- FreeSWITCH-users - FreeSWITCH Mailing Lists

Connecting Call Between Two SIP Trunks
by Dan Lan 31 Aug '11

31 Aug '11

Hi, I want to use FS to accept call from SIP_TrunkA and terminate to SIP_TrunkB Both SIP trunks are using IP authentication, no need for username and password. for incoming call (SIP_TrunkA), I have add the IP address of SIP_TrunkA in to acl.conf.xml <node type="allow" cidr="xxx.xxx.xxx.xxx/32"/> I understand the incoming call will go to the public context, so I think I need to do something here. I dont know what to do next. 1. I try to establish a gateway for SIP_TrunkB for my outgoing call, but sofia require me to have the username and password for the trunk. I dont know where to add the SIP_TrunkB in freeswitch, since the provider of SIP_TrunkB only need to recoginize my FS IP address. 2. After I establish SIP_TrunkB, how should I do on public dialplan to route the call from SIP_TrunkA to SIP_TrunkB? should I use "transfer" or "bridge", could I make a dialplan that can route all the call from IP address of A to IP address of B? Sorry for the newbie question, I try to look up on wiki but only got partial information for me. Any help and any directions or hints are appreciated Dan Lan

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FS behind an IPCOP firewall
by Praveen Ray 31 Aug '11

31 Aug '11

Hi Here's is my setup. It's not that complicated. 172.168.128.0/24 subnet FS <-------------------------------------------> IPCOP Firewall | | 10.10.0.0/24 subnet Extension 1000 <--------------------------------->| |<-------------------------------------------- Extension 1001 (172.168.128.188) (10.10.0.170) Firewall is set to allow all UDP traffic in the port rane 1024-65000 between these two subnets. Both 1000 and 1001 register successfully with FS. I can make outgoing calls (using vitelity) using both. I can call 1000 from 1001 ok. However, calling 1001 from 1000 doesn't work. It goes right to 1001's VM. It's as if FS doesn't know 1001 is registered. My config is almost same as what comes with vanilla FS install (except I defined outgoing vitelity and 1000.xml and 1001.xml extensions). Any obvious config I'm missing? Thanks in advance.

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FreeSWITCH™ in Mexico
by João Mesquita 31 Aug '11

31 Aug '11

Estimados, Quisiera ante todo pedirles disculpas por enviarles este mensaje off-topic. Yo estaré dando una charla gratuita de FreeSWITCH™ en México(DF) en el dia 30 de Agosto. La charla será como un taller en donde instalaremos placas TDM y crearemos casos reales de instalaciones. La idea es presentar un poco las posibilidades de FreeSWITCH™ y difundir un poco más la idea en México. El evento es gratuito pero con cupos limitados. El evento es auspiciado por Khomp y Toga Soluciones. Para inscribirse, basta ingresar al link: http://togakhomp.eventbrite.com/ ******************************** Guys, First of all, sorry for the off topic. I am going to give a speech on FreeSWITCH™ in Mexico this next 28th of August and would like to invite the mexican community to join me. It will be more like a workshop presenting real world scenarios using TDM cards provided by Khomp. The event is free but with limited room. This is being sponsored by Khomp and Toga Soluciones. If you are interested, please join us at: http://togakhomp.eventbrite.com/ Thank you all. Regards, João Mesquita

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FreeSWITCH Conference Call Today
by Michael Collins 31 Aug '11

31 Aug '11

Hello all, Today's agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2011_08_31 I'd like to talk about the concept of "giving back" which I touched upon in this article: http://www.freeswitch.org/node/339 Talk to you at 1PM EST/10AM Pacific/1700 UTC! -MC

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rtp natting
by Chris Martineau 31 Aug '11

31 Aug '11

Hi, I am new to freeswitch and wish to bolt it into our existing network to provide conferencing services. Currently we just use an opensips/rtpproxy configuration for simple proxy switching. However looking at the features of freeswitch it seems to me that I could replace my entire setup with just the freeswitch platform. Would that be a fair comment? If so then there a number of things that I am struggling to find clearly defined in the documentation. 1. Currently our rtp proxy scenario issues a port to come back on but waits for incoming rtp packets to determine the exact port to return rtp on when the caller is behind a nat. Does freeswitch do this, does it need to be configured to work that way? Freeswitch would be on a public address. 2. Currently opensips stores all user info in a mysql database and with 500000+ users is easy to manipulate. How do you deal with such a large user database in a pure xml environment such as freeswitch? Many thanks for any help you can offer. Regards Chris

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FreeSwitch Utilizes 100% of CPU Sometimes When User Quits Conference
by Erwin Merioles 31 Aug '11

31 Aug '11

Hey guys, I've been having trouble with FreeSwitch for quite some time. We're trying to use mod_rtmp to add sound to one of our applications, www.321meet.com. Unfortunately, FreeSwitch's CPU usage spikes when the host ( the first one to join the conference ), quits or closes the browser window. I've checked and this always happen when the following line is called - 2011-08-24 12:24:28.057200 [NOTICE] rtmp_tcp.c:73 Pollout: true FS Console Log follows : 2011-08-24 12:22:59.177294 [NOTICE] mod_rtmp.c:743 New RTMP session [4a97320c-e50d-4ed3-a59c-4aef799d379d] 2011-08-24 12:22:59.477278 [NOTICE] rtmp_sig.c:121 Sent connect reply 2011-08-24 12:23:15.597196 [INFO] rtmp_sig.c:136 Replied to createStream (1) 2011-08-24 12:23:16.237201 [INFO] rtmp_sig.c:274 Got publish on stream 1. 2011-08-24 12:23:17.177293 [INFO] rtmp_sig.c:136 Replied to createStream (2) 2011-08-24 12:23:17.177293 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled control packet (type=0x3) 2011-08-24 12:23:17.457195 [WARNING] sofia.c:4403 Ping succeeded voip9.telsome.com with code 404 - count -1/1/1, state UP 2011-08-24 12:23:17.457195 [INFO] rtmp_sig.c:136 Replied to createStream (3) 2011-08-24 12:23:17.457195 [NOTICE] switch_channel.c:897 New Channel rtmp/default/3213533 [8e7ccce8-172c-4b91-9183-1ad9d2f0e6dd] 2011-08-24 12:23:17.457195 [ERR] rtmp_sig.c:305 Couldn't create call. 2011-08-24 12:23:17.457195 [WARNING] sofia.c:4403 Ping succeeded testin with code 404 - count -1/1/1, state UP 2011-08-24 12:23:17.497199 [INFO] mod_dialplan_xml.c:336 Processing <0000000000>->3213533 in context default 2011-08-24 12:23:17.497199 [NOTICE] mod_rtmp.c:497 Channel [rtmp/default/3213533] has been answered 2011-08-24 12:23:17.497199 [INFO] mod_conference.c:6644 using channel sound prefix: /usr/local/freeswitch/sounds/en/us/callie 2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 rtmp/default/3213533 binding '0' to 'mute' 2011-08-24 12:23:17.597289 [INFO] switch_ivr_async.c:164 Digit parser mod_conference: Setting realm to conf 2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 rtmp/default/3213533 binding '*' to 'deaf mute' 2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 rtmp/default/3213533 binding '9' to 'energy up' 2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 rtmp/default/3213533 binding '8' to 'energy equ' 2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 rtmp/default/3213533 binding '7' to 'energy dn' 2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 rtmp/default/3213533 binding '3' to 'vol talk up' 2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 rtmp/default/3213533 binding '2' to 'vol talk zero' 2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 rtmp/default/3213533 binding '1' to 'vol talk dn' 2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 rtmp/default/3213533 binding '6' to 'vol listen up' 2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 rtmp/default/3213533 binding '5' to 'vol listen zero' 2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 rtmp/default/3213533 binding '4' to 'vol listen dn' 2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 rtmp/default/3213533 binding '#' to 'hangup' 2011-08-24 12:23:17.797195 [INFO] rtmp_sig.c:159 Sending audio 2011-08-24 12:23:17.797195 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled control packet (type=0x3) 2011-08-24 12:23:18.117290 [INFO] rtmp_sig.c:274 Got publish on stream 3. 2011-08-24 12:23:36.697195 [ERR] rtmp.c:678 Read error 2011-08-24 12:23:36.697195 [NOTICE] mod_rtmp.c:788 RTMP session ended [4a97320c-e50d-4ed3-a59c-4aef799d379d] 2011-08-24 12:23:36.697195 [NOTICE] mod_rtmp.c:803 Hangup rtmp/default/3213533 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] 2011-08-24 12:23:36.777294 [NOTICE] switch_core_session.c:1347 Session 1 (rtmp/default/3213533) Ended 2011-08-24 12:23:36.777294 [NOTICE] switch_core_session.c:1349 Close Channel rtmp/default/3213533 [CS_DESTROY] 2011-08-24 12:23:48.237202 [NOTICE] mod_rtmp.c:743 New RTMP session [3ff4786f-7cae-4653-879a-ae95a9d50742] 2011-08-24 12:23:48.557204 [NOTICE] rtmp_sig.c:121 Sent connect reply 2011-08-24 12:23:55.357293 [NOTICE] mod_rtmp.c:743 New RTMP session [af763930-9054-4c76-a0eb-9c351e75949d] 2011-08-24 12:23:55.757290 [NOTICE] rtmp_sig.c:121 Sent connect reply 2011-08-24 12:23:58.957195 [INFO] rtmp_sig.c:136 Replied to createStream (1) 2011-08-24 12:23:59.517204 [INFO] rtmp_sig.c:274 Got publish on stream 1. 2011-08-24 12:24:00.557201 [INFO] rtmp_sig.c:136 Replied to createStream (2) 2011-08-24 12:24:00.557201 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled control packet (type=0x3) 2011-08-24 12:24:00.557201 [INFO] rtmp_sig.c:136 Replied to createStream (3) 2011-08-24 12:24:00.857222 [NOTICE] switch_channel.c:897 New Channel rtmp/default/3213533 [5de55413-5cb9-4c18-8b65-9c438833449f] 2011-08-24 12:24:00.857222 [ERR] rtmp_sig.c:305 Couldn't create call. 2011-08-24 12:24:00.857222 [INFO] mod_dialplan_xml.c:336 Processing <0000000000>->3213533 in context default 2011-08-24 12:24:00.857222 [NOTICE] mod_rtmp.c:497 Channel [rtmp/default/3213533] has been answered 2011-08-24 12:24:00.857222 [INFO] mod_conference.c:6644 using channel sound prefix: /usr/local/freeswitch/sounds/en/us/callie 2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 rtmp/default/3213533 binding '0' to 'mute' 2011-08-24 12:24:00.857222 [INFO] switch_ivr_async.c:164 Digit parser mod_conference: Setting realm to conf 2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 rtmp/default/3213533 binding '*' to 'deaf mute' 2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 rtmp/default/3213533 binding '9' to 'energy up' 2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 rtmp/default/3213533 binding '8' to 'energy equ' 2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 rtmp/default/3213533 binding '7' to 'energy dn' 2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 rtmp/default/3213533 binding '3' to 'vol talk up' 2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 rtmp/default/3213533 binding '2' to 'vol talk zero' 2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 rtmp/default/3213533 binding '1' to 'vol talk dn' 2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 rtmp/default/3213533 binding '6' to 'vol listen up' 2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 rtmp/default/3213533 binding '5' to 'vol listen zero' 2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 rtmp/default/3213533 binding '4' to 'vol listen dn' 2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 rtmp/default/3213533 binding '#' to 'hangup' 2011-08-24 12:24:01.197247 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled control packet (type=0x3) 2011-08-24 12:24:01.197247 [INFO] rtmp_sig.c:274 Got publish on stream 3. 2011-08-24 12:24:01.197247 [INFO] rtmp_sig.c:159 Sending audio 2011-08-24 12:24:13.637291 [INFO] rtmp_sig.c:136 Replied to createStream (1) 2011-08-24 12:24:14.157289 [INFO] rtmp_sig.c:274 Got publish on stream 1. 2011-08-24 12:24:17.057195 [INFO] rtmp_sig.c:136 Replied to createStream (2) 2011-08-24 12:24:17.057195 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled control packet (type=0x3) 2011-08-24 12:24:17.057195 [INFO] rtmp_sig.c:136 Replied to createStream (3) 2011-08-24 12:24:17.057195 [NOTICE] switch_channel.c:897 New Channel rtmp/default/3213533 [4e6c35ce-ec56-4ef1-816a-b94704efbfbd] 2011-08-24 12:24:17.057195 [ERR] rtmp_sig.c:305 Couldn't create call. 2011-08-24 12:24:17.057195 [INFO] mod_dialplan_xml.c:336 Processing <0000000000>->3213533 in context default 2011-08-24 12:24:17.057195 [NOTICE] mod_rtmp.c:497 Channel [rtmp/default/3213533] has been answered 2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 rtmp/default/3213533 binding '0' to 'mute' 2011-08-24 12:24:17.057195 [INFO] switch_ivr_async.c:164 Digit parser mod_conference: Setting realm to conf 2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 rtmp/default/3213533 binding '*' to 'deaf mute' 2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 rtmp/default/3213533 binding '9' to 'energy up' 2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 rtmp/default/3213533 binding '8' to 'energy equ' 2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 rtmp/default/3213533 binding '7' to 'energy dn' 2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 rtmp/default/3213533 binding '3' to 'vol talk up' 2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 rtmp/default/3213533 binding '2' to 'vol talk zero' 2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 rtmp/default/3213533 binding '1' to 'vol talk dn' 2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 rtmp/default/3213533 binding '6' to 'vol listen up' 2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 rtmp/default/3213533 binding '5' to 'vol listen zero' 2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 rtmp/default/3213533 binding '4' to 'vol listen dn' 2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 rtmp/default/3213533 binding '#' to 'hangup' 2011-08-24 12:24:17.497196 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled control packet (type=0x3) 2011-08-24 12:24:17.497196 [INFO] rtmp_sig.c:274 Got publish on stream 3. 2011-08-24 12:24:17.497196 [INFO] rtmp_sig.c:159 Sending audio 2011-08-24 12:24:27.657196 [ERR] rtmp.c:678 Read error 2011-08-24 12:24:27.657196 [NOTICE] mod_rtmp.c:788 RTMP session ended [3ff4786f-7cae-4653-879a-ae95a9d50742] 2011-08-24 12:24:27.657196 [NOTICE] mod_rtmp.c:803 Hangup rtmp/default/3213533 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] 2011-08-24 12:24:27.677201 [NOTICE] switch_core_session.c:1347 Session 2 (rtmp/default/3213533) Ended 2011-08-24 12:24:27.677201 [NOTICE] switch_core_session.c:1349 Close Channel rtmp/default/3213533 [CS_DESTROY] 2011-08-24 12:24:28.057200 [NOTICE] rtmp_tcp.c:73 Pollout: true Which results to : PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 1608 root -2 -10 191m 20m 6588 R 94.3 3.5 5:39.50 fs Any ideas? Help is VERY much appreciated. TIA! Regards, Erwin D. Merioles merioles.net +63 922 837 9466 | +63 917 501 1010 | +1 760 670 3241 aY!M : erwin_merioles | Skype : erwin.merioles This message (including any attachments) contains information that may be confidential. Unless you are the intended recipient (or is authorized to receive for the intended recipient), you may not read, print, retain, use, copy, distribute, or disclose to anyone, any information contained here. If you have received this in error, please advise the sender by reply e-mail, and delete all copies of the original message (including attachments).

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ringing status and mod_rtmp
by Madovsky 31 Aug '11

31 Aug '11

Hi folks, is there any callback event for ringing status of leg B with mod_rtmp ? Thanks Franck

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CDR
by Dhairya Vora 31 Aug '11

31 Aug '11

I need to show detailed information of a particular calluuid. I tried parsing .xml file generated by freeswitch(/usr/local/freeswitch/log/xml_cdr/archive/) but it does not show *the time at which a digit was pressed*. I checked plivo log(/usr/src/plivo/tmp/plivo-outbound.log) also but it stores the log for all the calls in one file only. I want this kind of information: TIMEEVENTDETAILS2011-08-31 10:51:57PLAY please_leave_your_message_after_the_beep.wav2011-08-31 10:52:04RECORD1_1.wav2011-08-31 10:52:08PLAYyour_recorded_message_is.wav2011-08-31 10:52:10PLAY1_1.wav*2011-08-31 10:52:21**DIGIT PRESSED**0 *2011-08-31 10:52:22PLAYthank_you.wav2011-08-31 10:52:23PLAY your_message_has_been_recorded_successfully.wav using .xml file generated by freeswitch(/usr/local/freeswitch/log/xml_cdr/archive/) I could parse all the information *other than "DIGIT PRESSED"* information. Can anyone help ?

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Re: [Freeswitch-users] mod_conference - 2-digit DTMF and events
by Michael Kemp 31 Aug '11

31 Aug '11

Hi Moises Thank you for implementing this feature request so quickly! I will endeavour to install and test it and let you know the results this week. Best Wishes Michael Kemp

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Calling FMS with mod_rtmp
by Kevin Reeves 31 Aug '11

31 Aug '11

I'm interested in making a call from inside Freeswitch out to Flash Media Server using mod_rtmp. I'd like to do this so I can attach audio from a phone call into FMS. If I'm reading the documentation correctly, the only way to attach the 2 is to have FMS register with Freeswitch. Any suggestions you have would be greatly appreciated. Thanks, Kevin

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