May 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Bind to multiple ports with Sofia
by Emrah 07 May '13

07 May '13

Hi guys, I need to bind to multiple ports with Sofia. If I start one profile per port, I cannot get a consistent registration table working. A domain can only be aliased to one profile if my understanding is correct. If you have a working set up, I'd appreciate some advice. Thanks, Emrah

5 10

FreeSWITCH Weekly News and Notes
by Michael Collins 06 May '13

06 May '13

Happy first May Monday to all. We are please to let everyone know that the new FreeSWITCH book<http://www.packtpub.com/freeswitch-1-2/book>is nearing completion. Each day the Packt editors have been sending us revised and re-revised chapters to polish off. If we keep up our current pace then it is quite likely the book will be available in early June. We hope you enjoy it. On last week's call <http://wiki.freeswitch.org/wiki/FS_weekly_2013_05_01>we enjoyed a nice presentation from Omar over at OrecX <http://www.orecx.com> who showed us some of the useful features of their call recording solutions. They have an open source version as well as commercial versions. If you are looking for a call recording solution then we recommend that you review what OrecX has to offer. This week <http://wiki.freeswitch.org/wiki/FS_weekly_2013_05_08> we continue in our series of presentations on FreeSWITCH-compatible software applications. We look forward to having Dan Bogos from CGRateS<http://www.cgrates.org/>join us to talk about their project. Click here <http://cgrates.readthedocs.org/en/latest/introduction.html> to get a quick look at what CGRateS does, and then make plans to join our call on Wednesday. Regarding ClueCon 2013 we would like to remind everyone that we are still accepting talk proposals. If you have a talk idea please let us know. In the meantime feel free to get registered <http://cluecon.com/register/> and book your hotel room at the Hyatt. Have a great week! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

1 0

WiFi IP VOIP SIP phones
by Sean Devoy 06 May '13

06 May '13

HI, Can anyone recommend a good quality WiFi (wireless IP) VOIP Phone that works with FS? Looking for < $200 and top quality functionality. Thanks, Sean

12 15

Early Media indication from B in CHANNEL_STATE event
by David Wilkie 06 May '13

06 May '13

Hi, Is there a way to get an early media indication from a B-leg in the CHANNEL_STATE event. According to http://wiki.freeswitch.org/wiki/Event_list#CHANNEL_STATE there doesn't seem to be a variable which indicates early media from B. We need to use this information to figure out whether to join B to A or generate early media and join on answer. If there is no way to get this info can we query it using Event Socket?

1 0

FS transcoding
by thomas peterseil 06 May '13

06 May '13

hello, i am running a new FS and i have two sip clients registered, one which supports G.722 and the other one supports G.711. i am still running mostly the default config, both codecs are listed when i do a show codecs. when i call now from 1000 to 1001 with different codecs it doesnt work. but i can call from both clients with different codecs to a 3300 conference. is there something i have to enable in the default config to activate transcoding? thanks for your help! thomas

4 4

ekiga to flex client call not happening
by Navnath Sonavne 06 May '13

06 May '13

Hi All, 1.Flex client to Ekiga : I can make call from flex client given in freeswitch souce to ekiga(h323 protocol) softphone successfully. Flex client is registered on freeswitch as 1103.xml extension and ekiga phone is registered as 1104.xml extension. Here is dialplan for flex to ekiga call : (These actions are written for testing purpose only not a permanent solution.) <extension name="Local_Extension" continue="true"> <condition field="destination_number" expression="^(1104)$"> <action application="export" data="dialed_extension=$1"/> <action application="set" data="effective_caller_id_number=${dialed_extension}"/> <action application="set" data="PEER_IP=192.168.8.27"/> <action application="set" data="call_timeout=30"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="bypass_media=true"/> <action application="bridge" data="h323/$1@${PEER_IP}"/> </condition> </extension> 2.Ekiga to Flex client : Now I am trying to make call from my Ekiga(h323 protocol) softphone to flex client given in freeswitch souce but it is failing every time. Flex client html page don't show any dialog box for incoming call but its javascript code contains code for dialog for incoming call. Does that flex client is made only to make outging call?It can make outgoing call to ekiga phone. Isn't it made for incoming call? If that flex client is also made for incoming call too then please tell how to make call from ekiga to flex client. Here is dialplan for ekiga to flex call: (These actions are written for testing purpose only not a permanent solution.) <extension name="Local_Extension" continue="true"> <condition field="destination_number" expression="^(1103)$"> <action application="export" data="dialed_extension=$1"/> <action application="set" data="effective_caller_id_number=${dialed_extension}"/> <action application="set" data="PEER_IP=192.168.9.165"/> <action application="set" data="call_timeout=30"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="bypass_media=true"/> <action application="bridge" data="${rtmp_contact($${rtmp_profile}/${dialed_extension}@$${domain_name})}"/> </condition> </extension> -Navnath.

2 1

Any one with bluebox working?
by mehroz 06 May '13

06 May '13

Hi, I have recently installed bluebox for FS GUI ! I have no issue with audio calling, but video calling is messed up. Codecs have been changed and the same set of configuration works on Local LAN (no NAT), however, it is troublesome with NAT! Tried all combination of NAT settngs in SIP profile setting but none of it works. Please respond if anyone undergone with this issue. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Any-one-with-bluebox-working-… Sent from the freeswitch-users mailing list archive at Nabble.com.

1 0

[ERR] ftdm_io.c:5740 Failed to load module type: libpri
by Ashish gautam 06 May '13

06 May '13

Hi, I am using FS on freeBSD with dahdi-libpri-ftdm stack. I am getting this error when I try to load mod_ftdm. dahdi_scan shows all channels ok. 2013-05-06 11:41:26.456705 [ERR] ftdm_io.c:5482 Error loading /usr/local/freeswitch/mod/ftmod_libpri.so [Cannot open "/usr/local/freeswitch/mod/ftmod_libpri.so"] 2013-05-06 11:41:26.456705 [ERR] ftdm_io.c:5740 Failed to load module type: libpri 2013-05-06 11:41:26.456705 [ERR] mod_freetdm.c:4076 Error configuring FreeTDM span FreeTDM1 Kindly help, Its really urgent. Regards, Ashish

1 0

ESL Client Library Managed wrappers (Windows)
by Dave R. Kompel 04 May '13

04 May '13

I just spent the good part of this wekend trying to get them compilied, tested and working, it's been over 2 years since I had to build them, and it's a bitch. To save ppl time, I put the binaries up, and made them available if anyone needs them. You can grab them at: http://download.drknetworking.com/eslmanaged.zip --Dave

6 14

Is it possible to record a conference with only one attendee?
by Joel Rosenfield 04 May '13

04 May '13

I am starting/stopping the conference recording for an existing conference using the api commands "conference <confname> recording start/stop <file_path>". I have didn't find a conference flag that would cause the audio to be recorded when there is just one person in the conference. Did I miss something? I did see that the auto-record flag only starts recording when there are two attendees, and I wonder if what I am observing when trying to record with only one attendee happens for a similar reason. I did notice that when I played an announcement on a 1-person conference, that announcement does get recorded: conference <confname> play <file_path> My scenario for when I start a recording is this with just myself on the conference is this: (1) play an annoucement (2) I say something (3) wait for someone to join and say I some more. In my recording file, I hear the announcement (1), then silence, then I hear the audio from the two-person conference (3). I do not hear my own voice (2). The use case is that when people try the recording feature, the first thing they want to do is try it out themselves before they adopt it. Thanks, - Joel

1 0