- FreeSWITCH-dev - FreeSWITCH Mailing Lists

4 WEEKS LEFT! FreeSWITCH Official Training Course - June 28th-30th in San Francisco, CA
by Darren Schreiber 29 May '10

29 May '10

Hi folks, In case you haven't heard, we're pleased to offer the first ever Official FreeSWITCH Training course, held June 28th-30th in San Francisco, CA. The class has been filling up (and airline tickets will go up, too) so if you are thinking of attending we *strongly* encourage you to sign-up now. The training is different then a basic PBX training - we'll be diving a bit deeper into cool features within FreeSWITCH. Since we're designing the course from scratch we can also customize it so you learn what you want to learn. So far, topics requested have focused more along the lines of specific modules and carrier-style switching services, so we're going to try and focus on those topics. The training is also hands-on. You'll be working with FreeSWITCH on individual workstations during the course - doing labs, etc. We'll try and keep it as interactive as we can. And of course, if you like saving money... We have two deals going right now: 1. If you send two or more people we are now offering a good discount. It will vary based on the number of people you send. Please contact me directly if you are interested in this option. 2. If you sign up for the Training + ClueCon option you save ~$300! Flood yourself with FreeSWITCH knowledge and contacts by attending both events. This is a really good deal! For more information, please visit the website at http://www.voipkb.com/ or drop me a note. Thanks much and I look forward to seeing you at the training! Sincerely, Darren Schreiber

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Register On-Line For ClueCon!
by Michael Collins 29 May '10

29 May '10

Hey there! We just wanted to let everyone know that our Web site has been updated: http://www.cluecon.com You may now click the link to get registered. Remember that you need to create a user for the ClueCon Web site or use the same username/password that you used last year. Feel free to contact us if you have any concerns or questions. Looking forward to Chicago! The ClueCon Team http://www.cluecon.com 877.742.CLUE

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Freeswitch-dev archive - nabble version
by David Swardstrom 28 May '10

28 May '10

Is there a nabble version of the freeswitch-dev archives? Someone sent me a link to the freeswitch-users one. Perhaps these links should be made available in the pages: http://lists.freeswitch.org/mailman/listinfo/freeswitch-??? For example in page: http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev Regards, Paul David Swardstrom

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GIT Access
by Georgiewskiy Yuriy 27 May '10

27 May '10

Hi, where i can get ssh key to acces GIT? I am autor of mod_h323. C уважением With Best Regards Георгиевский Юрий. Georgiewskiy Yuriy +7 4872 711666 +7 4872 711666 факс +7 4872 711143 fax +7 4872 711143 Компания ООО "Ай Ти Сервис" IT Service Ltd http://nkoort.ru http://nkoort.ru JID: GHhost(a)jabber.tula-ix.net.ru JID: GHhost(a)jabber.tula-ix.net.ru YG129-RIPE YG129-RIPE

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Get on IRC day!!
by Michael Collins 27 May '10

27 May '10

Okay everyone, please join #freeswitch on IRC! We want everyone to come hang out with us. Also, we have many new ones coming in asking for advice. Please join up and help spread the knowledge. See you online! -Michael (IRC: mercutioviz)

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Outbound call Status
by Sathish Babu 27 May '10

27 May '10

Hi, I have inbound call on one channel after process that call i am calling the below function to make outbound and get connected with the inbound. @con.execute("bridge", "sofia/gateway/junctionnetworks/#{@callout_number}") It is working perfectly. But i need to know whether the outbound call has been ringing or answered else it is failed. Is there any way to get the event about the newly made call status? Thanks and Regards, Sathishbabu

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ACD/CallCenter
by Jan Berger 27 May '10

27 May '10

Hi, Is anyone doing work on ACD or CallCenter aided for open source/FS ? Jan

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FreeSWITCH Conference Call Starting Now!
by Michael Collins 26 May '10

26 May '10

C'mon over: http://wiki.freeswitch.org/wiki/FS_weekly_2010_05_26 -MC

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Re: [Freeswitch-dev] Call to registered phone behind NAT
by Oleg Khovayko 26 May '10

26 May '10

Hi, I little narrowed problem with NATed client. There is his registration: Call-ID: 41f120d4(a)192.168.1.146 User: 1012(a)192.168.1.5 Contact: "user" <sip:1012@192.168.1.146:5060;fs_nat=yes;fs_path=sip%3A1012%4091.207.244.1%3A5060> Agent: AcctonVoIP/2.5 Status: Registered(UDP-NAT)(unknown) EXP(2010-05-26 02:45:48) Host: olegh.ath.cx IP: 91.207.244.1 Port: 5060 Auth-User: 1012 Auth-Realm: olegh.ath.cx MWI-Account: 1012(a)192.168.1.5 When he calls me, conversation is OK. When I (from the local phone) call him (he is so far, and behind NAT), ring is OK, but when he picked up handset -- we both heard nothing. Looks like FS try to establish connection to IP address, assigned him behind his NAT: 192.168.1.146. FS sends ARP request to resolve this address: 6191 packets received by filter 0 packets dropped by kernel [root@olegh /usr/local/freeswitch/conf]# tcpdump host '(91.207.244.1 or 192.168.1.146)' tcpdump: verbose output suppressed, use -v or -vv for full protocol decode listening on xl0, link-type EN10MB (Ethernet), capture size 96 bytes 00:49:27.911731 IP deskpro.khovayko.com.sip > 91.207.244.1.sip: SIP, length: 1339 00:49:28.299781 IP 91.207.244.1.sip > deskpro.khovayko.com.sip: SIP, length: 349 00:49:28.317358 IP 91.207.244.1.sip > deskpro.khovayko.com.sip: SIP, length: 403 00:49:40.420237 IP 91.207.244.1.sip > deskpro.khovayko.com.sip: SIP, length: 829 00:49:40.451060 IP deskpro.khovayko.com.sip > 91.207.244.1.sip: SIP, length: 368 00:49:40.691549 arp who-has 192.168.1.146 tell deskpro.khovayko.com 00:49:41.107754 arp who-has 192.168.1.146 tell deskpro.khovayko.com 00:49:42.105717 arp who-has 192.168.1.146 tell deskpro.khovayko.com 00:49:43.107377 arp who-has 192.168.1.146 tell deskpro.khovayko.com 00:49:44.106215 arp who-has 192.168.1.146 tell deskpro.khovayko.com 00:49:45.107216 arp who-has 192.168.1.146 tell deskpro.khovayko.com 00:49:46.105957 arp who-has 192.168.1.146 tell deskpro.khovayko.com 00:49:47.105975 arp who-has 192.168.1.146 tell deskpro.khovayko.com 00:49:48.105804 arp who-has 192.168.1.146 tell deskpro.khovayko.com 00:49:49.107058 arp who-has 192.168.1.146 tell deskpro.khovayko.com 00:49:50.107512 arp who-has 192.168.1.146 tell deskpro.khovayko.com 00:49:50.533533 IP deskpro.khovayko.com.sip > 91.207.244.1.sip: SIP, length: 626 00:49:50.794874 IP 91.207.244.1.sip > deskpro.khovayko.com.sip: SIP, length: 355 About SIP trace: I do not know, how to save trace to file from fs_cli. On the sccreen, there is too many text lines, they very fast gone... Brian West wrote: > You don't hav eto do this if you have upnp freeswitch will do this for you even for the media ports... I am not sure what your problem is grab the default config and use it... I know it works if you really do have upnp. > /b > > On May 24, 2010, at 8:01 PM, Oleg Khovayko wrote: > > >> It is behind UPNP, and I forwarded ports 5060-5060 (both TCP+UDP) to server. >> Also, I have forwarded ports 20000~30000, and for FS setup in the >> file autoload_configs/switch.conf.xml >> >> <param name="rtp-start-port" value="20000"/>  >> <param name="rtp-end-port" value="25000"/> >> >

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FreeSWITCH Architecture
by Jan Berger 26 May '10

26 May '10

hi, I would like to address some changes to FreeSWITCH, but the challenge I face is that I am not sure where and how I should do this. The objective with this phase is to throw some ideas around and get feedback before I consolidate the architecture and move on coding. In an ordinary project it would be a few engineers that made a draft, made a presentation and evolved the draft through the discussions that followed until a majority accepted the draft. Jan

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