July 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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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grandstream Gateway installation question
by Sherif Omran 18 Jul '13

18 Jul '13

Dear all, I have a grandstream GXW4104 gateway (voip-landline gateway) and i have problems in installing it. The gateway (GW) registers as an extension (1005) to my server. I need to send landline calls to extension 1005 and change the SIP header (effective caller ID) to the new landline number (ex: 0352209xxx) could you help me with the dial plan? thank you regards, Sherif

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External SIP Trunk IP Authentication
by bjamm90 18 Jul '13

18 Jul '13

Hello all, I need to set up a freeswitch trunk to a termination provider that authenticates via IP. Previously I was set up through Skype so the set up was pretty easy because the wiki has plenty of information on setting up Skype. Alcazar networks is the termination provider I am using. The only configuration I could find on the wiki is for sonovoip so I assume my configuration would be close to the same. I assume it would look something like this: <include> <gateway name="alcazar networks"> </gateway> </include> Any advice would be greatly appreciated -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/External-SIP-Trunk-IP-Authent… Sent from the freeswitch-users mailing list archive at Nabble.com.

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PRI Help
by Blake Priddy 17 Jul '13

17 Jul '13

Does anyone here have any insight as to why when placing a call out to a 3G ATT cell phone caller ID shows unknown But all other networks it works just fine.. LOL -- * * *Blakelund Priddy* Network & Systems Engineer Bryant Public School District Bryant, Arkansas 72022 http://www.bryantschools.org p 501-653-5038 f 501-847-5656

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FreeSWITCH Community Conference Call Today
by Michael Collins 17 Jul '13

17 Jul '13

Hello all, Don't forget to join us today for our weekly conference call. The agenda page is here: https://wiki.freeswitch.org/wiki/FS_weekly_2013_07_17 We have a few announcements and will be doing a community scrum. Please bring your subjects for discussion! Thanks, -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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Maximum number of conferences
by Young, Christopher 17 Jul '13

17 Jul '13

Hi all, Is there a recommended maximum number of simultaneous conferences that FreeSWITCH can handle or is it simply determined by the specification of the server on which it's running? Many thanks, Chris Chris Young Software Engineer direct: +44 (0) 118 943 9249 web: www.enghouseinteractive.com<http://www.enghouseinteractive.com/> | email: cyoung(a)enghouse.com<mailto:cyoung@enghouse.com> A Division of Enghouse Systems Ltd.<http://www.enghouse.com/> (TSX: ESL)

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Leaking stream handle!
by Lloyd Aloysius 17 Jul '13

17 Jul '13

Hello I saw the following when play MOH. Leaking stream handle! in read and switch hang and start to work in 45 secs. But call never disconnect. Version - FreeSWITCH Version 1.2.11+git~20130714T023816Z~f90e828bd5 (git f90e828 2013-07-14 02:38:16Z) 2013-07-15 21:48:32.081433 [DEBUG] sofia.c:5719 Channel sofia/sipinterface_1/lloyd.aloysius(a)mydomain.com entering state [ready][200] 2013-07-15 21:48:32.081433 [DEBUG] sofia.c:5727 Duplicate SDP v=0 o=- 1167618390 1167618392 IN IP4 192.168.240.65 s=Polycom IP Phone c=IN IP4 192.168.240.65 t=0 0 m=audio 2236 RTP/AVP 9 0 8 18 127 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000 2013-07-15 21:48:32.081433 [DEBUG] sofia.c:5719 Channel sofia/sipinterface_1/lloyd.aloysius(a)mydomain.com entering state [calling][0] 2013-07-15 21:48:32.081433 [DEBUG] sofia.c:5719 Channel sofia/sipinterface_1/lloyd.aloysius(a)mydomain.com entering state [ready][200] 2013-07-15 21:48:32.081433 [DEBUG] sofia.c:5727 Duplicate SDP v=0 o=- 1167618390 1167618392 IN IP4 192.168.240.65 s=Polycom IP Phone c=IN IP4 192.168.240.65 t=0 0 m=audio 2236 RTP/AVP 9 0 8 18 127 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000 2013-07-15 21:48:32.081433 [DEBUG] sofia_reg.c:2084 Changing expire time to 180 by request of proxy sip:sip02.unlimitel.ca 2013-07-15 21:48:32.081433 [DEBUG] sofia_reg.c:2084 Changing expire time to 300 by request of proxy sip:sip05.unlimitel.ca 2013-07-15 21:48:32.081433 [DEBUG] sofia_reg.c:2084 Changing expire time to 300 by request of proxy sip:sip05.unlimitel.ca 2013-07-15 21:48:32.081433 [DEBUG] sofia_reg.c:2084 Changing expire time to 300 by request of proxy sip:sip.unlimitel.ca 2013-07-15 21:48:32.081433 [WARNING] sofia_reg.c:1533 SIP auth challenge (REGISTER) on sofia profile 'sipinterface_1' for [ lloyd.aloysius(a)mydomain.com] from ip 99.247.42.206 2013-07-15 21:48:56.731449 [DEBUG] sofia_reg.c:2084 Changing expire time to 300 by request of proxy sip:sip05.unlimitel.ca 2013-07-15 21:48:57.741431 [DEBUG] sofia_reg.c:2084 Changing expire time to 180 by request of proxy sip:sip02.unlimitel.ca 2013-07-15 21:49:07.301449 [DEBUG] switch_core_session.c:999 Send signal sofia/sipinterface_1/lloyd.aloysius(a)mydomain.com [BREAK] 2013-07-15 21:49:07.311444 [DEBUG] sofia.c:5719 Channel sofia/sipinterface_1/lloyd.aloysius(a)mydomain.com entering state [calling][0] 2013-07-15 21:49:07.361445 [DEBUG] switch_core_session.c:999 Send signal sofia/sipinterface_1/lloyd.aloysius(a)mydomain.com [BREAK] 2013-07-15 21:49:07.361445 [DEBUG] switch_core_session.c:999 Send signal sofia/sipinterface_1/lloyd.aloysius(a)mydomain.com [BREAK] 2013-07-15 21:49:07.361445 [DEBUG] switch_core_session.c:999 Send signal sofia/sipinterface_1/lloyd.aloysius(a)mydomain.com [BREAK] 2013-07-15 21:49:08.831430 [CRIT] mod_local_stream.c:322 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1246] 2013-07-15 21:49:16.551430 [CRIT] mod_local_stream.c:322 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1246] Thanks Lloyd

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