- FreeSWITCH-users - FreeSWITCH Mailing Lists

local-network-acl parameter
by Dmitry Bely 20 Feb '11

20 Feb '11

Can you explain what objects is associated with this ACL? What does it allow or deny? http://wiki.freeswitch.org/wiki/Acl is silent on that. - Dmitry Bely

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Strange warning in ftmod_sangoma_isdn
by Mariusz Kołodziejczyk 20 Feb '11

20 Feb '11

Hi, I have sometimes strange warning in module ftmod_sangoma_isdn .... 2011-02-20 11:31:35.190942 [WARNING] ftmod_wanpipe.c:946 [s1c31][1:16] Failed to read from sangoma device: No buffer space available (-65) 2011-02-20 11:31:35.190942 [WARNING] ftdm_io.c:3756 [s1c31][1:16] raw I/O read filed 2011-02-20 11:31:35.190942 [WARNING] ftmod_sangoma_isdn.c:386 [s1c31][1:16] Failed to read from channel 2011-02-20 11:31:36.002943 [WARNING] ftmod_wanpipe.c:946 [s1c31][1:16] Failed to read from sangoma device: No buffer space available (-65) 2011-02-20 11:31:36.002943 [WARNING] ftdm_io.c:3756 [s1c31][1:16] raw I/O read filed 2011-02-20 11:31:36.002943 [WARNING] ftmod_sangoma_isdn.c:386 [s1c31][1:16] Failed to read from channel 2011-02-20 11:31:36.390942 [WARNING] ftmod_wanpipe.c:946 [s1c31][1:16] Failed to read from sangoma device: No buffer space available (-65) 2011-02-20 11:31:36.390942 [WARNING] ftdm_io.c:3756 [s1c31][1:16] raw I/O read filed 2011-02-20 11:31:36.390942 [WARNING] ftmod_sangoma_isdn.c:386 [s1c31][1:16] Failed to read from channel 2011-02-20 11:31:36.998941 [INFO] ftmod_sangoma_isdn_stack_rcv.c:688 [SNGISDN Q921] wp1: Protocol: T200 expired N200 times(15): T200 Timeout(260) 2011-02-20 11:31:36.998941 [INFO] ftmod_sangoma_isdn_stack_rcv.c:688 [SNGISDN Q921] wp1: Protocol: SABME reset started(17): SABME tx to initate link reset(264) 2011-02-20 11:31:37.002947 [WARNING] ftmod_wanpipe.c:946 [s1c31][1:16] Failed to read from sangoma device: No buffer space available (-65) 2011-02-20 11:31:37.002947 [WARNING] ftdm_io.c:3756 [s1c31][1:16] raw I/O read filed 2011-02-20 11:31:37.002947 [WARNING] ftmod_sangoma_isdn.c:386 [s1c31][1:16] Failed to read from channel 2011-02-20 11:31:38.002947 [WARNING] ftmod_wanpipe.c:946 [s1c31][1:16] Failed to read from sangoma device: No buffer space available (-65) 2011-02-20 11:31:38.002947 [WARNING] ftdm_io.c:3756 [s1c31][1:16] raw I/O read filed 2011-02-20 11:31:38.002947 [WARNING] ftmod_sangoma_isdn.c:386 [s1c31][1:16] Failed to read from channel 2011-02-20 11:31:39.002947 [WARNING] ftmod_wanpipe.c:946 [s1c31][1:16] Failed to read from sangoma device: No buffer space available (-65) 2011-02-20 11:31:39.002947 [WARNING] ftdm_io.c:3756 [s1c31][1:16] raw I/O read filed 2011-02-20 11:31:39.002947 [WARNING] ftmod_sangoma_isdn.c:386 [s1c31][1:16] Failed to read from channel 2011-02-20 11:31:40.002947 [WARNING] ftmod_wanpipe.c:946 [s1c31][1:16] Failed to read from sangoma device: No buffer space available (-65) 2011-02-20 11:31:40.002947 [WARNING] ftdm_io.c:3756 [s1c31][1:16] raw I/O read filed 2011-02-20 11:31:40.002947 [WARNING] ftmod_sangoma_isdn.c:386 [s1c31][1:16] Failed to read from channel 2011-02-20 11:31:40.998941 [INFO] ftmod_sangoma_isdn_stack_rcv.c:688 [SNGISDN Q921] wp1: Protocol: T200 expired N200 times(15): T200 Timeout(260) 2011-02-20 11:31:40.998941 [INFO] ftmod_sangoma_isdn_stack_rcv.c:688 [SNGISDN Q921] wp1: Protocol: Data Link connection DN(4): Disconnect initiated(263) .... this series of warning appearances almost every half hour, sometimes as single warning. this is'nt related to any monitoring params in system (for example 'vmstat 1' shows nothing special in logs just before i get this warning message, during and a bit after, for warning above vmstat shows (i added time stamp to the end every each rows): procs -----------memory---------- ---swap-- -----io---- -system-- ----cpu---- r b swpd free buff cache si so bi bo in cs us sy id wa 0 0 0 4631820 167932 1087348 0 0 0 0 100 2697 0 0 100 0 02-20-2011 11:31:30 0 0 0 4631944 167932 1087348 0 0 0 0 100 2693 0 0 100 0 02-20-2011 11:31:31 0 0 0 4631820 167932 1087348 0 0 0 0 100 2687 0 0 100 0 02-20-2011 11:31:32 0 0 0 4631944 167932 1087348 0 0 0 72 121 2711 0 0 100 0 02-20-2011 11:31:33 0 0 0 4631944 167932 1087348 0 0 0 0 104 2713 0 0 100 0 02-20-2011 11:31:34 0 0 0 4631944 167932 1087352 0 0 0 0 104 2703 0 0 100 0 02-20-2011 11:31:35 0 0 0 4631944 167932 1087352 0 0 0 0 104 2716 0 0 100 0 02-20-2011 11:31:36 0 0 0 4631944 167932 1087352 0 0 0 152 137 2773 0 0 100 0 02-20-2011 11:31:37 0 0 0 4631944 167932 1087356 0 0 0 0 107 2708 0 0 100 0 02-20-2011 11:31:38 0 0 0 4631820 167932 1087356 0 0 0 0 104 2700 0 0 100 0 02-20-2011 11:31:39 0 0 0 4631820 167932 1087360 0 0 0 0 104 2701 0 0 100 0 02-20-2011 11:31:40 0 0 0 4631820 167932 1087360 0 0 0 0 104 2719 0 0 100 0 02-20-2011 11:31:41 0 0 0 4631820 167932 1087360 0 0 0 156 124 2736 0 0 100 0 02-20-2011 11:31:42 0 0 0 4631696 167932 1087360 0 0 0 8 109 2697 0 0 100 0 02-20-2011 11:31:43 0 0 0 4631696 167932 1087360 0 0 0 0 103 2695 0 0 100 0 02-20-2011 11:31:44 0 0 0 4631696 167932 1087360 0 0 0 0 101 2684 0 0 100 0 02-20-2011 11:31:45 When i've tested system i can make outgoing and incoming calls without any problem (max 3 calls simultaneously with different configuration, but i'm affraid if i use this in production environment (about 30-50 calls via isdn) this warning may cause addiction problems. I have installed: freeswitch : FreeSWITCH Version 1.0.head (git-a3b18e5 2011-02-19 16-47-19 +0100) linux (Debian) : Linux PBX 2.6.26-2-amd64 #1 SMP Thu Sep 16 15:56:38 UTC 2010 x86_64 GNU/Linux wanrouter: WANPIPE Release: 3.5.18 libsng_isdn : libsng_isdn-7.3.0.x86_64 sangoma card: A102DE Is anybody have the same warning ? Thanks

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destination number variable
by Serge Yuriev 20 Feb '11

20 Feb '11

Hello In INFO app after call i see (shortened for readability) 2011-02-09 00:20:00.824547 [INFO] mod_dptools.c:1202 CHANNEL_DATA: Caller-Direction: [inbound] Caller-Username: [nevian] Caller-Dialplan: [XML] Caller-Caller-ID-Name: [Serge S. Yuriev] Caller-Caller-ID-Number: [nevian] Caller-Callee-ID-Name: [Outbound Call] Caller-Callee-ID-Number: [79645835822(a)81.16.114.33] Caller-Network-Addr: [109.173.67.229] Caller-ANI: [nevian] Caller-Destination-Number: [79645835822] variable_sip_from_display: [Serge S. Yuriev] variable_sip_full_from: ["Serge S. Yuriev" <sip:nevian@cranz.nevian.org>;tag=624579331] variable_sip_full_to: [<sip:79645835822@cranz.nevian.org>] variable_sip_req_user: [79645835822] variable_sip_req_uri: [79645835822(a)cranz.nevian.org] variable_sip_req_host: [cranz.nevian.org] variable_sip_to_user: [79645835822] variable_sip_to_uri: [79645835822(a)cranz.nevian.org] In mod_radius_cdr profile->destination_number returns it as 79645835822(a)81.16.114.33 Which field should I use to get only number? Termination via h323 if this matters -- wbr, Serge

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RTP Proxy config and DTMF conversion problem
by Mac 19 Feb '11

19 Feb '11

Dear ALL, Thats my first post here. I am struggling for some time with DTMF issue. Let me introduce you my configuration. The main task is to configure RTP Proxy with full topology hiding - OPERATOR_1 ------- Freeswitch ------- OPERATOR_2 Here is a prt of my config: - sip profile. I have one new sip profile with two gateways (OPERATOR_1 and OPERATOR_2) <include> <gateway name="OP1"> <param name="username" value="X1.Y1.V1.Z1"/> <param name="password" value="09321234"/> <param name="proxy" value="X1.Y1.V1.Z1"/> <param name="from-domain" value="test.com"/> <param name="register" value="false"/> <param name="caller-id-in-from" value="true"/> <param name="extension" value="test"/> <param name="extension-in-contact" value="true"/> <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/> <param name="disable-transcoding" value="true"/> <param name="inbound-late-negotiation" value="true"/> </gateway> </include> - dialplan <include> <context name="public"> <extension name="term"> <condition field="network_addr" expression="^(X1\.Y1\.V1\.Z1|X2\.Y2\.V1\.Z1)$" break="on-false"/> <condition field="destination_number" expression="^46(\d{9})$"> <action application="set" data="proxy_media=true"/> <action application="set" data="bypass_media=false"/> <action application="start_dtmf"/> <action application="bridge" data="sofia/gateway/OP1/$0"/> </condition> </extension> </context> </include> Everything is fine, but i have problem with DTMF conversion from RFC2833 to inband. Refering to http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmfand http://wiki.freeswitch.org/wiki/Proxy_Media i have done all the necessary things. vars.xml <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMA,G.729,PCMU,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMA,G.729,PCMU,GSM"/> Placing <param name="dtmf-type" value="none"/> does not help anyway. Could sb point me where the problem is located? Thanks in advance, Mac

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DTMF not working properly
by Aloysius Lloyd 19 Feb '11

19 Feb '11

Hi All, I have a DTMF problem in version FreeSWITCH Version 1.0.head (git-39ff78b 2011-02-18 20-16-11 -0600) Phone - Polycom 335 Connected to the FreeSWITCH. Dial out a External Number . Then dial a Extension on the Destination IVR ( Destination is not FreeSWITCH) ... most of the time FreeSWITCH not sending the correct digits. I can see from the Console only the First Digit Captured by FreeSWITCH not rest of the Digits this is happen randomly I try with different SIP Trunk from different provider.. Same behavior --- here is partial of DTMF first digit then no digits shows up on the CLI 2011-02-19 14:18:20.688643 [DEBUG] switch_rtp.c:2294 Send start packet for [2] ts=1463986596 dur=160/160/2080 seq=54483 2011-02-19 14:18:20.708130 [DEBUG] switch_rtp.c:2230 Send middle packet for [2] ts=1463986596 dur=320/320/2080 seq=54484 2011-02-19 14:18:20.728688 [DEBUG] switch_rtp.c:2230 Send middle packet for [2] ts=1463986596 dur=480/480/2080 seq=54485 2011-02-19 14:18:20.748147 [DEBUG] switch_rtp.c:2230 Send middle packet for [2] ts=1463986596 dur=640/640/2080 seq=54486 2011-02-19 14:18:20.768727 [DEBUG] switch_rtp.c:2230 Send middle packet for [2] ts=1463986596 dur=800/800/2080 seq=54487 2011-02-19 14:18:20.788050 [DEBUG] switch_rtp.c:2230 Send middle packet for [2] ts=1463986596 dur=960/960/2080 seq=54488 2011-02-19 14:18:20.808506 [DEBUG] switch_rtp.c:2230 Send middle packet for [2] ts=1463986596 dur=1120/1120/2080 seq=54489 2011-02-19 14:18:20.828043 [DEBUG] switch_rtp.c:2230 Send middle packet for [2] ts=1463986596 dur=1280/1280/2080 seq=54490 2011-02-19 14:18:20.848501 [DEBUG] switch_rtp.c:2230 Send middle packet for [2] ts=1463986596 dur=1440/1440/2080 seq=54491 2011-02-19 14:18:20.867984 [DEBUG] switch_rtp.c:2230 Send middle packet for [2] ts=1463986596 dur=1600/1600/2080 seq=54492 2011-02-19 14:18:20.888589 [DEBUG] switch_rtp.c:2230 Send middle packet for [2] ts=1463986596 dur=1760/1760/2080 seq=54493 2011-02-19 14:18:20.908531 [DEBUG] switch_rtp.c:2230 Send middle packet for [2] ts=1463986596 dur=1920/1920/2080 seq=54494 2011-02-19 14:18:20.927937 [DEBUG] switch_rtp.c:2230 Send end packet for [2] ts=1463986596 dur=2080/2080/2080 seq=54495 2011-02-19 14:18:20.927937 [DEBUG] switch_rtp.c:2230 Send end packet for [2] ts=1463986596 dur=2080/2080/2080 seq=54496 2011-02-19 14:18:20.927937 [DEBUG] switch_rtp.c:2230 Send end packet for [2] ts=1463986596 dur=2080/2080/2080 seq=54497 --- Please let me know how to fix this or is it a Bug? Thanks Lloyd

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newbies is stuck / windows version of freeswitch won't start up after reboot
by Edward de Jong 19 Feb '11

19 Feb '11

I installed Freeswitch (by downloading the .EXE version for windows x64) and was running it okay on Windows 7 Professional, 64 bit edition, but after a restart, when I try to run fs_cli it gives me a socket error and fails. I am not really running anything else much on my machine, and i figure somehow that port has gotten blocked. I turned off windows firewall, but still it won't run, i can't figure out why it was running so well, but now on a clean restart it won't start up. I didn't install it as a startup program, I was manually running fs_cli and freeswitchconsole, which are the two freeswitch programs, yes? thanks in advance. edj

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Skype 2.0.72 for Linux
by Michael Collins 19 Feb '11

19 Feb '11

Does anyone know where to find this? It's the only one that the wiki says works with mod_skypopen but not even Google seems to be able to find one. (I find "2.0.72" but it links to the latest beta on the skype.com site which is not what we want.) Any suggestions are appreciated. -MC

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IVR application
by Jason b.a 19 Feb '11

19 Feb '11

aha thx MC ! Ken give me ur suggestions pls , if i need a ivr application that controlled the media server using SIP and an embedded control language !

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ESL socket outbound in C++
by Stephen Wilde 19 Feb '11

19 Feb '11

I'm learning to use ESL in C++ to do a call control and I want to share my code with the community. I had a little difficult at startup so I hope that this code can help someone. The code is: http://pastebin.freeswitch.org/15418 Using "esl_oop" module, I have implemented a basic ESL Channel class (that can be extended) so the "main" event handling loop can be very simple, for example for an unuseful esl bridge application, can be: ESLchannel legA(&connection, inbound_uuid); ESLchannel legB(&connection); legB.Originate(dialstring) while (1) { ESLevent * event = connection.recvEventTimed(1000); if (event) { const char * ename = event->getHeader("Event-Name"); const char * event_uuid = event->getHeader("unique-id"); // It's an event for LegB ? //------------------------- if (legB.Uuid() == event_uuid) { if (!strcmp(ename, "CHANNEL_PROGRESS_MEDIA")) { legA.PreAnswer(); legA.BridgeTo(legB); } else if (!strcmp(ename, "CHANNEL_ANSWER")) { legA.Answer(); legA.BridgeTo(legB); } else if (!strcmp(ename,"CHANNEL_HANGUP")) { const char * cause = event->getHeader("Hangup-Cause"); legA.Hangup(cause); } } // It's an event for LegA ? //------------------------- else if (legA.Uuid() == event_uuid) { if (!strcmp(ename,"CHANNEL_HANGUP")) { const char * cause = event->getHeader("Hangup-Cause"); legB.Hangup(cause); } } delete event; } }

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NAT issue
by Spencer Thomason 19 Feb '11

19 Feb '11

Hello all, I'm having a strange NAT problem. My Freeswitch server has a public ip and the endpoints which are Polycom IP 650s are all behind NAT with mainly Netgear SRX 5308 routers at multiple locations. The server is on a public class C network and one of the locations has its router on the same class C. I.e x.x.x.7/24-router, x.x.x.240/24-server. I'm only experiencing this with 2 of about 30 extensions. The first endpoint to register from this location gets detected as NAT but the others with the ports remapped do not. This maybe explains it better: Call-ID: 4feea796-bd96e2ed-e16d27ac(a)10.6.71.103 User: 1103(a)x.x.x.240 Contact: "user" <sip:1103@10.6.71.103;received=x.x.x. 7:5060;fs_nat=yes;fs_path=sip%3A1103%40x.x.x.7%3A5060> Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0769 Status: Registered(UDP-NAT)(unknown) EXP(2011-02-18 11:15:47) EXPSECS(1143) Host: pbx10.5ninesolutions.com IP: x.x.x.7 Port: 5060 Auth-User: 1103 Auth-Realm: pbx10.5ninesolutions.com MWI-Account: 1103(a)x.x.x.x Call-ID: d21b71b2-75baa80d-768c0b08(a)10.6.71.104 User: 1101(a)x.x.x.240 Contact: "user" <sip:1101@10.6.71.104> Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0769 Status: Registered(UDP)(unknown) EXP(2011-02-18 11:16:48) EXPSECS(1204) Host: pbx10.5ninesolutions.com IP: x.x.x.7 Port: 1036 Auth-User: 1101 Auth-Realm: pbx10.5ninesolutions.com MWI-Account: 1101(a)x.x.x.x Call-ID: a74e61a6-1863ea7d-789686c4(a)10.6.71.105 User: 1102(a)x.x.x.240 Contact: "user" <sip:1102@10.6.71.105> Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0769 Status: Registered(UDP)(unknown) EXP(2011-02-18 11:17:23) EXPSECS(1239) Host: pbx10.5ninesolutions.com IP: x.x.x.7 Port: 1036 Auth-User: 1102 Auth-Realm: pbx10.5ninesolutions.com MWI-Account: 1102(a)x.x.x.x Call-ID: 100b5148-6a47060a-13c4-50029-c844-22fc4dd3-c844 User: 1801(a)97.67.110.240 Contact: "user" <sip:1801@10.6.71.106:5060> Agent: MP201 B 1FXS/3.0.1_build_19 Status: Registered(UDP)(unknown) EXP(2011-02-18 11:26:23) EXPSECS(1779) Host: pbx10.5ninesolutions.com IP: 97.67.110.7 Port: 1036 Auth-User: 1801 Auth-Realm: pbx10.5ninesolutions.com MWI-Account: 1801(a)x.x.x.x ------- Another Location ---- Call-ID: dca2a276-4eb94f97-c55177b0(a)10.1.10.186 User: 4204(a)x.x.x.240 Contact: "user" <sip:4204@10.1.10.186;received=x.x.x.x: 5060;fs_nat=yes;fs_path=sip%3A4204%40x.x.x.x%3A5060> Agent: PolycomSoundStationIP-SSIP_5000-UA/3.3.1.0769 Status: Registered(UDP-NAT)(unknown) EXP(2011-02-18 11:28:04) EXPSECS(1880) Host: pbx10.5ninesolutions.com IP: x.x.x.x Port: 5060 Auth-User: 4204 Auth-Realm: pbx10.5ninesolutions.com MWI-Account: 4204(a)x.x.x.x The port and IP are detected correctly but the contact is a private IP. Any clue as to why the two extension 1101 and 1102 are any different? Thanks, Spencer

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