- FreeSWITCH-dev - FreeSWITCH Mailing Lists

Connecting FS to MS UM
by Michael Collins 29 Jun '12

29 Jun '12

Hey, if anyone knows about how this can be done with FreeSWITCH, please contact me off list: http://help.outlook.com/en-us/140/gg702670%28d=loband%29.aspx I know someone looking to provide a solution for his customer. I'd like to get this solution documented on the wiki... Thanks, MC

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questions about making a new endpoint
by Seven Du 29 Jun '12

29 Jun '12

Hi, I'm working on a mod_rtsp endpoint the idea is to stream sip video to a rtsp client such as VLC and also a sip client could "call" a rtsp camera later. I copied some codes from mod_sofia and it is too long to understand for me so I'd like to get some inputs either highlight me how to do or point out where to read from mod_sofia. 1) VLC "calls" in as a-leg and bridge to a sip client with video(or play_fsv), I made it sort of working now, but - How to disable the read_frame and read_video frame io_routine ? I set them to NULL and it fails on read, so I made a fake read function there, kind of ugly. - I hard coded to H264 and I can see video on VLC, I'd like to do like sip later negotiation then when b-leg sip client offers SDP and offer to a-leg when DESCRIBE - looks like switch_io complains when no audio codec, is it possible to do video only calls? or is it possible to set audio and video rtp to a different path ? 2) Call a rtsp camera would be easy later 3) Maybe a client can call in and record? I don't have such a client. I would like to share the code later if others interested on this. Thanks, Seven. -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn Sent with Sparrow (http://www.sparrowmailapp.com)

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FreeSWITCH Conf Call Today
by Michael Collins 27 Jun '12

27 Jun '12

Hello gang, This week's conference call will be starting in less than an hour. Here's the agenda page: http://wiki.freeswitch.org/wiki/FS_weekly_2012_06_27 Talk to you soon! -Michael

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stream a file multicast
by Xijing Dai 27 Jun '12

27 Jun '12

hey all, Just wonder is there any progress on on stream a file multicast? I saw some topics on it back to 2010. Currently, I need the similar functionalities. Cheers Xijing Dai

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I can not see sendonly/recvonly from SDP
by Xijing Dai 26 Jun '12

26 Jun '12

hey I am using mod_esf to do the multicasting. In bypass_media mode, it will send recvonly SDP to other side. I use wireshark to catch the sip/sdp packets, but I can not see freeswitch sent recvonly inside SDP. After that, I try to set NDLB-sendrecv-in-session to be true (specified here: http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#NDLB-sendrecv-in-… ). It still does not send recvonly to the other side. Did I miss anything? I also want to allow softphone to be multicasting listener, is there any extension module to do it?? Cheers Xijing Dai

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FreeSWITCH Weekly News and Notes
by Michael Collins 25 Jun '12

25 Jun '12

Welcome to the last week of June! Things quieted down a bit last week, but we still had some interesting items for your consideration. Darren Schreiber from the 2600hz<http://www.2600hz.com?cc12-0625>project joined us last week on our conference call <http://wiki.freeswitch.org/wiki/Weekly_Conference_Call?cc12-0625> to present "SIP 101" - a gentle introduction to the SIP protocol for those doing VoIP. It was a nice discussion and we plan to do a follow up, tentatively scheduled for July 18. The audio is available in the usual place<http://torrents.freeswitch.org/conf_call_2012-06-20.torrent?cc12-0625>and the slides are available here <http://wiki.freeswitch.org/files/SIP_101_Presentation.ppt?cc12-0625>. For this week's<http://wiki.freeswitch.org/wiki/FS_weekly_2012_06_27?cc12-0625>conference call we are going to have an open discussion about topics of interest to our community members. Our ClueCon schedule <http://www.cluecon.com/schedule?cc12-0625> is now full! Please visit the schedule to see a list of speakers and talk titles. We have a few presentations whose titles are yet to be announced, so check back often to stay updated. We are pleased to have Brad Pitt joining us this year! (No, not that one <http://en.wikipedia.org/wiki/Brad_pitt> - we have this one <http://www.cluecon.com/speaker/108?cc12-0625>.) Bradley J Pitt is a long-time veteran of the technology and telecom industries and recently joined Barracuda Networks <http://www.cudatel.com?cc12-0625>. His discussion will touch upon both technology and business aspects of the IP PBX world. As a reminder, you still have about one week before the "bits shift" and you go from 16 down to 8 tickets for your ClueCon registration<http://www.cluecon.com/register?cc12-0625>. Register right away so that you have the best chance possible to win one of the 10 Android tablets that will be given away this year! See you in Chicago! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE cc12-0625

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Are you using NDLB-force-rport = "safe" ?
by Michael Collins 25 Jun '12

25 Jun '12

Hi gang, A friend of mine asked how many people are using this. Are you? It's mostly for Polycoms and non-polys to have force rport in a typical hosted environment. If you have been using this setting in production please give us some feedback. Thanks! -Michael

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bandtel SIP TRUNK config
by victor.velo＠videotron.ca 21 Jun '12

21 Jun '12

Hello, I need some help to validate my bandtel sip trunking config... I get the following info from bandtel: Your SIP User Name: 201XXXXXXX Your SIP Password: XXXXXX Your ANI: 207XXXXXX Your Toll Free/s: N/A Your DID/s: 207XXXXX (prefix with0317) --- DNS Server Addresses: >65.175.129.149 >66.237.65.90 Registration must be sent to: registrar.bandtel.com •Outbound Calls: Outbound calls must be sent to the following proxy: Primary: proxy1.bandtel.com; Secondary: proxy2.bandtel.com my setting: ....... cat /etc/resolv.conf nameserver 65.175.129.149 nameserver 66.237.65.90 .................. cat 01_bandtel.xml <include> <gateway name="bandtel.com"> <param name="username" value="201XXXXXX"/> <param name="password" value="XXXXX"/> <param name="from-user" value="+1207XXXXXXX"/> <param name="proxy" value="registrar.bandtel.com"/> <param name="outbound-proxy" value="proxy1.bandtel.com"/> <param name="expire-seconds" value="600"/> <param name="register" value="true"/> <param name="caller-id-in-from" value="true"/> </gateway> </include> ....................... cat 01_bandtel_did.xml <include> <extension name="bandtel_did"> <condition field="destination_number" expression="^(207XXXXX)$"> <action application="set" data="domain_name=$${domain}"/> <action application="transfer" data="XXXX XML default"/> </condition> </extension> </include> ............. I don't get any error in the log file and I don't get anything by calling the bandtel DID.... Do I miss some settings or params somewhere???? Thank for the helps, Victor Velo

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New wiki page: Language_Files
by Michael Collins 21 Jun '12

21 Jun '12

Hello all, I've created a new wiki page: http://wiki.freeswitch.org/wiki/Language_Files I've wanted this for some time and Brian's OCD finally took over and I just *had* to create this page. :) I've added languages that I know about or have heard about. *This page is not complete, and it needs your help.* If you know anything about any language files then please please go to that page and add what you know. If you have any issues with editing the wiki then let me know and I will be happy to assist. In fact, if need be I will do a 20-minute tutorial on wiki editing on an upcoming community conference call. Thanks! -Michael

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FreeSWITCH Conf Call Today: SIP 101
by Michael Collins 20 Jun '12

20 Jun '12

Hey all, Don't forget that at 1pm EDT, 10am PDT we have the FreeSWITCH community conference call. Darren Schreiber will be doing the SIP 101 discussion. We look forward to having you all with us. -Michael

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