October 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

How to Register using radiusclient digest authentication
by Masood Ahmed 11 Oct '11

11 Oct '11

Dear , I had successfully installed and configured radiusclient but now as I am newbie I really do not know that what next should i do to get users registered on my freeswitch using digest authentication using my radius server, Like I have my radsius server and billing installed on other and working with other system. Now from where should i begin shouldd i change something in directory or in dial plans or in sofia.conf that whenever any user hits to get register freeswitch should trigger the radius client to send request to the server and if gets a return code 0 it should register the client on the basis of his username and password. I know i may be asking too much but I really am confused at this point that from where should i beging and which file to edit or start editing. Help would be appreciated. -- *Regards, Masood Ahmed *

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PRI Passthrough
by fssupport＠highergear.com 11 Oct '11

11 Oct '11

I'm in the process of building a freeswitch box to replace our existing Mitel PBX setup. I was hoping I could configure the freeswitch box with a dual port PRI card, such Sangoma A102, plug my PRI into it, then use the other port to connect to the Mite's PRI jack. That way I can test the new system, but still have the Mitel online. Does anyone have a link to how to do this type of configuration?

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outbound call with spa-3000 behind nat how?
by Alex Chan 11 Oct '11

11 Oct '11

Hi folks, I want to use a spa-3000 with freeswitch setup. The freeswitch box is on the internet, and the spa-3000 is used at home which access the internet via ADSL connection behind nat. I have follow the instruction as link below: http://wiki.freeswitch.org/wiki/SPA3102_FreeSwitch_HowTo Inbound call from PSTN just works. The problem is make a outbound call to PSTN via spa-3000. This line did not make any sense in my scenarios <action application="bridge" data="sofia/internal/${destination_number}@[IP address of SPA3102]:5061" /> as: 1. the WAN ip address is dynamic (ADSL) 2. the port is dynamic too (behind nat) so please advice. Thanks. -- Alex Chan

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CISCO FXO - IVR
by Oliver Schenk 11 Oct '11

11 Oct '11

Hi All, I have posted here some months back regarding interfacing CISCO with FreeSWITCH. I have had basic success, however there is quite a lot of fine tuning to do which is beyond my general knowledge or what I can find via Google. Currently I can do the following: 1. Make a call from any phone to a specific phone number. This phone number ends up at the CISCO FXO card. This then redirects to a specific extension at the FreeSwitch box. The dialing plan then causes a managed C# .NET application to start playing. 2. In the other direction, I have a windows service that monitors a given database. If any alarm conditions occur it uses ESL and the “Originate” API call to create a connection to the CISCO. The CISCO is then configured to redirect the given number back through the FXO card via plain old telephone (POTS). So, I can successfully make incoming and outgoing connnections on demand. PROBLEMS: 1. Using ESL for outgoing call, I don’t know how to determine whether a) the phone on the far end was picked up, b) whether the phone on the far end just rang out c) whether it was busy. In all cases the IVR just starts playing right away even if I don’t even pick up the phone. (I THINK THAT HAS SOMETHING TO DO WITH CISCO CONFIG). It seems that FreeSwitch simply triggers the CISCO to make a call, but then has no ability to track progress and assumes a connection was made. How do I get POTS to talk to VOIP/SIP to talk to FreeSwitch? 2. Incoming calls simply come in as “Processing unknown <0000000000>->1024 in context public. The actual incoming phone number does not seem to be getting from the CISCO to the FreeSwitch box. 3. I have 2 lines. I was simply going to assign extension 1024 for line 1 and 1025 for line 2. But for outgoing calls from my managed code how can I use ESL to say: a. Is line 1 busy? Then use Line 2 else make call. b. Is line 2 busy? Then wait for next scan else make call. My Code in the outgoing service: ESLconnection eslConection = new ESLconnection(“x.x.x.x”, “8021”, “password”); If (eslConnection.Connected() == ESL_SUCCESS) { eslConnection.Api(“originate”, “sofia/internal/0” + phoneNumber + “@” + CISCO_IP + “ ‘&managed(xNetIVR)’”); // determine if call is ringing, busy or picked up here // no idea how to do that… } Relevant CISCO config: version 12.4 ! ! ! voice service pots fax rate disable ! voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip no supplementary-service h450.2 no supplementary-service h450.3 supplementary-service h450.12 no supplementary-service sip moved-temporarily no supplementary-service sip refer fax protocol cisco sip registrar server expires max 3600 min 3600 no update-callerid no call service stop ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! voice register global mode cme source-address <CISCO IP> port 5060 max-dn 10 max-pool 10 authenticate register hold-alert tftp-path flash: ! ! ! voice-card 0 dsp services dspfarm ! ! ! voice-port 0/3/0 output attenuation -3 no comfort-noise cptone AU connection plar opx 1024 impedance complex1 caller-id enable ! voice-port 0/3/1 ! voice-port 0/3/2 ! voice-port 0/3/3 ! ! ! ! ! dial-peer voice 10 pots incoming called-number .T no digit-strip direct-inward-dial port 0/3/0 forward-digits all ! dial-peer voice 100 pots preference 1 destination-pattern 0T no digit-strip port 0/3/0 forward-digits all ! dial-peer voice 1 voip destination-pattern 1024 session protocol sipv2 session target ipv4:<FreeSwitch IP> session transport tcp dtmf-relay rtp-nte codec g711alaw no vad ! ! sip-ua presence enable ! ! Relevant FreeSwitch config: conf\dialplan\public\00_xNetIVR.xml and conf\dialplan\default\00_xNetIVR.xml: <include> <extension name="testapp"> <condition field="destination_number" expression="^(1024)"> <action application="managed" data="xNetIVR"/> </condition> </extension> </include> conf\directory\default\1024.xml <include> <user id="1024"> <params> <param name="password" value="xxxx"/> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="1024"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Extension 1024"/> <variable name="effective_caller_id_number" value="1024"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> </variables> </user> </include> Public.xml and default.xml dial plans are basically the same as default installation. Cheers, Oliver Schenk

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Problem with Sangoma D100 Transconding card
by Antonio 11 Oct '11

11 Oct '11

Hi, I'm using a sangoma D100 transconding card for the codec g729, it works perfectly until recently that it stops working with the following error: 2011-09-27 10:32:45.192555 [DEBUG] mod_sangoma_codec.c:1095 sngtc_get_existing_rtp_session 2011-09-27 10:32:45.192555 [DEBUG] mod_sangoma_codec.c:1095 sngtc_get_free_rtp_session 2011-09-27 10:32:45.192555 [DEBUG] mod_sangoma_codec.c:224 New allocated port 13284 for IP 10.254.254.1/10.254.254.1 2011-09-27 10:32:45.192555 [DEBUG] mod_sangoma_codec.c:224 New allocated port 13286 for IP 10.254.254.1/10.254.254.1 2011-09-27 10:32:45.192555 [ERR] mod_sangoma_codec.c:1106 Create Transcoding Session Error 2011-09-27 10:32:45.192555 [DEBUG] mod_sangoma_codec.c:243 Released port 13284 for IP 10.254.254.1/10.254.254.1 2011-09-27 10:32:45.192555 [DEBUG] mod_sangoma_codec.c:243 Released port 13286 for IP 10.254.254.1/10.254.254.1 2011-09-27 10:32:45.192555 [DEBUG] mod_sangoma_codec.c:1095 sngtc_release_rtp_session 2011-09-27 10:32:45.192555 [ERR] mod_sangoma_codec.c:486 Failed to create Sangoma encoding session. 2011-09-27 10:32:45.192555 [ERR] switch_core_io.c:1060 Codec Sangoma G729 encoder error! This error has trigger by the stop/start of freeswitch. I had to restart the sangoma daemon (sngtc_server) to make it work again. Do i have to restart the sangoma daemon every time i restart the freeswitch? shouldn't they be independent? -- Best regards, _________________________ António Silva E-mail:asilva@wirelessmundi.com

2 5

portaudio with 5.1 sound card
by Nicolás Menoni 11 Oct '11

11 Oct '11

Hello! I have a 5.1 sound card. The question is: Can I use the Front, Rear and Center channels for independents streams with mod_portaudio? I have been trying but I couldn't find the way! Thank you from Uruguay!

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hangup error missing remote port
by Allan Piske 11 Oct '11

11 Oct '11

Hi, I'm having some trouble discovering the source and solution to one problem. I setup a FS box for receiving faxes, and a already have another FS box as SBC. SPA3102 ---> opensips ---> FS_SBC --> opensips --> FS_FAX this works ok, call gets answered, i hear the fax tone, the SPA3102 changes to T38 and everyone is happy. Nortel CS2000 ------> FS_SBC -------> opensips(regitrar,rtpproxy,etc) ---> FS_FAX but his doen't. This little devil CS2000 sends every invite with m=image header along with m=audio. With aparently FS doesn't like. like that: freeswitch@internal> recv 1215 bytes from udp/[10.143.82.250]:5060 at 08:26:08.226858: ------------------------------------------------------------------------ INVITE sip:4730365302@10.143.82.253:5060;transport=UDP;user=phone SIP/2.0 Record-Route: <sip:10.143.82.250;lr=on;ftag=-45026-10d9cdb-2668a317-10d9cdb> f: <sip:4784064435@10.150.65.16:5060 ;user=phone>;tag=-45026-10d9cdb-2668a317-10d9cdb t: <sip:4730365302@10.143.82.250:5060;user=phone> i: 416cf3e01041960a13c410d9cdb1d2cba42b82409c10f14bb8-0322-4481 CSeq: 1 INVITE User-agent: CS2000_NGSS/9.0 P-Asserted-Identity: <sip:4784064435@10.150.65.16;user=phone> Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK,UPDATE Via: SIP/2.0/UDP 10.143.82.250;branch=z9hG4bK1d78.5262ccd1.0 v: SIP/2.0/UDP PAE1CS2K:5060;maddr=10.150.65.16;branch=z9hG4bK-10d9cdb-1d2cba43-57ebb21d Max-Forwards: 69 m: <sip:10.150.65.16:5060;transport=UDP> k: 100rel c: application/sdp l: 420 v=0 o=PVG 1315110636830 1315110636830 IN IP4 10.152.204.202 s=- p=+1 6135555555 c=IN IP4 10.152.204.202 t=0 0 m=audio 55920 RTP/AVP 18 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=fmtp:18 annexb=no m=image 64112 udptl t38 a=T38FaxVersion:0 a=T38FaxMaxBuffer:1100 a=T38FaxMaxDatagram:612 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPRedundancy But then we have a problem ... My beloved FS_SBC changes the SDP before sending it to the next hop .. as it should becase they are on diferent subnets and set to proxy_media 2011-09-04 01:26:08.231102 [DEBUG] sofia_glue.c:1759 sofia/tpa/4730365302 Patched SDP --- v=0 o=PVG 1315110636830 1315110636830 IN IP4 10.152.204.202 s=- p=+1 6135555555 c=IN IP4 10.152.204.202 t=0 0 m=audio 55920 RTP/AVP 18 8 101 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 m=image 64112 udptl t38 a=T38FaxVersion:0 a=T38FaxMaxBuffer:1100 a=T38FaxMaxDatagram:612 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPRedundancy +++ v=0 o=FreeSWITCH 2718891449 2718891450 IN IP4 A.B.C.D s=FreeSWITCH p=+1 6135555555 c=IN IP4 A.B.C.D t=0 0 m=audio 19262 RTP/AVP 18 8 101 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 m=image 19262 udptl t38 a=T38FaxVersion:0 a=T38FaxMaxBuffer:1100 a=T38FaxMaxDatagram:612 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPRedundancy so we have now 2 m= fields with the same port !!!!! then when it gets answered on the FS_FAX box .. we got disconnection on the FS_SBC box 2011-09-04 01:26:08.257064 [DEBUG] sofia.c:4761 Channel sofia/tpa/4730365302 entering state [completing][200] 2011-09-04 01:26:08.257064 [DEBUG] sofia.c:4772 Remote SDP: v=0 o=FreeSWITCH 1315079202 1315079203 IN IP4 189.45.192.19 s=FreeSWITCH c=IN IP4 189.45.192.19 t=0 0 m=audio 31436 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 m=image 0 udptl 19 2011-09-04 01:26:08.258409 [DEBUG] sofia.c:4761 Channel sofia/tpa/4730365302 entering state [ready][200] 2011-09-04 01:26:08.258409 [DEBUG] switch_channel.c:2830 (sofia/tpa/4730365302) Callstate Change RINGING -> ACTIVE 2011-09-04 01:26:08.258409 [DEBUG] switch_channel.c:2842 Send signal sofia/voxip/4784064435@10.150.65.16:5060 [BREAK] 2011-09-04 01:26:08.258409 [NOTICE] sofia.c:5318 Channel [sofia/tpa/4730365302] has been answered 2011-09-04 01:26:08.258409 [DEBUG] sofia_glue.c:2774 Set Codec sofia/tpa/4730365302 PROXY/8000 20 ms 160 samples 0 bits 2011-09-04 01:26:08.258409 [DEBUG] sofia_glue.c:3079 PROXY AUDIO RTP [sofia/tpa/4730365302] A.B.C.D:19262->W.X.Y.Z:0 codec: 0 ms: 20 2011-09-04 01:26:08.258409 [ERR] sofia_glue.c:3512 AUDIO RTP REPORTS ERROR: [Missing remote port] 2011-09-04 01:26:08.258409 [DEBUG] switch_channel.c:2563 (sofia/tpa/4730365302) Callstate Change ACTIVE -> HANGUP 2011-09-04 01:26:08.258409 [NOTICE] sofia_glue.c:3513 Hangup sofia/tpa/4730365302 [CS_CONSUME_MEDIA] [DESTINATION_OUT_OF_ORDER] 2011-09-04 01:26:08.258409 [DEBUG] switch_channel.c:2579 Send signal sofia/tpa/4730365302 [KILL] 2011-09-04 01:26:08.258409 [DEBUG] switch_core_session.c:1116 Send signal sofia/tpa/4730365302 [BREAK] 2011-09-04 01:26:08.259605 [DEBUG] switch_ivr_originate.c:3299 Originate Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER] 2011-09-04 01:26:08.259605 [INFO] mod_dptools.c:2647 Originate Failed. Cause: DESTINATION_OUT_OF_ORDER port = 0 !!! is that maybe a fault in opensips in the middle because the both m= fields with the same port ? , or one of the FS boxes ? something is broken here, sadly :( Appreciate any help. Allan

2 2

Attendant transfer Fails
by Joe Flemmings 11 Oct '11

11 Oct '11

My attendant transfer and can not seam to see eny errors. It just hangs up. See log bellow. Please help 2011-10-11 01:04:47.844969 [DEBUG] sofia.c:5649 Replaces: [ 11968503-b9ad5388-5ff2c6b1(a)192.168.1.72 ;to-tag=gU5125FNXNQ1K;from-tag=DAF258EB-4AB67790] 2011-10-11 01:04:47.844969 [NOTICE] sofia.c:5766 Attended Transfer [e4c3df1b-f06c-41bb-972c-8bca0c454b55][21baf737-8e51-4409-9d39-3c0b1ed654cb] 2011-10-11 01:04:47.844969 [DEBUG] switch_ivr_bridge.c:1487 (sofia/internal/ sip:df45234@99.65.217.24:52783) State Change CS_EXCHANGE_MEDIA -> CS_HIBERNATE 2011-10-11 01:04:47.844969 [DEBUG] switch_core_session.c:1114 Send signal sofia/internal/sip:df45234@99.65.217.24:52783 [BREAK] 2011-10-11 01:04:47.844969 [DEBUG] switch_ivr_bridge.c:1489 (sofia/external/ 12132568954(a)10.10.58.26) State Change CS_EXCHANGE_MEDIA -> CS_HIBERNATE 2011-10-11 01:04:47.844969 [DEBUG] switch_core_session.c:1114 Send signal sofia/external/12132568954(a)10.10.58.26 [BREAK] 2011-10-11 01:04:47.844969 [DEBUG] switch_core_session.c:769 Send signal sofia/external/12132568954(a)10.10.58.26 [BREAK] 2011-10-11 01:04:47.844969 [DEBUG] switch_core_session.c:769 Send signal sofia/internal/sip:df45234@99.65.217.24:52783 [BREAK] 2011-10-11 01:04:47.844969 [DEBUG] sofia.c:5796 (sofia/internal/ pbx_70222_2564(a)66.113.78.36) State Change CS_EXECUTE -> CS_PARK 2011-10-11 01:04:47.844969 [DEBUG] switch_core_session.c:1114 Send signal sofia/internal/pbx_70222_2564(a)66.113.78.36 [BREAK] 2011-10-11 01:04:47.854974 [DEBUG] switch_ivr_play_say.c:1649 done playing file 2011-10-11 01:04:47.854974 [DEBUG] switch_core_session.c:707 Send signal sofia/external/12132568954(a)10.10.58.26 [BREAK] 2011-10-11 01:04:47.854974 [DEBUG] switch_ivr_bridge.c:329 Send signal sofia/internal/pbx_70222_2564(a)66.113.78.36 [BREAK] 2011-10-11 01:04:47.854974 [DEBUG] switch_ivr_bridge.c:584 BRIDGE THREAD DONE [sofia/internal/pbx_70222_2564(a)66.113.78.36] 2011-10-11 01:04:47.854974 [DEBUG] switch_ivr_bridge.c:604 Send signal sofia/internal/sip:df45234@99.65.217.24:52783 [BREAK] 2011-10-11 01:04:47.854974 [DEBUG] switch_core_session.c:707 Send signal sofia/internal/sip:df45234@99.65.217.24:52783 [BREAK] 2011-10-11 01:04:47.854974 [DEBUG] switch_core_session.c:707 Send signal sofia/internal/pbx_70222_2564(a)66.113.78.36 [BREAK] 2011-10-11 01:04:47.854974 [DEBUG] switch_cpp.cpp:988 sofia/internal/ pbx_70222_2564(a)66.113.78.36 destroy/unlink session from object 2011-10-11 01:04:47.854974 [DEBUG] switch_core_state_machine.c:371 (sofia/internal/pbx_70222_2564(a)66.113.78.36) State EXECUTE going to sleep 2011-10-11 01:04:47.854974 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/pbx_70222_2564(a)66.113.78.36) Running State Change CS_PARK 2011-10-11 01:04:47.854974 [DEBUG] switch_core_state_machine.c:380 (sofia/internal/pbx_70222_2564(a)66.113.78.36) State PARK 2011-10-11 01:04:47.854974 [DEBUG] switch_core_state_machine.c:210 sofia/internal/pbx_70222_2564(a)66.113.78.36 Standard PARK 2011-10-11 01:04:47.854974 [DEBUG] switch_ivr_bridge.c:584 BRIDGE THREAD DONE [sofia/internal/sip:df45234@99.65.217.24:52783] 2011-10-11 01:04:47.854974 [DEBUG] switch_ivr_bridge.c:604 Send signal sofia/internal/pbx_70222_2564(a)66.113.78.36 [BREAK] 2011-10-11 01:04:47.854974 [DEBUG] switch_core_state_machine.c:374 (sofia/internal/sip:df45234@99.65.217.24:52783) State EXCHANGE_MEDIA going to sleep 2011-10-11 01:04:47.854974 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/sip:df45234@99.65.217.24:52783) Running State Change CS_HIBERNATE 2011-10-11 01:04:47.854974 [DEBUG] switch_core_state_machine.c:386 (sofia/internal/sip:df45234@99.65.217.24:52783) State HIBERNATE 2011-10-11 01:04:47.854974 [DEBUG] mod_sofia.c:223 sofia/internal/ sip:df45234@99.65.217.24:52783 SOFIA HIBERNATE 2011-10-11 01:04:47.854974 [DEBUG] switch_ivr_bridge.c:734 (sofia/internal/ sip:df45234@99.65.217.24:52783) State Change CS_HIBERNATE -> CS_RESET 2011-10-11 01:04:47.854974 [DEBUG] switch_core_session.c:1114 Send signal sofia/internal/sip:df45234@99.65.217.24:52783 [BREAK] 2011-10-11 01:04:47.854974 [DEBUG] switch_core_state_machine.c:386 (sofia/internal/sip:df45234@99.65.217.24:52783) State HIBERNATE going to sleep 2011-10-11 01:04:47.854974 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/sip:df45234@99.65.217.24:52783) Running State Change CS_RESET 2011-10-11 01:04:47.854974 [DEBUG] switch_core_state_machine.c:367 (sofia/internal/sip:df45234@99.65.217.24:52783) State RESET 2011-10-11 01:04:47.854974 [DEBUG] mod_sofia.c:167 sofia/internal/ sip:df45234@99.65.217.24:52783 SOFIA RESET 2011-10-11 01:04:47.854974 [DEBUG] switch_ivr_bridge.c:719 sofia/internal/ sip:df45234@99.65.217.24:52783 CUSTOM RESET 2011-10-11 01:04:47.854974 [DEBUG] switch_ivr_bridge.c:726 (sofia/internal/ sip:df45234@99.65.217.24:52783) State Change CS_RESET -> CS_SOFT_EXECUTE 2011-10-11 01:04:47.854974 [DEBUG] switch_core_session.c:1114 Send signal sofia/internal/sip:df45234@99.65.217.24:52783 [BREAK] 2011-10-11 01:04:47.854974 [DEBUG] switch_core_state_machine.c:367 (sofia/internal/sip:df45234@99.65.217.24:52783) State RESET going to sleep 2011-10-11 01:04:47.854974 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/sip:df45234@99.65.217.24:52783) Running State Change CS_SOFT_EXECUTE 2011-10-11 01:04:47.854974 [DEBUG] switch_core_state_machine.c:377 (sofia/internal/sip:df45234@99.65.217.24:52783) State SOFT_EXECUTE 2011-10-11 01:04:47.854974 [DEBUG] mod_sofia.c:560 SOFIA SOFT_EXECUTE

1 0

continue_on_fail
by Boris Kovalenko 11 Oct '11

11 Oct '11

Hello! Have read docs but misundertood :( I use mod_lcr which return two or more gateways to the same destination. And when the user is busy with the first gateway the call continues with the second and so on. How may I to not continue gateways if user is busy? -- С уважением, Борис Коваленко ЗАО "Тагнет" тел. +7 (3435) 230001 факс +7 (3435) 230005

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How can make mod_rmtp support text message?
by 大羽.风向 11 Oct '11

11 Oct '11

We can find that there are two mods can bind to work together. One is mod_rmtp , another is mod_sms. How can make mod_rmtp work with mod_sms to support text message on flash client ? Anybody have suggestions？ Regards. Martin.

4 3