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FreeSWITCH Windows Service does not Start
by Ron Avriel 21 Jan '09

21 Jan '09

Hi, If I try starting FreeSWITCH Windows service it immediately fails with a messagebox: "The FreeSWITCH service on local computer started and then stopped. Some services stop automatically...etc." I noticed that the service is installed to log on as "NT AUTHORITY\NetworkService". If I change this to "Local System account" then FreeSWITCH starts and runs OK. This failure occurred on multiple Windows XP servers. Why does it fail and why does it not use the local system account like almost all services? Thanks, Ron

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New wiki page needs your attention!
by Michael Collins 21 Jan '09

21 Jan '09

Hello FreeSWITCHers! I'm putting the finishing touches on a wiki page that we hope will make it easier for users to request help and for the dev team and power users to digest and process those requests. What I need first and foremost is for everyone to please read this page: http://wiki.freeswitch.org/wiki/Reporting_Bugs Please give me feedback. Put yourself in the shoes of a relative newbie. Is the information easy to follow? On the flip side, if you wanted to help someone, ask yourself, if they follow the steps on this page will that suffice? Are there places that need improvement? Can you think of anything else that can be added? NOTE: I'm still working on the TDM/OpenZAP section as well as the sections on scripting, event socket, elements of a jira ticket, etc. If you have suggestions for content on those pages please email me off list or hop and and fill in some of the blanks. The core development team really appreciates all of your help. Now that FS is growing like mad we are at the point where it is imperative that we have reliable documentation for new ones so that the developers and other experts can focus on advancing the project even further. Let's all lend a hand by improving the documentation. Thanks again! -MC (mercutioviz)

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Bridge destination returned from a python script?
by Brian Deacon 21 Jan '09

21 Jan '09

Greetings my soon-to-be-BFF's :) After the last 6 weeks or so thrashing about in the land of Asterisk, I must say I'm quite impressed with what I'm learning about FreeSwitch. So the scenario I'm trying to deal with is bridging a call between two not-internal endpoints where the numbers to connect are determined by database-fed business logic. The recipe here is very close to what I need: http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#exec_in_answer_con… Except that I'm a language snob and would greatly prefer python to javascript. :) So my minor question is can this line: <action application="set" data="group_confirm_file=javascript test.js"/> Become this line: <action application="set" data="group_confirm_file=python foo.bar.baz"/> But javascript isn't a deal-breaker. I have two bridging scenarios that were too tricky for me in asterisk land. (And my apologies, I'm new to telephony, so I don't have the lingo down). Our FS server would be reachable by multiple 800 numbers. Based on the number that was dialed to reach our FS server, we would look up an external phone number + extension to be part 1 of a bridge (which we would use the group_confirm_file script to make sure we had gotten past the IVR). So for that scenario, how would I accomplish something like: <action application="bridge" data=" ${someVariableSetByScript}"/> Can my foo.bar.baz script make a session.setVariable call that I could drop into data=""? My other scenario is similar, but we need to externally signal the FS server to place a call to phone + extension, confirm it was connected, and then bridge it to a DID. Is XmlRpc the best (only?) way to poke FS into doing something? Thanks muchly! Brian

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Caller ID presentation
by Paul D. 21 Jan '09

21 Jan '09

Hi, I am fairly new to FS. Is there a way to switch off/on caller ID presentation dynamically, similar to "callingpres" in *? Also, probably a stupid question, but how do I search this mailing list archive? Thx.

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SBC functions in FS
by Jon Bruel 20 Jan '09

20 Jan '09

I'm looking into the first design phase of a setup like this: PSTN <-> FS <-> Various SIP PBXs or nothing <-> SBC <-> Internet <-> Customer Where I'm considering using the FS as the SBC and as an alternative to OpenSIPS (OpenSER), where the FS has the advantage of a design having only one type of switch, e.g. less education to the team supporting the system. If this should work, the FS should be able to route SIP messages such as NOTIFY, SUBSCRIBE etc. Is that possible, and what are the measures to implement such a routing function? /Jon

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Stun disabled
by info＠freeswitch.es 20 Jan '09

20 Jan '09

Hi, very basic question: how can I disable stun for internal and external profiles? Thank you in advance Regards

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what could be efficiency of this hardware ?
by Marek Gorecki 20 Jan '09

20 Jan '09

Hi, Merit question: how many calls could be processed by FS on the following hardware: Intel Core 2 Duo 2*2,13 Ghz 3072 MB RAM 400 GB HD 100mbit/s port By processing calls I understand just switching with billing - no processing of any media, no transcoding, just call-through according to dialplan. Question is related to availablility of such server for 90€ / month with no charge and no limit for bandwidth ( say: NO LIMIT IN BANDWIDTH ) ( http://www.giga-international.com/detail_server.php?id=87 ) Which parameters should be upgraded for to reach better efficiency ? Regards, /\/\arekg

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Problem with digium te220p
by Krzysztof Zimnicki 20 Jan '09

20 Jan '09

>Can you post your openzap.conf file? >-MC Sure. [span zt] name => OpenZAP number => 1 trunk_type => E1 b-channel => 1-15 d-channel => 16 b-channel => 17-31 [span zt] name => OpenZAP number => 2 trunk_type => E1 b-channel => 32-46 d-channel => 47 b-channel => 48-62

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Freeswitch not sending out 183
by Juan Backson 20 Jan '09

20 Jan '09

Hi, I am running some continuous testing hitting FS. There are a couple errors that gets popped up occasionally and I am trying to find out why. In one of the trace, I am seeing FS not sending 183. The weird thing is that this problem is not happening everything, but on a very rarely basis. Also, there is no error message being sent. Does anyone know under what situation would freeswitch not sending out 183? recv 771 bytes from udp/[192.168.1.122]:5060 at 14:28:50.460222: ------------------------------------------------------------------------ INVITE sip:0019008@192.168.1.116:5070 SIP/2.0 Record-Route: <sip:192.168.1.122;lr=on;ftag=9> Via: SIP/2.0/UDP 192.168.1.122;branch=z9hG4bKe6e9.09c88aa1.0 Via: SIP/2.0/UDP 192.168.1.6:7001 From: 19008 <sip:19008@192.168.1.122:7001>;tag=9 To: 0019008 <sip:0019008@192.168.1.122 <sip%3A0019008(a)192.168.1.122>> Call-ID: 9-10894(a)192.168.1.6 CSeq: 2 INVITE Contact: <sip:19008@192.168.1.6:7001> Max-Forwards: 69 User-Agent: Performance Test Content-Type: application/sdp Content-Length: 272 P-hint: inbound->inbound v=0 o=user1 53655765 2353687637 IN IP4 192.168.1.6 s=- t=0 0 c=IN IP4 192.168.1.6 m=audio 6032 RTP/AVP 0 9 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv ------------------------------------------------------------------------ send 354 bytes to udp/[192.168.1.122]:5060 at 14:28:50.460805: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.122;branch=z9hG4bKe6e9.09c88aa1.0 Via: SIP/2.0/UDP 192.168.1.6:7001 Record-Route: <sip:192.168.1.122;lr=on;ftag=9> From: 19008 <sip:19008@192.168.1.122:7001>;tag=9 To: 0019008 <sip:0019008@192.168.1.122 <sip%3A0019008(a)192.168.1.122>> Call-ID: 9-10894(a)192.168.1.6 CSeq: 2 INVITE User-Agent: Freeswitch Media Gateway Content-Length: 0 ------------------------------------------------------------------------ [36m2009-01-20 09:28:50 [NOTICE] switch_channel.c:565 switch_channel_set_name() New Channel sofia/internal/19008@192.168.1.122:7001[a7f0594e-e6fe-11dd-9a2c-fbbebdd2887f] [32m2009-01-20 09:28:50 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 19008->0019008 in context public [32m2009-01-20 09:28:50 [INFO] switch_ivr_async.c:1653 switch_ivr_bind_dtmf_meta_session() Bound A-Leg: 1 execute_extension::a_record XML features [32m2009-01-20 09:28:50 [INFO] switch_ivr_async.c:1653 switch_ivr_bind_dtmf_meta_session() Bound A-Leg: 2 execute_extension::a_stoprecord XML features [32m2009-01-20 09:28:50 [INFO] switch_ivr_async.c:1653 switch_ivr_bind_dtmf_meta_session() Bound A-Leg: 3 execute_extension::a_att_xfer XML features [32m2009-01-20 09:28:50 [INFO] switch_ivr_async.c:1660 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 1 execute_extension::b_record XML features [32m2009-01-20 09:28:50 [INFO] switch_ivr_async.c:1660 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 2 execute_extension::b_stoprecord XML features [32m2009-01-20 09:28:50 [INFO] switch_ivr_async.c:1660 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 3 execute_extension::b_att_xfer XML features [36m2009-01-20 09:28:50 [NOTICE] switch_channel.c:565 switch_channel_set_name() New Channel sofia/internal/19008 [a816c7fa-e6fe-11dd-9a2c-fbbebdd2887f] send 1229 bytes to udp/[192.168.1.122]:5060 at 14:28:50.715136: ------------------------------------------------------------------------ INVITE sip:19008@192.168.1.122:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.116:5070;rport;branch=z9hG4bKKe15F6Dap2jBa Max-Forwards: 68 From: "19008" <sip:FreeswitchMediaGateway@openser ;transport=udp>;tag=mB8gaveZvmF8K To: <sip:19008@192.168.1.122:5060> Call-ID: 7f75543a-61a1-122c-8282-001517871e28 CSeq: 110112529 INVITE Contact: <sip:FreeswitchMediaGateway@192.168.1.116:5070;transport=udp> User-Agent: Freeswitch Media Gateway Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 335 P-hint: inbound->inbound Remote-Party-ID: "19008" <sip:19008@openser>;screen=yes;privacy=off v=0 o=FreeSWITCH 5551695261821560906 6736752020354058642 IN IP4 192.168.1.116 s=FreeSWITCH c=IN IP4 192.168.1.116 t=0 0 m=audio 12016 RTP/AVP 0 9 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 ------------------------------------------------------------------------ recv 354 bytes from udp/[192.168.1.122]:5060 at 14:28:50.745112: ------------------------------------------------------------------------ SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 192.168.1.116:5070 ;rport=5070;branch=z9hG4bKKe15F6Dap2jBa From: "19008" <sip:FreeswitchMediaGateway@openser ;transport=udp>;tag=mB8gaveZvmF8K To: <sip:19008@192.168.1.122:5060> Call-ID: 7f75543a-61a1-122c-8282-001517871e28 CSeq: 110112529 INVITE Server: OpenSIPS (1.4.3-notls (x86_64/linux)) Content-Length: 0 ------------------------------------------------------------------------ recv 438 bytes from udp/[192.168.1.122]:5060 at 14:28:50.748929: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.116:5070 ;received=192.168.1.116;rport=5070;branch=z9hG4bKKe15F6Dap2jBa Record-Route: <sip:192.168.1.122;lr=on;ftag=mB8gaveZvmF8K> From: "19008" <sip:FreeswitchMediaGateway@openser ;transport=udp>;tag=mB8gaveZvmF8K To: <sip:19008@192.168.1.122:5060>;tag=9 Call-ID: 7f75543a-61a1-122c-8282-001517871e28 CSeq: 110112529 INVITE Contact: <sip:192.168.1.7:7000;transport=UDP> Content-Length: 0 ------------------------------------------------------------------------ recv 740 bytes from udp/[192.168.1.122]:5060 at 14:28:50.749130: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.116:5070 ;received=192.168.1.116;rport=5070;branch=z9hG4bKKe15F6Dap2jBa Record-Route: <sip:192.168.1.122;lr=on;ftag=mB8gaveZvmF8K> From: "19008" <sip:FreeswitchMediaGateway@openser ;transport=udp>;tag=mB8gaveZvmF8K To: <sip:19008@192.168.1.122:5060>;tag=9 Call-ID: 7f75543a-61a1-122c-8282-001517871e28 CSeq: 110112529 INVITE Contact: <sip:192.168.1.7:7000;transport=UDP> Content-Type: application/sdp Content-Length: 272 [36m2009-01-20 09:28:50 [NOTICE] sofia.c:2627 sofia_handle_sip_i_state() Ring-Ready sofia/internal/19008! v=0 o=user1 53655765 2353687637 IN IP4 192.168.1.7 s=- c=IN IP4 192.168.1.7 t=0 0 m=audio 6000 RTP/AVP 0 9 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv ------------------------------------------------------------------------ send 463 bytes to udp/[192.168.1.122]:5060 at 14:28:50.749566: ------------------------------------------------------------------------ ACK sip:192.168.1.7:7000;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 192.168.1.116:5070;rport;branch=z9hG4bKmQtyH1yDKB9XN Route: <sip:192.168.1.122;lr=on;ftag=mB8gaveZvmF8K> Max-Forwards: 70 From: "19008" <sip:FreeswitchMediaGateway@openser ;transport=udp>;tag=mB8gaveZvmF8K To: <sip:19008@192.168.1.122:5060>;tag=9 Call-ID: 7f75543a-61a1-122c-8282-001517871e28 CSeq: 110112529 ACK Contact: <sip:FreeswitchMediaGateway@192.168.1.116:5070;transport=udp> Content-Length: 0 ------------------------------------------------------------------------ [36m2009-01-20 09:28:50 [NOTICE] sofia.c:3065 sofia_handle_sip_i_state() Channel [sofia/internal/19008] has been answered [32m2009-01-20 09:28:50 [INFO] mod_sofia.c:1294 sofia_receive_message() Asked to send early media by sofia/internal/19008@192.168.1.122:7001 [32m2009-01-20 09:28:50 [INFO] mod_sofia.c:1335 sofia_receive_message() Ring SDP: v=0 o=FreeSWITCH 1232449708 1232449709 IN IP4 192.168.1.116 s=FreeSWITCH c=IN IP4 192.168.1.116 t=0 0 m=audio 12022 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv [36m2009-01-20 09:28:50 [NOTICE] mod_sofia.c:1338 sofia_receive_message() Ring-Ready sofia/internal/19008@192.168.1.122:7001! [36m2009-01-20 09:28:50 [NOTICE] mod_sofia.c:1338 sofia_receive_message() Pre-Answer sofia/internal/19008@192.168.1.122:7001! [36m2009-01-20 09:28:50 [NOTICE] switch_ivr_originate.c:1838 switch_ivr_originate() Channel [sofia/internal/19008@192.168.1.122:7001] has been answered send 1058 bytes to udp/[192.168.1.122]:5060 at 14:28:50.820186: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.122;branch=z9hG4bKe6e9.09c88aa1.0 Via: SIP/2.0/UDP 192.168.1.6:7001 Record-Route: <sip:192.168.1.122;lr=on;ftag=9> From: 19008 <sip:19008@192.168.1.122:7001>;tag=9 To: 0019008 <sip:0019008@192.168.1.122 <sip%3A0019008(a)192.168.1.122> >;tag=K2er80XUZBSNr Call-ID: 9-10894(a)192.168.1.6 CSeq: 2 INVITE Contact: <sip:mod_sofia@192.168.1.116:5070;transport=udp> User-Agent: Freeswitch Media Gateway Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 267 v=0 o=FreeSWITCH 6403206013178204633 6807646593930218658 IN IP4 192.168.1.116 s=FreeSWITCH c=IN IP4 192.168.1.116 t=0 0 m=audio 12022 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 ------------------------------------------------------------------------ send 1058 bytes to udp/[192.168.1.122]:5060 at 14:28:51.321221: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.122;branch=z9hG4bKe6e9.09c88aa1.0 Via: SIP/2.0/UDP 192.168.1.6:7001 Record-Route: <sip:192.168.1.122;lr=on;ftag=9> From: 19008 <sip:19008@192.168.1.122:7001>;tag=9 To: 0019008 <sip:0019008@192.168.1.122 <sip%3A0019008(a)192.168.1.122> >;tag=K2er80XUZBSNr Call-ID: 9-10894(a)192.168.1.6 CSeq: 2 INVITE Contact: <sip:mod_sofia@192.168.1.116:5070;transport=udp> User-Agent: Freeswitch Media Gateway Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 267 v=0 o=FreeSWITCH 6403206013178204633 6807646593930218658 IN IP4 192.168.1.116 s=FreeSWITCH c=IN IP4 192.168.1.116 t=0 0 m=audio 12022 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 ------------------------------------------------------------------------

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Re: [Freeswitch-users] Dialing Out Problem via Gateway
by Alberto Ceccarelli 20 Jan '09

20 Jan '09

Sorry, but the problem remains. I can add some details. After response, we can listen the DTMF tones, but not the voice. Now my conf. files: vars.xml: <X-PRE-PROCESS cmd="set" data="outbound_caller_name=FreeSWITCH"/> <X-PRE-PROCESS cmd="set" data="outbound_caller_id=0000000000"/> dialplan\default.xml: <extension name="Local_Extension"> <condition field="destination_number" expression="^(10[01][0-9])$"> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> </condition> <condition field="destination_number" expression="^${caller_id_number}$"> <action application="set" data="voicemail_authorized=${sip_authorized}"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="check default ${domain_name} ${dialed_extension}"/>  <anti-action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <anti-action application="bind_meta_app" data="2 b s record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <anti-action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/> <anti-action application="set" data="ringback=${us-ring}"/> <anti-action application="set" data="transfer_ringback=$${hold_music}"/> <anti-action application="set" data="call_timeout=30"/>  <anti-action application="set" data="hangup_after_bridge=true"/>  <anti-action application="set" data="continue_on_fail=true"/> <anti-action application="db" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> <anti-action application="db" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> <anti-action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/>  <anti-action application="db" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> <anti-action application="bridge" data="user/${dialed_extension}@ ${domain_name}"/> <anti-action application="answer"/> <anti-action application="sleep" data="1000"/> <anti-action application="voicemail" data="default ${domain_name} ${dialed_extension}"/> </condition> </extension> Ciao. Alby. 2009/1/19 Brian West <brian(a)freeswitch.org> You modified the Local_Extension in default.xml > > 2009-01-17 16:10:08 [DEBUG] mod_dialplan_xml.c:117 parse_exten() > Regex: [Local_Extension] destination_number(1008) =~ /^+39020000000$/ > 2009-01-17 16:10:08 [ERR] switch_regex.c:94 switch_regex_perform() > COMPILE ERROR: 1 [nothing to repeat][^+39020000000$] > 2009-01-17 16:10:08 [DEBUG] mod_dialplan_xml.c:119 parse_exten() Regex > mismatch > > > Remove that match for the /^+39020000000$/ and put it back to the > default value... it should work fine. > > /b > > On Jan 17, 2009, at 10:02 AM, Alberto Ceccarelli wrote: > > > <freeswitch.log> > > > _______________________________________________ > 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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