January 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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

3 2

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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ODBC in place of db/core.db
by Adam Long 20 Jan '09

20 Jan '09

Hi Guys, I have read the documentation on the wiki and have successfully compiled FreeSwtich with odbc core support. I am able to get my SIP Profiles and Voicemail databases to load, create, and utilize the ODBC database tables successfully. However, I cannot seem to figure out how to make FreeSwitch use the ODBC database instead of the SQL Lite db/core.db I'm sure it's just a matter of properly placing the following line in one of the configs. <param name="odbc-dsn" value="dns-name:user:password"/> If anyone can point me in the right direction I would greatly appreciate it. Thank you!

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Gateways without Registration or User/Pass Authentication
by Adam Long 20 Jan '09

20 Jan '09

I am trying to setup a couple of gateways on my external profile that do not user usernames or passwords . How can I setup these gateways to match based on originating IP address? At the moment I have them defined in separate xml files in the sip_profiles/external directory. I am using the default external.xml I would like to be able to dial via this gateway and receive calls from these gateways. For example NYGW1.xml ------- > <include> <gateway name="NYGW1" cidr="10.10.10.1/32" description="Cisco Gateway Connected to Level3 PRI"> <param name="register" value="false"/> <param name="username" value="none"/> <param name="password" value="none"/> <param name="realm" value="10.10.10.1"/> <param name="proxy" value="10.10.10.1"/> <param name="from-domain" value="oneofmydomains.com"/> <param name="extension" value="inbound_oneofmydomains"/> <param name="caller-id-in-from" value="true"/> <param name="ping" value="25"/> </gateway> </include> When I remove the username/password params or set them to "" FreeSWITCH refuses to load the gateway. I tried adding the cidr="" to the gateway but no luck. Really all I want is for calls flowing in from 10.10.10.1 --- > FreeSWITCH External SIP Profile ---- > Match to NYGW1 --- > Dump into a context or extension from where I can route them as I please. Kind of how an Asterisk sip "peer" is matched on IP and routed to a context. Any ideas greatly appreciated!

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Interesting Problem with SPA 3000
by Scott Ellis 19 Jan '09

19 Jan '09

I have an interesting problem with a set up using two (or more) SPA 3000's. I make a call out on the unit to a PSTN line - great. I then try and make another call out on that line - it fails and moves on to the next one in the bridge call statement. It then goes through on the second unit. Almost always 1:30 later, I get an inbound call from the PSTN showing up in FreeSwitch, which goes to an extension - and when answered bridges that extension onto the existing call. Now I am sure that this behaviour from the SPA is a little odd, but does anyone have any tips for dealing with it from the dialplan? Most obvious being a way to not call the first unit when it already is active... I do not have it defined as a gateway - just using sofia/internal/$1@10.0.0.18:5061 so make a call. Scott

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Is anyone our there using an SPA 3000? (or 3102)
by Scott Ellis 19 Jan '09

19 Jan '09

If so, could you please share your set up? directory files, and dial plan details (gateway details if configured this way)? Scott

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Conference auto-record not working
by Pablo Hernan Saro 19 Jan '09

19 Jan '09

Hi there, I've set up in my default conference context auto-record as follows: <param name="auto-record" value="$${base_dir}/recordings/${conference_name}_${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> but recording never starts. I've checked this by starting FS in console mode but no errors or useful messages to figure out what is going on. My conference extension is set up as follows: <context name="default"> <extension name="conf_quality_1"> <condition field="destination_number" expression="^(58233)$"> <action application="answer"/> <action application="conference" data="$1@default+1633"/> </condition> </extension> </context> Could you please help me finding some missing configuration or error? Let me know if you need further information. Thanks in advance. Pablo

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