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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

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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.

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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.

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[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

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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

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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

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Re: [Freeswitch-users] How to retreive FreeSWITCH-IPv4 value ? (Julien Terrasson)
by julien terrasson 04 May '13

04 May '13

Hi Michael, I'm actually using a single IP address. But i'm trying to think of an architecture where multiple freeswitch would be in use, each of them acting as user-controled recorders. The reason i wanted to have the Freeswitch-IPv4 from the dialplan was to be able to include it when defining the record name : Ex : <FreeSWITCH-IPv4>-Record_<caller_id>_<sequence_nb>.wav That way i can get the information of where the recording took place. But in the end i used a workaround : i defined a server_id var in the vars.xml.. But anyway, thanks for the prompt feedback ! Have a nice week-end. Julien ------------------------------------------------------------------------------------------------------------------------------------ Julien, "FreeSWITCH-IPv4" is not a channel variable. Rather, it is an event header. Even so, I'm not sure that an individual event is necessarily the best place from which to be extracting this value. What problem are you trying to solve? Also, how many IP addresses do you have on your system? -MC

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Voicemail greeting length?
by Nathan Neulinger 03 May '13

03 May '13

Seeing a problem where recording a voicemail greeting is truncated on freeswitch. The strange thing is - FS acts like it's recording the entire duration I'm speaking (30-40 seconds), but when you play back the recording, it stops after 16-17 seconds fairly consistently. (Same thing if I tell it to save and then go and look at the actual wav file.) There's no indication of anything happening in the logs. I'm only seeing the truncated recording with calls to VM from skinny phones. It does not affect voicemail message recording at all, only greetings. This is with current trunk. Any suggestions on places to start looking? -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

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Alarm, KISS OFF
by Miha 03 May '13

03 May '13

Hi, I have problem regarding alarm device. Alarm guys are saying that I am not detecting kiss off tone or something like that. Is is send on 1400Hz. I have been trying to detect this tone with mod spandsp but with no luck. has some manage to get this working or getting this to work on FS? Thanks! Miha

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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 83, Issue 10
by Navnath Sonavne 03 May '13

03 May '13

Hi, As you said create new extensions for flex client and h323 endpoint(Ekiga softphone). I created two new extensions 1103.xml for flex client and 1104.xml for ekiga soft phone. I login using 1103 on flex client and make call to 1104. Now when I call from flex client(1103) to Ekiga softphone(1104),ekiga phone rings and when i answer the call ,call terminates immediately by showing local user rejected call. In another case where I call from Ekiga phone(1104) to flex client(1103),it says person at 1104 is not available and says record your voice mail.In this case flex client did not get any incoming call. Here is my default.xml <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="bypass_media=true"/> <action application="bridge" data="user/${dialed_extension}@${domain_name}"/> </condition> <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="bypass_media=true"/> <action application="bridge" data="user/${dialed_extension}@${domain_name}"/> </condition> </extension> Correct me if my dialstrings are wrong. Tell me if any other changes to made in other files also. How can I make both way calling ?Please help me its emergency , I have demo next week. -Navnath. ________________________________ From: "freeswitch-users-request(a)lists.freeswitch.org" <freeswitch-users-request(a)lists.freeswitch.org> To: freeswitch-users(a)lists.freeswitch.org Sent: Thursday, 2 May 2013 8:59 PM Subject: FreeSWITCH-users Digest, Vol 83, Issue 10 ----- Forwarded Message ----- Send FreeSWITCH-users mailing list submissions to freeswitch-users(a)lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request(a)lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner(a)lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." Today's Topics: 1. make ekiga to flex call (Navnath Sonavne) 2. Re: make ekiga to flex call (Brian Foster) 3. Re: Check if UA is still there? (mehroz) 4. Re: Check if UA is still there? (Ken Rice) 5. Re: Check if UA is still there? (mehroz) 6. Re: Check if UA is still there? (Michael Collins) Hi All, I am using flex client(at 192.168.9.165) given in freeswitch source to make call to Ekiga(h323 protocol) softphone(at 192.168.8.27) via freeswitch server(at 192.168.8.41). I have two users registered on freeswitch with extension as 1100 and 1101 in default context. I loged in using one of extension in flex client,then I dail another extension. After I dail,call goes to Ekiga softphone at 192.168.8.27 because I have made changes in default.xml accordingly to forward calls to this ip. Here is My default.xml part : <extension name="Local_Extension"> <condition field="destination_number" expression="^(\d{4})$"> <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="proxy_media=true"/> <action application="bridge" data="h323/$1@${PEER_IP}"/> </condition> </extension> Ekiga phone can answer the call and there is audio transmission from both end successfully. But this is one way calling i.e. from flex to ekiga. Now I want to call from Ekiga softphone to flex client. How to call flex client from ekiga softphone? Anybody please me on this issue. -Navnath. According to your dialplan you're forcing all calls that hit Local_Extension (any call with a destination number of 4 digits) to call your ekiga softphone. Your other problem lies in the fact that you should have different dialplans for different methods of calling unlike endpoints one for h232 one for flex. See below: -BDF On May 2, 2013 9:05 AM, "Navnath Sonavne" <navnath.sonavne(a)yahoo.com> wrote: > > Hi All, > > I am using flex client(at 192.168.9.165) given in freeswitch source > to make call to Ekiga(h323 protocol) softphone(at 192.168.8.27) via freeswitch server(at 192.168.8.41). > I have two users registered on freeswitch with extension as 1100 and 1101 in default context. > I loged in using one of extension in flex client,then I dail another extension. > After I dail,call goes to Ekiga softphone at 192.168.8.27 because I have made changes > in default.xml accordingly to forward calls to this ip. > Here is My default.xml part : > > <extension name="Local_Extension"> > <condition field="destination_number" expression="^(\d{4})$"> > <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"/> <<Bad: you should never hardcode a specific endpoint IP if this dialplan should be dialing others as well > <action application="set" data="call_timeout=30"/> > <action application="set" data="hangup_after_bridge=true"/> > <action application="set" data="proxy_media=true"/> > <action application="bridge" data="h323/$1@${PEER_IP}"/> <<Bad: you set your dialplan to bridge the call to your softphone > </condition> > </extension> > > Ekiga phone can answer the call and there is audio transmission from both end successfully. > > But this is one way calling i.e. from flex to ekiga. > Now I want to call from Ekiga softphone to flex client. > How to call flex client from ekiga softphone? > > Anybody please me on this issue. > > -Navnath. > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > What if we want to even hungup the channels associated with those users? does freeswitch helps ? Like, i have a scenario in which , an active call disrupts , once BOTH users gets disconnected from network. Here, both users gets unregistered after option messages but the call sustains. Is there any thing i can handle this scenario? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Check-if-UA-is-still-there-tp… Sent from the freeswitch-users mailing list archive at Nabble.com. options pings are really just for registered users. a user does not have to be registered to place a call. more useful in your scenario is rtp timers and session timers Ken Sent from my iPad On May 2, 2013, at 8:49, mehroz <mehroz.ashraf85(a)gmail.com> wrote: > What if we want to even hungup the channels associated with those users? does > freeswitch helps ? > > Like, i have a scenario in which , an active call disrupts , once BOTH users > gets disconnected from network. > Here, both users gets unregistered after option messages but the call > sustains. Is there any thing i can handle this scenario? > > > > > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Check-if-UA-is-still-there-tp… > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org i cannot use session-timers, disabled due to an issue with video calling.... and rtp timers just does not seems to be working! i have But once network disconnects from both clients (works good, with a single client network outage), FS does not hangs up the call ..... -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Check-if-UA-is-still-there-tp… Sent from the freeswitch-users mailing list archive at Nabble.com. On Thu, May 2, 2013 at 8:13 AM, mehroz <mehroz.ashraf85(a)gmail.com> wrote: i cannot use session-timers, disabled due to an issue with video calling.... > >and rtp timers just does not seems to be working! > Can you elaborate on this? Do you have a full pcap and FS debug log of a call with RTP in both directions and then the incoming RTP stream stopping and yet the call not being disconnected? >i have > >But once network disconnects from both clients (works good, with a single >client network outage), FS does not hangs up the call ..... > Correct. No SIP signaling and no session timers means there's nothing left to tell FS that the call should be torn down, except for the RTP timers mentioned above. That's why you'll need to supply a complete debug w/ pcap in order to find out why RTP timers are not working. -MC -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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