July 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

[Weird FreeSwitch Duplicate UUID Error]
by Bilal Abbasi 02 Aug '16

02 Aug '16

Hi Users, I am getting some weird error on Freeswitch, that is coming on calling random extensions, I have a dialplan for 3 digit extensions , so when i can 194 it works perfect and when i call 197 it gaves me below error, Duplicate UUID! unable to create channel. What i have in my dial plan is, <action application="set" data="b-leg-uuid=${create_uuid()}"/> <action application="bridge" data="{origination_uuid=${b-leg-uuid},api_on_answer='uuid_audio ${b-leg-uuid} start write mute -4'}user/${destination_number}@ ${domain_name}"/> Can someone please guide me what i am doing wrong in this, P.S: some times calling 197 works perfect and it comes on 194 WEIRD. Regards Bilal Abbasi

3 6

Typo in Lua API confluence page
by Joel Serrano 01 Aug '16

01 Aug '16

Hi guys, There is a little typo in this page: https://freeswitch.org/confluence/display/FREESWITCH/Lua+API+Reference session:execute session:execute(app, data) local mySound = "/usr/local/freeswitch/sounds/music/16000/partita-no-3-in-e-major-bwv-1006-1-preludio.wav" session:execute("playback", mySound) NOTE: Callbacks (DTMF and friends) CAN NOT EXECUTE during an execute. session:executeString session:execute(api_string) *<<<< Shouldn't this be executeString???* local mySound = "/usr/local/freeswitch/sounds/music/16000/partita-no-3-in-e-major-bwv-1006-1-preludio.wav" session:executeString("playback "..mySound) Note: Callbacks (DTMF and friends) CAN NOT EXECUTE during an execute. Can anybody update it please? Can lead to confusion. Thanks! Joel.

2 1

freeswitch wan.auto ipv6 problem
by Sangram Rath 01 Aug '16

01 Aug '16

Hi Group, We have an issue here where we are not able to make inbound/outbound calls using a websocket application that uses sipjs. We think that wan.auto is not excluding IPv6. This problem is related to ICE. There was a bug on this that was fixed in 1.4.19. Additional info: - The platform is Kazoo - FS version: 1.4.23 (also tried upgrading to 1.4.26) - IPv6 is disabled in the OS We have the below in logs: [WARNING] switch_core_media.c:2790 NO candidate ACL defined, Defaulting to wan.auto [DEBUG] switch_core_media.c:5178 AUDIO RTP [sofia/sipinterface_1/ DPRD025488(a)somedomain.com] x.x.x.x port 24092 -> 2001::5ef5:79fd:207f:3d43:3f57:fe92 port 61975 codec: 8 ms: 20 [DEBUG] switch_rtp.c:3588 Not using a timer [INFO] switch_core_media.c:5352 Activating Audio ICE [NOTICE] switch_rtp.c:4029 Activating RTP audio ICE: bveNJceaOHdVG0Tq:tThuJX7NS3O8dtZ1 2001::5ef5:79fd:207f:3d43:3f57:fe92:61975 [INFO] switch_core_media.c:5395 Activating RTCP PORT 61975 [DEBUG] switch_rtp.c:3929 RTCP send rate is: 10000 and packet rate is: 20000 Remote Port: 61975 [DEBUG] switch_rtp.c:2349 Setting RTCP remote addr to :61975 [INFO] switch_core_media.c:5403 Skipping RTCP ICE (Same as RTP) [INFO] switch_rtp.c:3104 Activate RTP/RTCP audio DTLS client It would be great to have ideas on what could be the reason. Thanks, Sangram

2 2

CallWithUs hacked
by Mitch Capper 30 Jul '16

30 Jul '16

Incase you missed their email that just went out. " Dear Customers, On Jun 27th hackers broke in to our system and got access to SIP credentials. No other information leaked. We identified the way they gained access and applied corresponding updates to the system. No personal information or credit card data were compromised (we do not store credit card info in our system). As a security measure to prevent unauthorized use of your account we reset SIP passwords to random values. Please update the passwords in your SIP clients/devices, you can see new password in "VoIP accounts" menu in customer portal. Disregard this message if you use call authentication by IP address or do not use a SIP client. Sorry for the inconvenience caused. --- CallWithUs.com team " ~Mitch

1 0

Testing for call hangup in dialplan xml
by Cody Rosenbloom 29 Jul '16

29 Jul '16

Hi, Is there a way to check if call has been hung up in dialplan xml. I want to run lua script to process voicemail after caller has hungup the call. I tried checking if state is equal to CS_HANGUP but it does not work. Current dialplan xml is at http://pastebin.com/5PMKSrdh Thanks Cody

4 4

Difference between execute and executeString
by Joel Serrano 29 Jul '16

29 Jul '16

Hi, Using LUA or Python languages... What is the difference when running something with "api.execute(...)" or "api.executeString(...)" ?? I know there is a format difference, but anything else? Thanks! Joel.

1 0

absolute_codec_string help
by Cj B 29 Jul '16

29 Jul '16

Hi all, I have couple extensions where I want to force all incoming/outgoing calls to be g729 no matter what. I’ve added absolute_codec_string to the extension configuration. Here it is from memcache: <?xml version="1.0" encoding="UTF-8" standalone="no"?> <document type="freeswitch/xml"> <section name="directory"> <domain name="mydomain.com" alias="true"> <params> <param name="jsonrpc-allowed-methods" value="verto"/> <param name="jsonrpc-allowed-event-channels" value="demo,conference,presence"/> </params> <groups> <group name="default"> <users> <user id="1001" type=> <params> <param name="password" value=“mypasswordfromextension"/> <param name="vm-enabled" value="true"/> <param name="vm-password" value=“voicemailpassword"/> <param name="vm-email-all-messages" value="true"/> <param name="vm-attach-file" value="true"/> <param name="vm-keep-local-after-email" value="true"/> <param name="vm-mailto" value="hthoene(a)gmail.com"/> <param name="dial-string" value="{sip_invite_domain=mydomain.com,presence_id=1001@mydomain.com}${sofia_contact(1001@mydomain.com)}"/> <param name="verto-context" value="mydomain.com"/> <param name="verto-dialplan" value="XML"/> <param name="jsonrpc-allowed-methods" value="verto"/> <param name="jsonrpc-allowed-event-channels" value="demo,conference,presence"/> </params> <variables> <variable name="domain_uuid" value="d85584d5-5816-4621-9fa5-08b0f1e7ff0a"/> <variable name="domain_name" value="mydomain.com"/> <variable name="extension_uuid" value="e3d2cab9-cfcd-4235-9ea8-142b3caf1b0e"/> <variable name="call_timeout" value="30"/> <variable name="caller_id_name" value="1001"/> <variable name="caller_id_number" value="1001"/> <variable name="call_screen_enabled" value="false"/> <variable name="hold_music" value="local_stream://default"/> <variable name="accountcode" value="mydomain.com"/> <variable name="user_context" value="mydomain.com"/> <variable name="directory-visible" value="true"/> <variable name="directory-exten-visible" value="true"/> <variable name="limit_max" value="5"/> <variable name="absolute_codec_string" value=“g729"/> <variable name="record_stereo" value="true"/> <variable name="transfer_fallback_extension" value="operator"/> <variable name="export_vars" value="domain_name"/> </variables> </user> </users> </group> </groups> </domain> </section> </document> But I’m still seeing calls from the phone in PCMU codec. The phone is an SPA2102. I’ve also made sure that g729 is enabled on the phone itself… any other suggestions on how to force all incoming and outgoing calls on this leg to be g729? Thanks, Cj B

2 2

Not able to Send * DTMF
by Uday kumar 29 Jul '16

29 Jul '16

Hi All, I am using Freeswitch 1.6 with CentOS 7.2. When I am making outbound call from Freeswitch using extension on Softphone/Ip Phone then I am able to send all DTMF only * not able to send. When I have to send * DTMF then I am pressing two time * then its only working. So whenever I have to send 8 DTMF then I need press two time * from softphone/IP Phone. *On CLI I am getting only one like when pressing one time * after that nothing getting on CLI.* 2016-07-29 18:01:42.680250 [DEBUG] switch_rtp.c:6147 RTP RECV DTMF *:800 *When I am pressing two time ** 2016-07-29 18:01:42.680250 [DEBUG] switch_rtp.c:6147 RTP RECV DTMF *:800 2016-07-29 18:01:42.880250 [DEBUG] switch_rtp.c:6147 RTP RECV DTMF *:640 2016-07-29 18:01:42.880250 [DEBUG] switch_channel.c:488 RECV DTMF *:640 2016-07-29 18:01:42.880250 [DEBUG] switch_ivr_bridge.c:470 Send signal sofia/external/525585262618 [BREAK] 2016-07-29 18:01:42.900250 [DEBUG] switch_rtp.c:4610 Send start packet for [*] ts=36960 dur=160/160/640 seq=23645 lw=36960 2016-07-29 18:01:42.920250 [DEBUG] switch_rtp.c:4509 Send middle packet for [*] ts=36960 dur=320/320/640 seq=23646 lw=37120 2016-07-29 18:01:42.940250 [DEBUG] switch_rtp.c:4509 Send middle packet for [*] ts=36960 dur=480/480/640 seq=23647 lw=37280 2016-07-29 18:01:42.960250 [DEBUG] switch_rtp.c:4509 Send end packet for [*] ts=36960 dur=640/640/640 seq=23648 lw=37280 2016-07-29 18:01:42.960250 [DEBUG] switch_rtp.c:4509 Send end packet for [*] ts=36960 dur=640/640/640 seq=23649 lw=37280 2016-07-29 18:01:42.960250 [DEBUG] switch_rtp.c:4509 Send end packet for [*] ts=36960 dur=640/640/640 seq=23650 lw=37280 2016-07-29 18:01:42.960250 [DEBUG] switch_rtp.c:4462 Queue digit delay of 40ms 2016-07-29 18:01:42.960250 [DEBUG] switch_core_session.c:1062 Send signal sofia/external/525585262618 [BREAK] Please help. -- Thanks & Regard Uday Site:- www.shareyourknowledge.in Mobile:- +91-9377579349

3 2

FussionPBX with FreeSwitch
by uzair iltifat 29 Jul '16

29 Jul '16

Hi All, Thanks Olgierd Ziolko for helping me with freeswitch Installation, Now i am trying to install FusionPBX for Freeswitch I have CentOS 6 (Desktop) installed on my Virtual Machine, my VM has internet access To install i have disabled the SELinux and then installed EPEL and OKay RPM repositories. When i run the yum install fusionpbx-all command, I am having this error ---- Error: Package: freeswitch-application-memcache-1.6.9-1.el6.x86_64 (okay) Requires: libmemcached.so.11()(64bit) Error: Package: freeswitch-application-memcache-1.6.9-1.el6.x86_64 (okay) Requires: libmemcachedutil.so.2()(64bit) You could try using --skip-broken to work around the problem You could try running: rpm -Va --nofiles --nodigest ---- Kindly reply me ASAP. Thanks in advanced. http://uzair.ahmed.netcv.com/http://uzair.ahmed.netcv.com/ Uzair Ahmed [image: http://]about.me/uzairIltifat <http://about.me/uzairIltifat?promo=email_sig> <http://about.me/uzairIltifat?promo=email_sig> http://uzair.ahmed.netcv.com/

4 8

Duplicate UUID
by Petr Parýzek 29 Jul '16

29 Jul '16

Hi all, I'm using Freeswitch with a few Javascript scripts like automatic phone dialer. Sometimes, when called party is BUSY, FS reports "Duplicate UUID!" error and the session was ended with case DESTINATION_OUT_OF_ORDER. This is contents of console from yesterday: 2016-07-27 20:46:56.249500 [INFO] amds.js:315 {origination_uuid=82b8a17d-c627-45dd-9eaf-86e9f31c57b1,origination_caller_id _number=950330980,origination_caller_id_name=AMDS}sofia/gateway/gw91/0725045 121|sofia/gateway/gw90/0725045121 2016-07-27 20:46:56.249500 [NOTICE] switch_channel.c:1105 New Channel sofia/external/0725045121 [82b8a17d-c627-45dd-9eaf-86e9f31c57b1] 2016-07-27 20:46:57.789511 [NOTICE] sofia.c:7891 Hangup sofia/external/0725045121 [CS_CONSUME_MEDIA] [USER_BUSY] 2016-07-27 20:46:57.799501 [CRIT] switch_core_session.c:2232 Duplicate UUID! 2016-07-27 20:46:57.799501 [CRIT] mod_sofia.c:4456 Error Creating Session 2016-07-27 20:46:57.799501 [NOTICE] switch_ivr_originate.c:2762 Cannot create outgoing channel of type [sofia] cause: [DESTINATION_OUT_OF_ORDER] 2016-07-27 20:46:57.799501 [NOTICE] switch_core_session.c:1664 Session 9903 (sofia/external/0725045121) Ended 2016-07-27 20:46:57.799501 [NOTICE] switch_core_session.c:1668 Close Channel sofia/external/0725045121 [CS_DESTROY] 2016-07-27 20:46:57.799501 [NOTICE] amds.js:86 Script exited with info [] Can anyone say what is wrong? Thanks! Petr

4 7