- FreeSWITCH-users - FreeSWITCH Mailing Lists

Any Interest in an extra Day @ Cluecon for New User Training?
by Ken Rice 19 Jul '13

19 Jul '13

Hey guys, We¹re considering adding an extra 1/2 day or so to Monday for a FreeSWITCH Crash Course. How many of you would be interested in this? K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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hi
by veera 19 Jul '13

19 Jul '13

hello, i am working on freeswitch-1.2.4 release. How can i play .PCMU files in my module using switch_ivr_play_file function thanks in advance, bye

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Re: [Freeswitch-users] G729 on virutal server (EC2)
by Colin Mason 19 Jul '13

19 Jul '13

I have g729 licenses in use on a bunch of xen based virtual machines. It works just fine. -------- Original message -------- From: Jose Suero <ms(a)mstn.com> Date: 07-18-2013 10:07 PM (GMT-05:00) To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Subject: [Freeswitch-users] G729 on virutal server (EC2) We're currently using g729 in pass-thru mode with no problems. I want to purchase g729 licenses for freeswitch, but mod_com_g729 documentation (https://wiki.freeswitch.org/wiki/Mod_com_g729) says I won't work for virtual machines, and my servers are on EC2, has anyone tried this?? Is there a way to test this before I buy?? _________________________________________________________________________ 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 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

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How to insert hash for b-leg?
by Garey Arrington 19 Jul '13

19 Jul '13

I'd like to devise a way to insert a hash for a scenario like this... 1. Receive inbound call from PSTN to IVR, caller enters an extension number or option leading to an extension or callgroup 2. Insert hash for the receiving extension or group with the domain_name of the receiving extension or group. (alternatively, is there a way to modify the eavesdrop config to accomplish the same thing?) I tried this but I have a different domain_name for calls from my carrier than for calls within a context/location (multitenant config). <extension name="postroute_global" continue="true"> <condition> <action application="hash" data="insert/${domain_name}-spymap/${caller_id_number}/${uuid}"/> <action application="hash" data="insert/${domain_name}-spymap/${destination_number}/${uuid}"/> <action application="hash" data="insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}"/> <action application="hash" data="insert/${domain_name}-last_dial/global/${uuid}"/> <action application="set" data="RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}"/> </condition> </extension> Eavesdrop extension config: <extension name="main_number_47" continue="true"> <condition field="destination_number" expression="^125(.+)$"> <action application="answer"/> <action application="eavesdrop" data="${hash(select/${domain_name}-spymap/$1)}"/> </condition> </extension>

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G729 on virutal server (EC2)
by Jose Suero 19 Jul '13

19 Jul '13

We're currently using g729 in pass-thru mode with no problems. I want to purchase g729 licenses for freeswitch, but mod_com_g729 documentation (https://wiki.freeswitch.org/wiki/Mod_com_g729) says I won't work for virtual machines, and my servers are on EC2, has anyone tried this?? Is there a way to test this before I buy??

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Call Intercept configuration confusion
by Garey Arrington 19 Jul '13

19 Jul '13

I figured this out. Was my own dumb fault for adding a * to the feature code. For the curious... do not use a * as part of the feature code for an intercept config or an eavesdrop config, it will bring you pain.

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Call Intercept configuration confusion
by Garey Arrington 19 Jul '13

19 Jul '13

I am trying to configure call intercept groups and I am having a hard time finding what is wrong with my config. When I dial the feature code to intercept *110(interceptgroup no) (in this case *110101) the dialplan doesn't match and fails. The hash is inserted successfully, I can see it in the cli via hash select/intercept_2/101. What am I missing here? Here is my user config from directory/default.xml: <user bluebox_id="3" id="1001"> <variables> <variable name="internal_caller_id_number" value="1001"/> <variable name="user_context" value="context_4"/> <variable name="force_transfer_context" value="context_4"/> <variable name="user_originated" value="true"/> <variable name="toll_allow" value="domestic"/> <variable name="accountcode" value="1001"/> <variable name="callrecord_inbound" value="1"/> <variable name="callrecord_outbound" value="1"/> <variable name="interceptgroup" value="101"/> </variables> <params> <param name="password" value="removed"/> <param name="dial-string" value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> </params> </user> <user bluebox_id="5" id="1238"> <variables> <variable name="internal_caller_id_number" value="1238"/> <variable name="user_context" value="context_4"/> <variable name="force_transfer_context" value="context_4"/> <variable name="user_originated" value="true"/> <variable name="toll_allow" value="domestic"/> <variable name="accountcode" value="1238"/> <variable name="callrecord_inbound" value="1"/> <variable name="callrecord_outbound" value="1"/> <variable name="interceptgroup" value="101"/> </variables> <params> <param name="password" value="removed"/> <param name="dial-string" value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> </params> </user> Here is my dialplan config from dialplan.xml: <extension name="main_number_43" continue="true"> <condition field="destination_number" expression="^1238$"> <action application="hash" data="insert/intercept_2/101/${uuid}"/> <action application="set" bluebox="settingTimeout" data="call_timeout=30"/> <action application="set" bluebox="settingRing" data="ringback=${us-ring}"/> <action application="set" bluebox="settingRingXfer" data="transfer_ringback=${us-ring}"/> <action application="export" bluebox="sipCalleeIdName" data="sip_callee_id_name=linksys"/> <action application="export" bluebox="sipCalleeIdNumber" data="sip_callee_id_number=1238"/> <action application="bridge" data="user/1238@$${location_4}"/> <action application="hangup"/> </condition> </extension> <extension name="main_number_45" continue="true"> <condition field="${regex(m:/${destination_number}/^\*110([0-9]+)$/$1)}" expression="^${interceptgroup}$"/> <condition field="destination_number" expression="^\*110([0-9]+)$"> <action application="answer"/> <action application="set" data="intercept_unanswered_only=true"/> <action application="intercept" data="${hash(select/intercept_2/$1)}"/> <action application="sleep" data="2000"/> <action application="hangup"/> </condition> </extension>

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200 OK has fatal syntax errors
by Yuriy Nasida 18 Jul '13

18 Jul '13

Hi guys, My FS ignore 200 OK message from one of my carriers. What can be the reson ?I heard about issue with Session-Expires: -1 But.... My 200 OK message hasn't such header. Please advice U 2013/07/18 11:29:21.673774 111.111.111.111:5060 -> 222.222.222.222:5060SIP/2.0 200 OK.Via: SIP/2.0/UDP 222.222.222.222;received=222.222.222.222;rport=5060;branch=z9hG4bKm9vjBFHyHycpc.From: <sip:78125555555@222.222.222.222>;tag=gNX1y2B8vXa1D.To: <sip:12127777777@ingress.dal.shangovoip.com>;tag=sbcsipuas_1_C51361_20130718112919309_b12sb02.Call-ID: a7727999-6a61-1231-ed92-90b11c5878d7.Record-Route: <sip:333.333.333.333;lr;ftag=gNX1y2B8vXa1D>.Record-Route: <sip:444.444.444.444;lr;ftag=gNX1y2B8vXa1D;did=f18.5b69f806>.Record-Route: <sip:555.555.555.555;lr;ftag=gNX1y2B8vXa1D>.Record-Route: <sip:111.111.111.111;lr;ftag=gNX1y2B8vXa1D;did=f18.b4617df.Contact: <sip:777.777.777.777:5060>.CSeq: 46744615 INVITE.Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,NOTIFY,INFO.Server: sbc_5.Content-Type: application/sdp.Content-Length: 236..v=0.o=SBCSIPUAS 900213980 1 IN IP4 888.888.888.888.s=SBCSIPUAS SIP STACK v1.0.c=IN IP4 888.888.888.888.t=0 0.m=audio 21526 RTP/AVP 8 101.a=rtpmap:8 PCMA/8000.a=rtpmap:101 telephone-event/8000.a=fmtp:101 0-15.a=sendrecv.a=maxptime:20. nta.c:3212 agent_recv_response() nta: received 200 OK for INVITE (46747876)nta.c:3239 agent_recv_response() nta: 200 OK has fatal syntax errors

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Event_handler using switch_event_bind and private user data issue
by Jon Lederman 18 Jul '13

18 Jul '13

Hi, I'm trying to set up an event handler that will send some user data to a callback function. I am using: pvt = (switch_audio_weaver_t*) switch_core_session_alloc(session, sizeof(*pvt)); if (switch_event_bind(modname, SWITCH_EVENT_CUSTOM, AUDIOWEAVER_SEND_MESSAGE, audioweaver_message_event_handler, pvt) != SWITCH_STATUS_SUCCESS) { switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Couldn't bind!\n"); /* indicate that the module should continue to be loaded */ return SWITCH_STATUS_FALSE; } This works fine and my event handler is called. But, I am unclear how to get the event_user_data. My event handler is: static void audioweaver_message_event_handler(switch_event_t *event) { switch_audio_weaver_t *pvt = (switch_audio_weaver_t *) event->event_user_data; switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "AudioWeaver Message Event Handler Reached!"); switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, switch_event_get_body(event)); std::string reply; int ret = pvt->aw_talker->SendMessage("set_value,Sine1.freq,110.0", reply); I am using mod_event_socket, which uses switch_event_fire. I note in switch_event.h, the following: /*! \brief Fire an event filling in most of the arguements with obvious values \param event the event to send (will be nulled on success) \return SWITCH_STATUS_SUCCESS if the operation was successful \note the body supplied by this function will supersede an existing body the event may have */ #define switch_event_fire(event) switch_event_fire_detailed(__FILE__, (const char * )__SWITCH_FUNC__, __LINE__, event, NULL) /*! How would I use switch_event_fire_detailed with mod_event_socket, which I believe is necessary in order to send the user_data? } However, the pointer doesn't seem to point to the private data. Am I setting up my event handler correctly to get this data? Thanks. -Jon

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Error Creating SIP UA
by mehroz 18 Jul '13

18 Jul '13

Hi, I have a fresh install of FS stable verison 1.3.17. When i enable <X-PRE-PROCESS cmd="set" data="internal_ssl_enable=true"/> in vars.xml I get, Error Creating SIP UA for profile: internal 2013-06-12 17:55:54.562423 [ERR] sofia.c:2431 Error Creating SIP UA for profile: internal-ipv6 (sip:mod_sofia@[::1]:5060) The likely causes for this are: 1) Another application is already listening on the specified address. 2) The IP the profile is attempting to bind to is not local to this system. 2013-06-12 17:55:54.562543 [ERR] sofia.c:2431 Error Creating SIP UA for profile: internal (sip:mod_sofia@198.84.61.52:5060) The likely causes for this are: 1) Another application is already listening on the specified address. 2) The IP the profile is attempting to bind to is not local to this system. but OK otherwise in case <X-PRE-PROCESS cmd="set" data="internal_ssl_enable=false"/> There is no instant of FS previously or 5061 previously occupied. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Error-Creating-SIP-UA-tp75917… Sent from the freeswitch-users mailing list archive at Nabble.com.

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