August 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

xml_curl bindings=phrases issue
by Davide Rossi 28 Aug '13

28 Aug '13

Hi everybody, I'm a newbie using freeswitch (1.2.7) and having an issue with xml_curl phrases: reading logs I don't see any request to fetch data and using "xml_curl debug_on" no file is generated. Bindings to "configuration" and "dialplan" worked like a charm... Do I have to configure something special for "phrases" bindings? I have static IVR menus calling a dynamic phrase macro in http://localhost/getphrases.php. Is it a misconfiguration or maybe a bug like http://jira.freeswitch.org/browse/FS-5563? Can't find help on wiki, any hint is really appreciated. ---xml_curl.conf.xml--- <configuration name="xml_curl.conf" description="cURL XML Gateway"> <bindings> <binding name="phrases"> <param name="gateway-url" bindings="phrases" value="http://localhost/getphrases.php"/> </binding> </bindings> </configuration> ---console log [7]--- [DEBUG] switch_ivr_menu.c:469 Executing IVR menu xxx [DEBUG] switch_ivr_play_say.c:70 No language specified - Using [xx] [ERR] switch_ivr_play_say.c:145 Can't find macro xxx_ivr_main_menu. [WARNING] switch_ivr_play_say.c:348 Macro [xxx_ivr_main_menu]: '' did not match any patterns [DEBUG] switch_ivr_menu.c:363 waiting for 4/4 digits t/o 2000 [DEBUG] switch_ivr_menu.c:410 digits ''

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freeswitch on ubuntu VM on hyper-v
by lloyd thomas 28 Aug '13

28 Aug '13

I just need a little advice setting up a freeswitch VM box shared with a http VM on hyper-v host and was wondering whether there are any gotcha's that I should be aware of. I expect about 50 devices to be registered on it at any time and not expecting a high volume of calls. It will be a fairly low spec dual core machine (dell sc1425), with 8gb ram Thanks in advance for any advice.

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Evidently I Got Hacked, Please Disregard
by Michael S Collins 28 Aug '13

28 Aug '13

Gang, Apologies but it seems someone or something sent out a naughty email. This looks eerily similar to something I rec'd from TJ Vanderpoel a few days ago. Please disregard that link while I go buy some Lysol to disinfect my computer. -MC

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Newbie question for remote office
by Charles Bujold 27 Aug '13

27 Aug '13

We have 2 offices. Office A has the freeswitch server and some phones. "Office B" is new and we have 2 phones that we want to connect to the Freeswitch server located in office A so that calls can be made by the staff using Freeswitch. The question is how is this done? Here is how I see it but it does not seem to work. Please help! Office A has the Freeswitch server which connects to an external VOIP provider. The Internal clients connect via port 5060 to the Freeswitch internal profile. There phones work no problems. The IP for the Freeswitch server in office A is 192.168.50.50. The Public IP is 142.xx.xxx.xxx. Office B has 2 phones both are configured to connect to the public IP of "office A" on port 5060 which is routed to the freeswitch server. The Phones have a local IP of 192.168.23.6 and 9. The issue is that the phones are never able to register ( both side have identical settings) or transmit a call from one office to the other. Freeswitch does not seem to know how to route the calls to the other office. Ultimately we would like the staff from "office B" to be able to receive and make calls coming into the Freeswitch server. 1) Is this not possible over the Internet? Do we need to set up a VPN between both offices for this to work? Is there a way to tell freeswitch that ext 340 is located in office B? Thanks cjb

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regarding block DTMF
by Veerabhadra Rao 27 Aug '13

27 Aug '13

hi, I am working on freeswitch-1.0.6 version. In this i want to block DTMF dialled in specific leg after bridging is done. When i worked on 1.2.4 version i found "switch_ivr_block_dtmf_session(session)" which will block DTMF but now i am working on 1.0.6 , the above function is not working in this version. So is there any replacement for blocking dtmf in 1.0.6 version. please help me, thank you

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Mod_gsmopen help
by Ashish Mishra 27 Aug '13

27 Aug '13

Hi jayanth, This is the new thread that i am starting. I am using the dongle E1732 huawei which works well as far as receiving and dialing the calls is concerned on windows. I am currently using ubuntu 12.04 LTS for freeswitch.

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Re: [Freeswitch-users] Issue with JSSIP + Freeswitch
by Shahrzad A. 27 Aug '13

27 Aug '13

Hi Michael Thanks for your reply, Would you please advice me how to change the ACL to fix this issue. Here is my log when I try to call from a JSSIP/Sipml5 to another JSSIP/Sipml5 client: 2013-08-27 13:33:37.742425 [NOTICE] switch_channel.c:1030 New Channel sofia/internal/1008(a)10.0.14.16 [771e9dfb-ef68-4adc-9e5a-380e172e1945] 2013-08-27 13:33:37.762422 [INFO] mod_dialplan_xml.c:558 Processing Lala <1008>->1004 in context public 2013-08-27 13:33:37.762422 [NOTICE] switch_ivr.c:1831 Transfer sofia/internal/1008(a)10.0.14.16 to XML[1004@default] 2013-08-27 13:33:37.762422 [INFO] mod_dialplan_xml.c:558 Processing Lala <1008>->1004 in context default 2013-08-27 13:33:37.762422 [INFO] switch_ivr_async.c:3628 Bound B-Leg: *1 execute_extension::dx XML features 2013-08-27 13:33:37.762422 [INFO] switch_ivr_async.c:3628 Bound B-Leg: *2 record_session::/home/saz/freeService/recordings/1008.2013-08-27-13-33-37.wav 2013-08-27 13:33:37.762422 [INFO] switch_ivr_async.c:3628 Bound B-Leg: *3 execute_extension::cf XML features 2013-08-27 13:33:37.762422 [INFO] switch_ivr_async.c:3628 Bound B-Leg: *4 execute_extension::att_xfer XML features 2013-08-27 13:33:37.762422 [NOTICE] switch_channel.c:1030 New Channel sofia/internal/sip:1004@df7jal23ls0d.invalid[876787a6-a8f5-4d1c-bd98-ab2246a9114f] 2013-08-27 13:33:37.802411 [NOTICE] sofia.c:5898 Ring-Ready sofia/internal/sip:1004@df7jal23ls0d.invalid! 2013-08-27 13:33:37.802411 [WARNING] switch_channel.c:3250 rtp_secure_media invalid in this context. 2013-08-27 13:33:37.802411 [INFO] switch_ivr_originate.c:1190 Sending early media 2013-08-27 13:33:37.802411 [WARNING] switch_core_media.c:2074 NO candidate ACL defined, Defaulting to wan.auto 2013-08-27 13:33:37.802411 [NOTICE] switch_core_media.c:2107 Choose audio Candidate cid: 1 proto: udp type: host addr: 10.0.14.254:33865 2013-08-27 13:33:37.802411 [NOTICE] switch_core_media.c:2107 Choose audio Candidate cid: 2 proto: udp type: host addr: 10.0.14.254:33865 2013-08-27 13:33:37.802411 [NOTICE] switch_core_media.c:2112 Save audio Candidate cid: 1 proto: udp type: host addr: 10.0.14.253:53815 2013-08-27 13:33:37.802411 [NOTICE] switch_core_media.c:2112 Save audio Candidate cid: 2 proto: udp type: host addr: 10.0.14.253:53815 2013-08-27 13:33:37.802411 [NOTICE] switch_core_media.c:2221 setting remote audio ice addr to 10.0.14.254:33865 based on candidate 2013-08-27 13:33:37.802411 [NOTICE] switch_core_media.c:2239 setting remote rtcp audio addr to 10.0.14.254:33865 based on candidate 2013-08-27 13:33:37.822462 [INFO] switch_core_media.c:4227 Activating Audio ICE 2013-08-27 13:33:37.822462 [NOTICE] switch_rtp.c:3323 Activating RTP audio ICE: ruUP7F49vDADu+Gp:GMkGzis5aAzjoC0v 10.0.14.254:33865 2013-08-27 13:33:37.822462 [INFO] switch_core_media.c:4270 Activating RTCP PORT 33865 2013-08-27 13:33:37.822462 [INFO] switch_core_media.c:4278 Skipping RTCP ICE (Same as RTP) 2013-08-27 13:33:37.822462 [INFO] switch_rtp.c:2711 Activating Audio Secure RTP SEND 2013-08-27 13:33:37.822462 [INFO] switch_rtp.c:2689 Activating Audio Secure RTP RECV 2013-08-27 13:33:37.822462 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1008(a)10.0.14.16! 2013-08-27 13:33:37.922423 [NOTICE] switch_rtp.c:1053 Auto Changing stun/rtp/dtls port from 10.0.14.254:33865 to 10.0.14.253:53815 2013-08-27 13:33:42.342422 [WARNING] switch_core_media.c:2074 NO candidate ACL defined, Defaulting to wan.auto 2013-08-27 13:33:42.342422 [NOTICE] switch_core_media.c:2107 Choose audio Candidate cid: 1 proto: udp type: host addr: 10.0.14.254:45217 2013-08-27 13:33:42.342422 [NOTICE] switch_core_media.c:2112 Save audio Candidate cid: 1 proto: udp type: host addr: 10.0.14.253:38004 2013-08-27 13:33:42.342422 [NOTICE] switch_core_media.c:2221 setting remote audio ice addr to 10.0.14.254:45217 based on candidate 2013-08-27 13:33:42.342422 [INFO] switch_core_media.c:4227 Activating Audio ICE 2013-08-27 13:33:42.342422 [NOTICE] switch_rtp.c:3323 Activating RTP audio ICE: gd9dRK830iEDLw2W:vECHRV23TjEY0c0I 10.0.14.254:45217 2013-08-27 13:33:42.342422 [INFO] switch_core_media.c:4270 Activating RTCP PORT 45217 2013-08-27 13:33:42.342422 [INFO] switch_rtp.c:2478 Activate RTP/RTCP audio DTLS server 2013-08-27 13:33:42.342422 [NOTICE] sofia.c:6561 Channel [sofia/internal/sip:1004@df7jal23ls0d.invalid] has been answered 2013-08-27 13:33:42.362407 [NOTICE] switch_ivr_originate.c:3437 Channel [sofia/internal/1008(a)10.0.14.16] has been answered As you can see there is no DTLS or handshake! But the interesting part is, when I try to call from SIPML5 client to a JSSIP client I have the following log with handshake and the voice is perfect! would you please tell me what i'm missing! 2013-08-27 13:37:05.242424 [NOTICE] switch_channel.c:1030 New Channel sofia/internal/1008(a)10.0.14.16 [14f54a7b-6077-4bc3-b845-0b8bbfa3d59d] 2013-08-27 13:37:05.282411 [INFO] mod_dialplan_xml.c:558 Processing Lala <1008>->1003 in context public 2013-08-27 13:37:05.282411 [NOTICE] switch_ivr.c:1831 Transfer sofia/internal/1008(a)10.0.14.16 to XML[1003@default] 2013-08-27 13:37:05.282411 [INFO] mod_dialplan_xml.c:558 Processing Lala <1008>->1003 in context default 2013-08-27 13:37:05.282411 [INFO] switch_ivr_async.c:3628 Bound B-Leg: *1 execute_extension::dx XML features 2013-08-27 13:37:05.282411 [INFO] switch_ivr_async.c:3628 Bound B-Leg: *2 record_session::/home/saz/freeService/recordings/1008.2013-08-27-13-37-05.wav 2013-08-27 13:37:05.282411 [INFO] switch_ivr_async.c:3628 Bound B-Leg: *3 execute_extension::cf XML features 2013-08-27 13:37:05.282411 [INFO] switch_ivr_async.c:3628 Bound B-Leg: *4 execute_extension::att_xfer XML features 2013-08-27 13:37:05.282411 [NOTICE] switch_channel.c:1030 New Channel sofia/internal/sip:75icg9r5@i3ifqjalvn7j.invalid[d1fb6893-6870-4b5d-8d2c-28b038df9965] 2013-08-27 13:37:05.322412 [NOTICE] sofia.c:5898 Ring-Ready sofia/internal/sip:75icg9r5@i3ifqjalvn7j.invalid! 2013-08-27 13:37:05.322412 [WARNING] switch_channel.c:3250 rtp_secure_media invalid in this context. 2013-08-27 13:37:05.322412 [INFO] switch_ivr_originate.c:1190 Sending early media 2013-08-27 13:37:05.322412 [WARNING] switch_core_media.c:2074 NO candidate ACL defined, Defaulting to wan.auto 2013-08-27 13:37:05.322412 [NOTICE] switch_core_media.c:2107 Choose audio Candidate cid: 1 proto: udp type: host addr: 10.0.14.254:49236 2013-08-27 13:37:05.322412 [NOTICE] switch_core_media.c:2107 Choose audio Candidate cid: 2 proto: udp type: host addr: 10.0.14.254:49236 2013-08-27 13:37:05.322412 [NOTICE] switch_core_media.c:2112 Save audio Candidate cid: 1 proto: udp type: host addr: 10.0.14.253:40898 2013-08-27 13:37:05.322412 [NOTICE] switch_core_media.c:2112 Save audio Candidate cid: 2 proto: udp type: host addr: 10.0.14.253:40898 2013-08-27 13:37:05.322412 [NOTICE] switch_core_media.c:2221 setting remote audio ice addr to 10.0.14.254:49236 based on candidate 2013-08-27 13:37:05.322412 [NOTICE] switch_core_media.c:2239 setting remote rtcp audio addr to 10.0.14.254:49236 based on candidate 2013-08-27 13:37:05.322412 [INFO] switch_core_media.c:4227 Activating Audio ICE 2013-08-27 13:37:05.322412 [NOTICE] switch_rtp.c:3323 Activating RTP audio ICE: 9nidt62aGODady6Y:Hbj7Rd3CDzM2yOIn 10.0.14.254:49236 2013-08-27 13:37:05.322412 [INFO] switch_core_media.c:4270 Activating RTCP PORT 49236 2013-08-27 13:37:05.322412 [INFO] switch_core_media.c:4278 Skipping RTCP ICE (Same as RTP) 2013-08-27 13:37:05.322412 [INFO] switch_rtp.c:2711 Activating Audio Secure RTP SEND 2013-08-27 13:37:05.322412 [INFO] switch_rtp.c:2689 Activating Audio Secure RTP RECV 2013-08-27 13:37:05.322412 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1008(a)10.0.14.16! 2013-08-27 13:37:17.302425 [WARNING] switch_core_media.c:2074 NO candidate ACL defined, Defaulting to wan.auto 2013-08-27 13:37:17.302425 [NOTICE] switch_core_media.c:2107 Choose audio Candidate cid: 1 proto: udp type: host addr: 10.0.14.254:48285 2013-08-27 13:37:17.302425 [NOTICE] switch_core_media.c:2112 Save audio Candidate cid: 1 proto: udp type: host addr: 10.0.14.253:53583 2013-08-27 13:37:17.302425 [NOTICE] switch_core_media.c:2221 setting remote audio ice addr to 10.0.14.254:48285 based on candidate 2013-08-27 13:37:17.302425 [INFO] switch_core_media.c:4227 Activating Audio ICE 2013-08-27 13:37:17.302425 [NOTICE] switch_rtp.c:3323 Activating RTP audio ICE: O8IjxMVvzmi10qL4:brG3ubnHrh04oRt9 10.0.14.254:48285 2013-08-27 13:37:17.302425 [INFO] switch_core_media.c:4270 Activating RTCP PORT 48285 2013-08-27 13:37:17.302425 [INFO] switch_rtp.c:2478 Activate RTP/RTCP audio DTLS server 2013-08-27 13:37:17.302425 [NOTICE] sofia.c:6561 Channel [sofia/internal/sip:75icg9r5@i3ifqjalvn7j.invalid] has been answered 2013-08-27 13:37:17.322410 [NOTICE] switch_ivr_originate.c:3437 Channel [sofia/internal/1008(a)10.0.14.16] has been answered 2013-08-27 13:37:19.322423 [NOTICE] switch_rtp.c:1053 Auto Changing stun/rtp/dtls port from 10.0.14.254:48285 to 10.0.14.253:53583 2013-08-27 13:37:21.362423 [INFO] switch_rtp.c:2370 Changing audio DTLS state from HANDSHAKE to SETUP 2013-08-27 13:37:21.362423 [INFO] switch_rtp.c:2289 audio Fingerprint Verified. 2013-08-27 13:37:21.362423 [INFO] switch_rtp.c:2711 Activating Audio Secure RTP SEND 2013-08-27 13:37:21.362423 [INFO] switch_rtp.c:2689 Activating Audio Secure RTP RECV 2013-08-27 13:37:21.362423 [INFO] switch_rtp.c:2329 Changing audio DTLS state from SETUP to READY 2013-08-27 13:37:31.362412 [INFO] switch_rtp.c:2391 Changing audio DTLS state from READY to HANDSHAKE 2013-08-27 13:37:31.362412 [INFO] switch_rtp.c:2370 Changing audio DTLS state from HANDSHAKE to SETUP 2013-08-27 13:37:31.362412 [INFO] switch_rtp.c:2289 audio Fingerprint Verified. 2013-08-27 13:37:31.362412 [INFO] switch_rtp.c:2329 Changing audio DTLS state from SETUP to READY

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Controlling eavesdrop with event instead of DTMF, problem with event headers
by Ola Bäckström 27 Aug '13

27 Aug '13

I have successfully used eavesdrop to allow a coach to help an agent during his call to a customer. At first coach hear the agent and the customer, and after pressing 2 on the coach phone can he give advice to the agent. Later he'd press 0 to just listen etc. Now I'd like disable the dtmf control by setting a session variable eavesdrop_enable_dtmf=false and use the eavesdrop-command event mechanism outlined in this mail thread: http://lists.freeswitch.org/pipermail/freeswitch-users/2009-October/048655.…. I'm using the mod_erlang_event (could also use mod_event_socket) mechanism sendevent. The problem is that I don't create proper events that reaches the correct session. I guess I need to properly name the event and provide the correct headers. Anybody has any ideas? Currently I'm calling, using Erlang freeswitch:sendevent(Node, 'COMMAND', [{"eavesdrop-command", "2"}, {"unique-id", CoachSessionUUID}]) (where CoachSessionUUID is the coach freeswitch session identifier). Using mod_event_socket, this would correspond to sendevent COMMAND eavesdrop-command: 2 unique-id: 533da8e2-0ef0-11e3-9642-d9e368e496c6 (where I've given a particular CoachSessionUUID) I've also tried to use NOTIFY instead of COMMAND but with no success. (using Freeswitch 1.2.10, but I don't think the particular version matters) Regards /Ola

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Tuning DTMF
by Grant Bagdasarian 27 Aug '13

27 Aug '13

Hello, Setup is as following: X-Lite (Test phone) ------> SIP Proxy ------> FS (as SBC) ------> Carrier ------> End-user phone Whenever I place a call using X-Lite to the End-user phone, in this case my own mobile phone or office phone, and start sending DTMF tones from the remote telephone, the DMTF tones arrive with a crackling sound at the end of the tone. Console debug: 2013-08-23 09:00:13.813207 [DEBUG] switch_rtp.c:3829 Send start packet for [2] ts=3778476449 dur=160/160/1280 seq=5098 lw=-516490847 2013-08-23 09:00:13.833208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=320/320/1280 seq=5099 lw=161 2013-08-23 09:00:13.853208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=480/480/1280 seq=5100 lw=161 2013-08-23 09:00:13.873208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=640/640/1280 seq=5101 lw=161 2013-08-23 09:00:13.893208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=800/800/1280 seq=5102 lw=161 2013-08-23 09:00:13.913208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=960/960/1280 seq=5103 lw=161 2013-08-23 09:00:13.933208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=1120/1120/1280 seq=5104 lw=161 2013-08-23 09:00:13.953207 [DEBUG] switch_rtp.c:3729 Send end packet for [2] ts=3778476449 dur=1280/1280/1280 seq=5105 lw=1 2013-08-23 09:00:13.953207 [DEBUG] switch_rtp.c:3729 Send end packet for [2] ts=3778476449 dur=1280/1280/1280 seq=5106 lw=1 2013-08-23 09:00:13.953207 [DEBUG] switch_rtp.c:3729 Send end packet for [2] ts=3778476449 dur=1280/1280/1280 seq=5107 lw=1 2013-08-23 09:00:14.813206 [DEBUG] switch_rtp.c:5117 RTP RECV DTMF 1:1280 2013-08-23 09:00:14.813206 [DEBUG] switch_channel.c:471 RECV DTMF 1:1280 Both internal and external sip profiles have the following configuration for DTMF: <param name="liberal-dtmf" value="true"/> <param name="rfc2833-pt" value="101"/> <param name="dtmf-duration" value="2000"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="inbound-late-negotiation" value="true"/> Internal.xml (Dialplan) <extension name="internal_to_external"> <condition field="${sip_req_host}" expression="^10\.0\.0\.1$"> <action application="set" data="hangup_after_bridge=false"/> <action application="set" data="continue_on_fail=true"/> <action application="bridge" data="sofia/external/${sip_req_uri}"/> </condition> </extension> Is there a way to tune the DTMF tones? If so, which parameters do I need to add or edit? Thanks! Grant

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Pacemaker Resource Agent
by Andrew Cassidy 27 Aug '13

27 Aug '13

Hi all, I know this has come up before but I can't find a pacemaker resource agent for FreeSWITCH. I'm running a master/slave config and wondering if anyone has something they're willing to share? I know there's mod_ha_cluster, however I already have corosync/pacemaker for other services, and would like to keep things uniform. -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T <info(a)cassidywebservices.co.uk> *03300 100 960 *F<info(a)cassidywebservices.co.uk> *03300 100 961 *E <info(a)cassidywebservices.co.uk> *andrew(a)cassidywebservices.co.uk *W <info(a)cassidywebservices.co.uk> *www.cassidywebservices.co.uk

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