/1.6.0+git~20150903T203652Z~6762f14140~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, PRACK, NOTIFY Require: 100rel Supported: precondition, 100rel, timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 158 v=0 o=FreeSWITCH 1442476665 1442476666 IN IP4 10.0.64.100 s=FreeSWITCH c=IN IP4 10.0.64.100 t=0 0 a=msid-semantic: WMS TqexzJilLgNDKOovoQ0kCWDMDNGQpja8 ———————————————————————————————————— How I can get rid of ICE/WebRTC/DTLS parts only? -- Serge S. Yuriev

2 1

gateway registration problem
by Mi Ke 24 Sep '15

24 Sep '15

3 6

Setting channel variable "campon=true", not populating "originating_leg_uuid" in bridge-call CDR in Master.csv file
by Debashis Bhowmick 24 Sep '15

24 Sep '15

Hi, We set the channel variable "campon=true" in freeswitch dialplan "/usr/local/freeswitch/conf/dialplan/default.xml" as below. <extension name="FS_bridge1"> <condition field="destination_number" expression="^(\d+)$"> <action application="ring_ready"/> <action application="set" data="campon=true" /> <action application="bridge" data="freetdm/outgoing/b/07022376046" /> <action application="hangup" /> </condition> </extension> We found that when we are doing a bridge call, the CDR for the bridge-call, does not have the channel-variable *originating_leg_uuid* set. On the other hand, when we made "campon=false" in dialplan (which is the default value for campon), we found that the field *originating_leg_uuid* is getting set in the bridge-call CDR. We needed the channel variable *originating_leg_uuid *for our work*.* So, is this a bug or like there is any workaround on this, so that we can get the channel variable *originating_leg_uuid* populating in bridge-call CDR, when campon=true in dialplan. A quick response would be highly appreciated. -- Regards, Debashis Bhowmick, <http://www.cse.iitd.ernet.in/~mcs102080/>

1 0

Audio files are not bein played
by Vikas Kumar 24 Sep '15

24 Sep '15

Hi All, We are using Linphone and zoiper on other hand. We have some limit in dial plan that only one call is admissible at one time. If I call from other softphone while call is in progress than this limit is hit. See logs: Dialplan: sofia/internal/1016(a)119.9.73.183<mailto:sofia/internal/1016@119.9.73.183> Action playback(/usr/local/videocallhandler/media/InQue1.wav) Dialplan: sofia/internal/1016(a)119.9.73.183<mailto:sofia/internal/1016@119.9.73.183> Action playback(/usr/local/videocallhandler/media/id.wav) Dialplan: sofia/internal/1016(a)119.9.73.183<mailto:sofia/internal/1016@119.9.73.183> Action playback(/usr/local/videocallhandler/media/InQue2.wav) Dialplan: sofia/internal/1016(a)119.9.73.183<mailto:sofia/internal/1016@119.9.73.183> Action endless_playback(/tmp/times.wav) 2015-09-22 11:38:43.542214 [DEBUG] switch_core_state_machine.c:167 (sofia/internal/1016(a)119.9.73.183<mailto:sofia/internal/1016@119.9.73.183>) State Change CS_ROUTING -> CS_EXECUTE 2015-09-22 11:38:43.542214 [DEBUG] switch_core_session.c:1357 Send signal sofia/internal/1016(a)119.9.73.183<mailto:sofia/internal/1016@119.9.73.183> [BREAK] 2015-09-22 11:38:43.542214 [DEBUG] switch_core_state_machine.c:474 (sofia/internal/1016(a)119.9.73.183<mailto:sofia/internal/1016@119.9.73.183>) State ROUTING going to sleep 2015-09-22 11:38:43.542214 [DEBUG] switch_core_state_machine.c:418 (sofia/internal/1016(a)119.9.73.183<mailto:sofia/internal/1016@119.9.73.183>) Running State Change CS_EXECUTE 2015-09-22 11:38:43.542214 [DEBUG] switch_core_state_machine.c:481 (sofia/internal/1016(a)119.9.73.183<mailto:sofia/internal/1016@119.9.73.183>) State EXECUTE 2015-09-22 11:38:43.542214 [DEBUG] mod_sofia.c:243 sofia/internal/1016(a)119.9.73.183<mailto:sofia/internal/1016@119.9.73.183> SOFIA EXECUTE 2015-09-22 11:38:43.542214 [DEBUG] switch_core_state_machine.c:209 sofia/internal/1016(a)119.9.73.183<mailto:sofia/internal/1016@119.9.73.183> Standard EXECUTE EXECUTE sofia/internal/1016(a)119.9.73.183<mailto:sofia/internal/1016@119.9.73.183> answer() 2015-09-22 11:38:43.542214 [DEBUG] sofia_glue.c:5308 Audio Codec Compare [GSM:3:8000:20:13200]/[G729:18:8000:20:8000] 2015-09-22 11:38:43.542214 [DEBUG] sofia_glue.c:5308 Audio Codec Compare [GSM:3:8000:20:13200]/[PCMA:8:8000:20:64000] 2015-09-22 11:38:43.542214 [DEBUG] sofia_glue.c:5308 Audio Codec Compare [GSM:3:8000:20:13200]/[PCMU:0:8000:20:64000] 2015-09-22 11:38:43.542214 [DEBUG] sofia_glue.c:5308 Audio Codec Compare [GSM:3:8000:20:13200]/[GSM:3:8000:20:13200] 2015-09-22 11:38:43.542214 [DEBUG] sofia_glue.c:3216 Set Codec sofia/internal/1016(a)119.9.73.183<mailto:sofia/internal/1016@119.9.73.183> GSM/8000 20 ms 160 samples 13200 bits 2015-09-22 11:38:43.542214 [DEBUG] switch_core_codec.c:111 sofia/internal/1016(a)119.9.73.183<mailto:sofia/internal/1016@119.9.73.183> Original read codec set to GSM:3 2015-09-22 11:38:43.542214 [DEBUG] sofia_glue.c:5477 Set 2833 dtmf send/recv payload to 101 2015-09-22 11:38:43.542214 [DEBUG] sofia_glue.c:3475 AUDIO RTP [sofia/internal/1016(a)119.9.73.183] 119.9.73.183 port 25094 -> 10.150.178.79 port 60204 codec: 3 ms: 20 2015-09-22 11:38:43.542214 [DEBUG] switch_rtp.c:2041 Starting timer [soft] 160 bytes per 20ms 2015-09-22 11:38:43.542214 [DEBUG] sofia_glue.c:3742 Set 2833 dtmf send payload to 101 2015-09-22 11:38:43.542214 [DEBUG] sofia_glue.c:3748 Set 2833 dtmf receive payload to 101 2015-09-22 11:38:43.542214 [DEBUG] sofia_glue.c:3775 sofia/internal/1016(a)119.9.73.183<mailto:sofia/internal/1016@119.9.73.183> Set rtp dtmf delay to 40 2015-09-22 11:38:43.542214 [NOTICE] sofia_glue.c:4386 Pre-Answer sofia/internal/1016(a)119.9.73.183<mailto:sofia/internal/1016@119.9.73.183>! 2015-09-22 11:38:43.542214 [DEBUG] switch_channel.c:3370 (sofia/internal/1016(a)119.9.73.183<mailto:sofia/internal/1016@119.9.73.183>) Callstate Change RINGING -> EARLY 2015-09-22 11:38:43.542214 [DEBUG] mod_sofia.c:866 Local SDP sofia/internal/1016(a)119.9.73.183<mailto:sofia/internal/1016@119.9.73.183>: v=0 o=FreeSWITCH 1442877029 1442877030 IN IP4 119.9.73.183 s=FreeSWITCH c=IN IP4 119.9.73.183 t=0 0 m=audio 25094 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv Warning given is "/usr/local/videocallhandler/media/Hold.wav sample rate 11025 doesn't match requested rate 8000" Should I ignore this warning, ?? Thanks Vikas Disclaimer :- The information in this e-mail and any attachments is confidential and may be legally privileged. It is intended solely for the addressee or addressees. If you are not an intended recipient, please delete the message and any attachments and notify the sender of misdelivery. Any use or disclosure of the contents of either is unauthorised and may be unlawful. All liability for viruses is excluded to the fullest extent permitted by law. Any views expressed in this message are those of the individual sender, except where the sender states them, with requisite authority, to be those of the specific TIMES GROUP company.

2 2

channel variable access issue
by John Nash 24 Sep '15

24 Sep '15

Here is my call flow and issue Step1 -- Call comes to freeswitch default dialplan Step2 -- I set some custom channel variables )sip_h_my_var Step3 -- I fork the call to multiple destinations using bridge application with coma separated list. Step4 - As one of the destination is my own freeswitch itself, call again comes to same profile with different destination prefix Step5 - I catch the call in different extension and try to access channel variable sip_h_my_var but get nil value I tried export application and Bridge application arguments to pass variables but I do not see my variables when i dump variables using Info application. I only see the variables as per the Initial invite (It converts some X- variables to sip_h_X variables) What else can I try?

2 6

Timeout counted from receiving 180
by Joseph Waite 23 Sep '15

23 Sep '15

Hi Everyone Im wondering if there is a way to set a timeout ina similar way to originate_timeout when placing a call but that only starts the timeout counter after freeswitch recieves the 180 Ringing from the carrier. We have a carrier which on an invalid number does not send a 404 straight away, instead they send early media with a recorded message for over 30 seconds saying number is invalid. This causes the call to timeout and be classed in CDR's as a NOANSWER instead of INVALID I have run a sip trace and both a valid and invalid number recieve a 183 eraly media message but only a valid number recieves a 180. I have tried ignore_early_media but this does not seem to work. Regards

3 3