November 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

FreeSWITCH TLS not able to receive calls
by Ahmed Habiba 01 Dec '14

01 Dec '14

Dears, I’ve configured FreeSWITCH with the below version with TLS/SRTPas per the recommendation in page “https://wiki.freeswitch.org/wiki/SIP_TLS” and it was strait forward, and I was able to connect and make make calls using zoiper, but I was not able to receive any calls after enabling the TLS/SRTP. "FreeSWITCH Version 1.4.13+git~20141103T195300Z~b942d0faa8~64bit (git b942d0f 2014-11-03 19:53:00Z 64bit)” Your kind feedback will be appreciate. Thanks, Ahmed Habiba.

2 1

multi tenet user directory management best practice
by ik 01 Dec '14

01 Dec '14

Hello, This question raises a lot, but no simple answer, so I'm trying to ask it differently. Let's say I have 5 FS servers and Opensips/Kamailio as a load balancer. I'm afraid that creating it using xml files is not dynamic enough, and requires duplications. mod_xml_rpc does not store that information on each update of the server itself, and there is no API for providing database management that I could find. So what is the best practice regarding of creating a user directory that all of the PBX will be able to handle ? Thanks Ido

2 1

USB GSM Dongle with FreeSwitch for voice calls
by Agha Ali Raza 01 Dec '14

01 Dec '14

Hi All, I want to use a USB GSM dongle (unlocked Huawei E-173L, voice enabled) with FreeSwitch for voice calls. Is mod_gsmopen reliable and does anyone have experience using it? Are there other options besides mod_gsmopen? I am also having trouble building mod_gsmopen on Windows. I would appreciate pointers to any available resources or reading. Thanks, Ali

7 22

mod_xml_curl dialplan not requested
by Zoltán Szabó 30 Nov '14

30 Nov '14

Hi, I have a multi tenant multi domain setup. For some reasons my FS won't try to request the dialplan configuration. *xml_curl:* <configuration name="xml_curl.conf" description="cURL XML Gateway"> <bindings> <binding name="dialplan"> <param name="gateway-url" value=" http://192.168.0.15:8080/ccc-provider/GetDialplan" bindings="dialplan"/> </binding> <binding name="directory"> <param name="gateway-url" value=" http://192.168.0.15:8080/ccc-provider/GetDirectory" bindings="directory"/> </binding> <binding name="configuration"> <param name="gateway-url" value=" http://192.168.0.15:8080/ccc-provider/GetConfiguration" bindings="configuration"/> </binding> </bindings> </configuration> My sofia conf that I send through xml_curl upon request: <document type="freeswitch/xml"> <section name="configuration"> <configuration name='sofia.conf' description='sofia endpoint'> <global_settings> <param name="log-level" value="9"/> <param name="debug" value="9"/> <param name="auto-restart" value="true"/> <param name="debug-presence" value="0"/> <param name='accept-blind-reg' value='false'/> <param name="sip-trace" value="no"/> <param name="sip-capture" value="no"/> <param name="auth-calls" value="false"/> <param name="dtmf-type" value="rfc2833"/> </global_settings> <profiles> <profile name='MainCCSIP'> <domains> <domain name="d.1.c.domain.com" alias="false" parse="false"/> <domain name="d.2.c.domain.com" alias="false" parse="false"/> </domains> <settings> <param name="odbc-dsn" value=""/> <param name="manage-presence" value="true"/> <param name="dbname" value="shere_presence"/> <param name="presence-hosts" value="_DISABLED_"/> <param name="send-presence-on-register" value="true"/> <param name="delete-subs-on-register" value="true"/> <param name="caller-id-type" value="rpid"/> <param name="auto-jitterbuffer-msec" value="1000"/> <param name="user-agent-string" value="pbx"/> <param name='dialplan' value='XML'/> <param name="ext-rtp-ip" value="auto"/> <param name="ext-sip-ip" value="auto"/> <param name='rtp-ip' value='auto'/> <param name='sip-ip' value='auto'/> <param name='sip-port' value='5060'/> <param name='nonce-ttl' value=''/> <param name='rtp-timer-name' value='soft'/> <param name='codec-prefs' value='"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name='rfc2833-pt' value='101'/> <param name='dtmf-duration' value='100'/> <param name="dtmf-type" value=""/> <param name='codec-ms' value='20'/> <param name='accept-blind-reg' value='false'/> <param name='auth-calls' value='false'/> <param name="enable-100rel" value="true"/> <param name="multiple-registrations" value="contact"/> <param name="minimum-session-expires" value="120"/> <param name="mailer-app" value="/usr/local/freeswitch/script/emailvm.py"/> <param name="mailer-app-args" value="-i -d -f voicemail(a)domain.org -t"/> <param name='accept-blind-reg' value='false'/> <param name='disable-register' value='false'/> <param name='log-auth-failures' value='true'/> <param name='auth-all-packets' value='false'/> <param name="mailer-app" value="/usr/local/freeswitch/scripts/emailvm.py"/> <param name="mailer-app-args" value="-i -d -f voicemail(a)domain.org -t"/> </settings> <gateways> <gateway name='gateway-1.d.1.c.domain.com'> <variables> <variable name="inbound_var_name" data="inbound_call=true" direction="inbound"/> <variable name="outbound_var_name" data="outbound_call=true" direction="outbound"/> <variable name="both_var_name" data="gw=gwname"/> </variables> <param name='username' value='username'/> <param name='password' value='password'/> <param name="realm" value="sip.voip.com"/> <param name="expire-seconds" value="600"/> <param name='register' value='true'/> <param name="register-transport" value="udp"/> <param name='retry-seconds' value='30'/> <param name="ping" value="60"/> <param name='register' value='true'/> </gateway> </gateways> </profile> </profiles> </configuration> </section> </document> My user directory looks like this: *User 1000 at domain 1:* <document type="freeswitch/xml"> <section name="directory"> <domain name='d.1.c.domain.com'> <params> <param name="dial-string" value="{presence_id=${dialed_user}@ ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> </params> <variables> <variable name="record_stereo" value="true"/> <variable name="transfer_fallback_extension" value="operator"/> <variable name="user_context" value="d.1.c.domain.com"/> </variables> <groups> <group name="d.1.c.domain.com"> <users> <user id='1000' mailbox='1000'> <params> <param name='password' value='123456'/> <param name='vm-password' value='123456'/> <param name="vm-keep-local-after-email" value="true"/> <param name="vm-message-ext" value="mp3"/> </params> <variables> <variable name="outbound_caller_id_name" value="Unknown"/> <variable name="outbound_caller_id_number" value="0000000000"/> <variable name="internal_caller_id_name" value="Unknown"/> <variable name="user_originated" value="true"/> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="1000"/> <variable name='sip-force-contact' value='nat-connectile-dysfunction'/> <variable name="record_stereo" value="false"/> <variable name="sip-allow-multiple-registrations" value="false"/> <variable name="default_gateway" value="something"/> <variable name="presence_id" value=" 1000(a)d.1.c.domain.com"/> <variable name="transfer_fallback_extension" value="operator"/> </variables> </user> </users> </group> </groups> </domain> </section> </document> *User 1000 at domain 2:* <document type="freeswitch/xml"> <section name="directory"> <domain name='d.2.c.domain.com'> <params> <param name="dial-string" value="{presence_id=${dialed_user}@ ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> </params> <variables> <variable name="record_stereo" value="true"/> <variable name="transfer_fallback_extension" value="operator"/> <variable name="user_context" value="d.2.c.domain.com"/> </variables> <groups> <group name="d.2.c.domain.com"> <users> <user id='1000' mailbox='1000'> <params> <param name='password' value='654321'/> <param name='vm-password' value='123456'/> <param name="vm-keep-local-after-email" value="true"/> <param name="vm-message-ext" value="mp3"/> </params> <variables> <variable name="outbound_caller_id_name" value="Unknown"/> <variable name="outbound_caller_id_number" value="0000000000"/> <variable name="internal_caller_id_name" value="Unknown"/> <variable name="user_originated" value="true"/> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="0"/> <variable name='sip-force-contact' value='nat-connectile-dysfunction'/> <variable name="record_stereo" value="false"/> <variable name="sip-allow-multiple-registrations" value="false"/> <variable name="default_gateway" value="something"/> <variable name="presence_id" value=" 1000(a)d.2.c.domain.com"/> <variable name="transfer_fallback_extension" value="operator"/> </variables> </user> </users> </group> </groups> </domain> </section> </document> So far the registration for both users in both domains are working and I can register with sip clients. But if I start to dial something, FS won't request for the dialplan. Here is what I can see in the cli trace: freeswitch@dev> tport.c:2749 tport_wakeup_pri() tport_wakeup_pri(0xb92ee0): events IN tport.c:2864 tport_recv_event() tport_recv_event(0xb92ee0) tport.c:3205 tport_recv_iovec() tport_recv_iovec(0xb92ee0) msg 0xd47840 from (udp/192.168.0.132:5060) has 972 bytes, veclen = 1 tport.c:3023 tport_deliver() tport_deliver(0xb92ee0): msg 0xd47840 (972 bytes) from udp/192.168.0.15:5060/sip next=(nil) nta.c:2880 agent_recv_request() nta: received INVITE sip:123456789@d.2.c.domain.com SIP/2.0 (CSeq 20) nta.c:3085 agent_recv_request() nta: INVITE (20) going to a default leg nta.c:1348 set_timeout() nta: timer shortened to 200 ms nua_server.c:102 nua_stack_process_request() nua: nua_stack_process_request: entering nua_stack.c:899 nh_create() nua: nh_create: entering nua_common.c:108 nh_create_handle() nua: nh_create_handle: entering nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering soa.c:280 soa_clone() soa_clone(static::0xb8f480, 0xb7cea0, 0xb85400) called soa.c:403 soa_set_params() soa_set_params(static::0xa98310, ...) called nta.c:4415 nta_leg_tcreate() nta_leg_tcreate(0xdcaf90) soa.c:1302 soa_init_offer_answer() soa_init_offer_answer(static::0xa98310) called soa.c:1171 soa_set_remote_sdp() soa_set_remote_sdp(static::0xa98310, (nil), 0xb54b3b, 433) called nua_dialog.c:338 nua_dialog_usage_add() nua(0xb85400): adding session usage tport.c:3257 tport_tsend() tport_tsend(0xb92ee0) tpn = UDP/192.168.0.15:5060 tport.c:4046 tport_resolve() tport_resolve addrinfo = 192.168.0.15:5060 tport.c:4680 tport_by_addrinfo() tport_by_addrinfo(0xb92ee0): not found by name UDP/192.168.0.15:5060 tport.c:3492 tport_send_msg() tport_vsend returned 307 nta.c:6789 incoming_reply() nta: sent 100 Trying for INVITE (20) nua_session.c:4137 signal_call_state_change() nua(0xb85400): call state changed: init -> received, received offer soa.c:1098 soa_get_remote_sdp() soa_get_remote_sdp(static::0xa98310, [0x7fe32eca6608], [0x7fe32eca6610], [(nil)]) called nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering 2014-11-29 18:31:52.853532 [NOTICE] switch_channel.c:1055 New Channel sofia/MainCCSIP/1000(a)d.2.c.domain.com [fd50511c-77f5-11e4-b25e-c755bd10628e] nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 2014-11-29 18:31:52.853532 [NOTICE] sofia.c:6946 Hangup sofia/MainCCSIP/ 1000(a)d.2.c.domain.com [CS_NEW] [INCOMPATIBLE_DESTINATION] nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:879 nua_respond() nua: nua_respond: entering nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering soa.c:403 soa_set_params() soa_set_params(static::0xa98310, ...) called nua_session.c:2318 nua_invite_server_respond() nua: nua_invite_server_respond: entering soa.c:1214 soa_clear_remote_sdp() soa_clear_remote_sdp(static::0xa98310) called tport.c:3257 tport_tsend() tport_tsend(0xb92ee0) tpn = UDP/192.168.0.15:5060 tport.c:4046 tport_resolve() tport_resolve addrinfo = 192.168.0.15:5060 tport.c:4680 tport_by_addrinfo() tport_by_addrinfo(0xb92ee0): not found by name UDP/192.168.0.15:5060 tport.c:3492 tport_send_msg() tport_vsend returned 892 nta.c:6789 incoming_reply() nta: sent 488 Not Acceptable Here for INVITE (20) nua_dialog.c:397 nua_dialog_usage_remove_at() nua(0xb85400): removing session usage nua_session.c:4137 signal_call_state_change() nua(0xb85400): call state changed: received -> terminated soa.c:356 soa_destroy() soa_destroy(static::0xa98310) called nta.c:4468 nta_leg_destroy() nta_leg_destroy(0xdcaf90) nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua_stack.c:529 nua_signal() nua(0xb85400): sent signal r_respond tport.c:2749 tport_wakeup_pri() tport_wakeup_pri(0xb92ee0): events IN tport.c:2864 tport_recv_event() tport_recv_event(0xb92ee0) tport.c:3205 tport_recv_iovec() tport_recv_iovec(0xb92ee0) msg 0xaa4690 from (udp/192.168.0.132:5060) has 324 bytes, veclen = 1 tport.c:3023 tport_deliver() tport_deliver(0xb92ee0): msg 0xaa4690 (324 bytes) from udp/192.168.0.15:5060/sip next=(nil) nta.c:2880 agent_recv_request() nta: received ACK sip:123456789@d.2.c.domain.com SIP/2.0 (CSeq 20) nta.c:3019 agent_recv_request() nta: ACK (20) is going to INVITE (20) 2014-11-29 18:31:52.853532 [NOTICE] switch_core_session.c:1633 Session 4 (sofia/MainCCSIP/1000(a)d.2.c.domain.com) Ended 2014-11-29 18:31:52.853532 [NOTICE] switch_core_session.c:1637 Close Channel sofia/MainCCSIP/1000(a)d.2.c.domain.com [CS_DESTROY] nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering nta.c:4468 nta_leg_destroy() nta_leg_destroy((nil)) nua_stack.c:529 nua_signal() nua(0xb85400): sent signal r_destroy nta.c:1296 agent_timer() nta: timer set next to 1976 ms The only request is coming from time to time is this: hostname = [dev] section = [directory] tag_name = [domain] key_name = [name] key_value = [d.2.c.domain.com] Event-Name = [REQUEST_PARAMS] Core-UUID = [d1af88e4-77f4-11e4-b1f2-c755bd10628e] FreeSWITCH-Hostname = [dev] FreeSWITCH-Switchname = [dev] FreeSWITCH-IPv4 = [192.168.0.132] FreeSWITCH-IPv6 = [::1] Event-Date-Local = [2014-11-29 18:35:05] Event-Date-GMT = [Sat, 29 Nov 2014 18:35:05 GMT] Event-Date-Timestamp = [1417286105312271] Event-Calling-File = [sofia_reg.c] Event-Calling-Function = [sofia_reg_parse_auth] Event-Calling-Line-Number = [2739] Event-Sequence = [898] action = [sip_auth] sip_profile = [MainCCSIP] sip_user_agent = [qutecom/rev-6653cd8f2afd-trunk] sip_auth_username = [1000] sip_auth_realm = [d.2.c.domain.com] sip_auth_nonce = [6ff7849c-77f6-11e4-b27c-c755bd10628e] sip_auth_uri = [sip:d.2.c.domain.com] sip_contact_user = [1000] sip_contact_host = [192.168.0.15] sip_to_user = [1000] sip_to_host = [d.2.c.domain.com] sip_via_protocol = [udp] sip_from_user = [1000] sip_from_host = [d.2.c.domain.com] sip_call_id = [1172541516(a)192.168.0.15] sip_request_host = [d.2.c.domain.com] sip_auth_response = [6836f04faa9f464b8d78f11e88d7ee5d] sip_auth_method = [REGISTER] client_port = [5060] key = [id] user = [1000] domain = [d.2.c.domain.com] ip = [192.168.0.15] Any thoughts why is the dialplan not requested? Thanks, Zoltan

2 2

mod_xml_curl sofia domain gateways
by Zoltán Szabó 28 Nov '14

28 Nov '14

Hi, What I would like to do is, I need to handle multiple domains. Every domain will have their own gateways and these domains will use only one sip profile. So first when I start FS I got a request: *REQUEST 1:* hostname = [dev] section = [configuration] tag_name = [configuration] key_name = [name] key_value = [sofia.conf] Event-Name = [REQUEST_PARAMS] Core-UUID = [d03d1242-7722-11e4-b04c-0f0151b60864] FreeSWITCH-Hostname = [dev] FreeSWITCH-Switchname = [dev] FreeSWITCH-IPv4 = [192.168.0.132] FreeSWITCH-IPv6 = [::1] Event-Date-Local = [2014-11-28 17:20:13] Event-Date-GMT = [Fri, 28 Nov 2014 17:20:13 GMT] Event-Date-Timestamp = [1417195213442735] Event-Calling-File = [sofia.c] Event-Calling-Function = [config_sofia] Event-Calling-Line-Number = [3948] Event-Sequence = [18] So this is requesting the sofia config, so I send back that xml: *REPLY 1:* <document type="freeswitch/xml" xmlns:py="http://genshi.edgewall.org/" xmlns:xsl="http://www.w3.org/1999/XSL/Transform" xmlns:xsd="http://www.w3.org/2001/XMLSchema" xmlns="http://www.w3.org/1999/xhtml"> <section name="configuration"> <configuration name='sofia.conf' description='sofia endpoint'> <global_settings> <param name="log-level" value="9"/> <param name="auto-restart" value="true"/> <param name="debug-presence" value="0"/> <param name='accept-blind-reg' value='false'/> <param name="sip-trace" value="no"/> <param name="sip-capture" value="no"/> <param name="auth-calls" value="true"/> <param name="dtmf-type" value="rfc2833"/> </global_settings> <profiles> <profile name='MainCCSIP'> <domains> <domain name="domain1.example.com" alias="false" parse="true"/> </domains> <settings> <param name="odbc-dsn" value=""/> lot of parameters... </settings> <gateways> </gateways> </profile> </profiles> </configuration> </section> </document> Than it realizes I have a profile specified here, so it requesting something similar, with the profile name key: *REQUEST 2:* hostname = [dev] section = [configuration] tag_name = [configuration] key_name = [name] key_value = [sofia.conf] Event-Name = [REQUEST_PARAMS] Core-UUID = [199eb63c-7716-11e4-a68f-272cf7c86620] FreeSWITCH-Hostname = [dev] FreeSWITCH-Switchname = [dev] FreeSWITCH-IPv4 = [192.168.0.132] FreeSWITCH-IPv6 = [::1] Event-Date-Local = [2014-11-28 16:46:53] Event-Date-GMT = [Fri, 28 Nov 2014 16:46:53 GMT] Event-Date-Timestamp = [1417193213090586] Event-Calling-File = [sofia.c] Event-Calling-Function = [config_sofia] Event-Calling-Line-Number = [3948] Event-Sequence = [934] profile = [MainCCSIP] I am not sure what does it want now but I will send back the same XML, containing my domains (1 for now), the profile settings and no gateways assigned to this profile: *SAME REPLY1:* <document type="freeswitch/xml" xmlns:py="http://genshi.edgewall.org/" xmlns:xsl="http://www.w3.org/1999/XSL/Transform" xmlns:xsd="http://www.w3.org/2001/XMLSchema" xmlns="http://www.w3.org/1999/xhtml"> <section name="configuration"> <configuration name='sofia.conf' description='sofia endpoint'> <global_settings> <param name="log-level" value="9"/> <param name="auto-restart" value="true"/> <param name="debug-presence" value="0"/> <param name='accept-blind-reg' value='false'/> <param name="sip-trace" value="no"/> <param name="sip-capture" value="no"/> <param name="auth-calls" value="true"/> <param name="dtmf-type" value="rfc2833"/> </global_settings> <profiles> <profile name='MainCCSIP'> <domains> <domain name="domain1.example.com" alias="false" parse="true"/> </domains> <settings> <param name="odbc-dsn" value=""/> lot of parameters... </settings> <gateways> </gateways> </profile> </profiles> </configuration> </section> </document> Please note I excluded all the parameters as there are a lot in the original response. Than I got a new request: *REQUEST 3:* hostname = [dev] section = [configuration] tag_name = [configuration] key_name = [name] key_value = [sofia.conf] Event-Name = [REQUEST_PARAMS] Core-UUID = [199eb63c-7716-11e4-a68f-272cf7c86620] FreeSWITCH-Hostname = [dev] FreeSWITCH-Switchname = [dev] FreeSWITCH-IPv4 = [192.168.0.132] FreeSWITCH-IPv6 = [::1] Event-Date-Local = [2014-11-28 16:46:53] Event-Date-GMT = [Fri, 28 Nov 2014 16:46:53 GMT] Event-Date-Timestamp = [1417193213190626] Event-Calling-File = [sofia.c] Event-Calling-Function = [launch_sofia_worker_thread] Event-Calling-Line-Number = [2658] Event-Sequence = [938] profile = [MainCCSIP] I am wondering why is this request? The only difference is the calling function. Anyway I am sending back the same as above. Then I got an other request: *REQUEST 4:* hostname = [dev] section = [directory] tag_name = [domain] key_name = [name] key_value = [domain1.example.com] Event-Name = [REQUEST_PARAMS] Core-UUID = [199eb63c-7716-11e4-a68f-272cf7c86620] FreeSWITCH-Hostname = [dev] FreeSWITCH-Switchname = [dev] FreeSWITCH-IPv4 = [192.168.0.132] FreeSWITCH-IPv6 = [::1] Event-Date-Local = [2014-11-28 16:46:53] Event-Date-GMT = [Fri, 28 Nov 2014 16:46:53 GMT] Event-Date-Timestamp = [1417193213330579] Event-Calling-File = [sofia.c] Event-Calling-Function = [launch_sofia_worker_thread] Event-Calling-Line-Number = [2676] Event-Sequence = [939] purpose = [gateways] profile = [MainCCSIP] This seems fine as I have set parse=true in the domain parameters so now it is requesting the gateways for my specific domain1.example.com domain. Now I am sending back the following: <document type="freeswitch/xml" xmlns:xsl="http://www.w3.org/1999/XSL/Transform" xmlns:xsd="http://www.w3.org/2001/XMLSchema" xmlns="http://www.w3.org/1999/xhtml"> <section name="directory"> <domain name='domain1.example.com'></domain> <gateways> <gateway name='TestGW'> <variables> <variable name="inbound_var_name" data="inbound_call=true" direction="inbound"/> <variable name="outbound_var_name" data="outbound_call=true" direction="outbound"/> <variable name="both_var_name" data="gw=TestGW"/> </variables> <param name='username' value='testuser'/> <param name='password' value='testpw'/> <param name="realm" value="sip.testgw.com"/> <param name="expire-seconds" value="600"/> <param name='register' value='true'/> <param name="register-transport" value="udp"/> <param name='retry-seconds' value='30'/> <param name="ping" value="60"/> <param name='register' value='true'/> </gateway> </gateways> </section> </document> Then I can not see anything in the fs cli related to this gateway. It does not try to connect to it, I can not list it, and also can not see any errors related to this even if I set every trace and debug level to the maximum. I also tried to enclose the gateways node into the domain like: <domain name='domain1.example.com'> <gateways> All gateway config here... </gateways> </domain> But I can not see anything which indicates that this is correct/incorrect or what. Is there any problem with this response or am I missing something? I have not send back any other config that FS requests at this stage, just these. Many thanks, Zoltan

1 0

High Availability Cluster Module for FreeSWITCH
by Борисов, Дмитрий 28 Nov '14

28 Nov '14

Hi, All! In what state this module is now? -- С уважением, Борисов Дмитрий ответственный за разработку VoIP решения системный администратор отдела облачных технологий

1 0

FreeSWITCH Friday FreeForAll Reminder!
by Ken Rice 28 Nov '14

28 Nov '14

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888@conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon

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while loading mod_xml_ldap module getting error
by Shisheer Teli 28 Nov '14

28 Nov '14

Dear, while loading mod_xml_ldap module i found following error 2014-11-28 14:55:18.308438 [CRIT] switch_loadable_module.c:1391 Error Loading module /usr/local/freeswitch/mod/mod_xml_ldap.so */usr/local/freeswitch/mod/mod_xml_ldap.so: undefined symbol: lutil_sasl_interact* can help me to troubleshoot -- Regards, Shisheer T

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Freeswitch and virtual FAX
by Dragan Saraginov 27 Nov '14

27 Nov '14

Hi guys, I hope someone can help me on the issue that I am facing. I am planing to implement a virtual FAX machine with freeswitch. My main concern is, will it be possible to do that without any type of card, analog, digital, ATA gateway or IAXmodem, but using the t38 support? Best regards, Dragan

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how to play a file to recipient
by kaleem rehman 27 Nov '14

27 Nov '14

Hi All, I want to play a message to recipient to advise them what line call came in, in Avaya world this is known as Whisper / VDN of Origin Announcement. what I want to achieve is when call hits the Gateway its setup to route call to an extension, when extension answers the call they hear a wav file or a message via (Freeswitch SAY command) saying "customer service" or "Return" etc... thanks in advance, kaleem

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