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Use java nio (netty) library to get fd (FileDescriptor)
by Marwan Idriss 10 Apr '13

10 Apr '13

Greeting, How I can use java nio (netty) library mentioned in http://wiki.freeswitch.org/wiki/Java_ESL_Client to get fd (FileDescriptor), that is need to make new connection to FreeSwitch Regards Marwan

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Freetdm Channel utilization in order(ascending/descending)
by Vishal Kakkar 10 Apr '13

10 Apr '13

[span wanpipe wp1] trunk_type => e1 group=dialGrp1 b-channel => 2:1-15 b-channel => 2:17-31 d-channel => 2:16 [span wanpipe wp2] trunk_type => e1 group=dialGrp1 b-channel => 1:1-15 b-channel => 1:17-31 d-channel => 1:16 freetdm/dialGrp1/*a*/xxxxx initiates call from channel 1 of span 2 i.e. (2:1) freetdm/dialGrp1/*A/*xxxxx initiates call from channel 31 of span 2 i.e. (2:31).. but i was expecting it to be 31st channel of Span1. Also 2nd question in above config all calls are going through wp1(span2) only.. even if i initiate more than 30 calls simultaneously.. no call goes to wp2(span 1) Can anyone please help how can i make ascending/descending channel utilization using one single group across both above spans.. Thanks a lot.. FS Rocks.

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loopback
by Steven Schoch 10 Apr '13

10 Apr '13

In the default dialplan, it bridges a call to voicemail using these actions: <action application="answer"/> <action application="sleep" data="1000"/> <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> I have found the the following also seems to work: <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/> My question is why does the default dialplan use a bridge to loopback instead of just calling the voicemail app directly? What is the difference? -- Steve

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Lost connection during voicemail session
by Sean Devoy 09 Apr '13

09 Apr '13

HI, I had a user say she got disconnected from voicemail and wanted to know why. I saw in the log where she called in, listened, pressed 1, listened again and then: 2013-04-09 11:14:37.134093 [DEBUG] switch_ivr_play_say.c:247 Handle play-file:[voicemail/vm-listen_saved.wav] (en:en) 2013-04-09 11:14:37.134093 [DEBUG] switch_ivr_play_say.c:1332 Codec Activated L16@16000hz 1 channels 20ms 2013-04-09 11:14:38.094065 [DEBUG] switch_core_session.c:975 Send signal sofia/external/<ext>@<our dom> [BREAK] 2013-04-09 11:14:38.094065 [DEBUG] switch_core_session.c:975 Send signal sofia/external/<ext>@<our dom> [BREAK] 2013-04-09 11:14:38.094065 [DEBUG] switch_core_session.c:975 Send signal sofia/external/<ext>@<our dom> [BREAK] 2013-04-09 11:14:38.114065 [DEBUG] sofia.c:5599 Channel sofia/external/<ext>@<our dom> entering state [terminated][487] 2013-04-09 11:14:38.114065 [DEBUG] switch_channel.c:2994 (sofia/external/<ext>@<our dom>) Callstate Change EARLY -> HANGUP 2013-04-09 11:14:38.114065 [NOTICE] sofia.c:6383 Hangup sofia/external/<ext>@<our dom> [CS_EXECUTE] [ORIGINATOR_CANCEL] So if FS says she hung up and she says FS hung up, is it safe to say her internet connection failed momentarily? Also, why "Callstate Change EARLY -> HANGUP". Shouldn't it be "Callstate Change ACTIVE -> HANGUP"? Thanks, Sean

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CDR
by Khue Nguyen Minh 09 Apr '13

09 Apr '13

Hi all, I want create cdr (filetype: csv) file in freeswitch. But, I don't known how I can do it from xml dialplan. Please guide me. Thanks & Best regards, Khue.

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pstn call control from computer
by Zhnupy Gonzalez 09 Apr '13

09 Apr '13

Hi I'm a total newbie looking a way to controll PSTN calls from my computer, by control I mean: see caller id on screen, answering with softh phone, pressing a button to play a song, that kind of stuff. Is it possible with hardware like grandstream Ht503 or obihai OBi110 or do I need more (and more expensive) hardware? regards zhnu

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Voicemail Logfile questions.
by Sean Devoy 09 Apr '13

09 Apr '13

HI, I had a user say she got disconnected from voicemail and wanted to know why. I saw in the log where she called in, listened, pressed 1, listened again and then: 2013-04-09 11:14:37.134093 [DEBUG] switch_ivr_play_say.c:247 Handle play-file:[voicemail/vm-listen_saved.wav] (en:en) 2013-04-09 11:14:37.134093 [DEBUG] switch_ivr_play_say.c:1332 Codec Activated L16@16000hz 1 channels 20ms 2013-04-09 11:14:38.094065 [DEBUG] switch_core_session.c:975 Send signal sofia/external/<ext>@<our dom> [BREAK] 2013-04-09 11:14:38.094065 [DEBUG] switch_core_session.c:975 Send signal sofia/external/<ext>@<our dom> [BREAK] 2013-04-09 11:14:38.094065 [DEBUG] switch_core_session.c:975 Send signal sofia/external/<ext>@<our dom> [BREAK] 2013-04-09 11:14:38.114065 [DEBUG] sofia.c:5599 Channel sofia/external/<ext>@<our dom> entering state [terminated][487] 2013-04-09 11:14:38.114065 [DEBUG] switch_channel.c:2994 (sofia/external/<ext>@<our dom>) Callstate Change EARLY -> HANGUP 2013-04-09 11:14:38.114065 [NOTICE] sofia.c:6383 Hangup sofia/external/<ext>@<our dom> [CS_EXECUTE] [ORIGINATOR_CANCEL] So if FS says she hung up and she says FS hung up, is it safe to say her internet connection failed momentarily? Also, why "Callstate Change EARLY -> HANGUP". Shouldn't it be "Callstate Change ACTIVE -> HANGUP"? Thanks, Sean

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Java ESL example for outbound connection
by Marwan Idriss 09 Apr '13

09 Apr '13

Greeting, I am trying to use Java API to control outbound socket, But I am not finding any example especially about this idea : listen for an incoming connection on a socket, accept the incoming connection from FreeSWITCH, fork a new copy of your process if you want to listen for more connections, and then pass the file number of the socket to new($fd). so if you can provide me with can example java example for using new($fd) is enough thanks for your effort

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Re: [Freeswitch-users] problems with freeswitch + zrtp in proxy-media mode
by Eli Burke 09 Apr '13

09 Apr '13

Yes, inbound-zrtp-passthru is true. I think it gets set automatically when you turn on ZRTP. The config I'm testing with is a stock 'enable_zrtp=yes' build of HEAD circa last week, with proxy-media=true added to Local_Extension in the default dial plan. -Eli On Feb 1, 2013, at 6:00 AM, royj(a)yandex.ru wrote: > Is there <param name="inbound-zrtp-passthru" value="true"/> in profile > > On Thu, 31 Jan 2013 17:43:34 -0500 > Eli Burke <eburke(a)edge-net.net> wrote: > >> I'm trying to use Freeswitch with an open source voip client called Linphone, but cannot manage to establish functioning ZRTP sessions. The problem *seems* to be related to the RTP stream's SSRC. I'm trying to conclusively determine if the problem lies with Linphone, Freeswitch, or one of the underlying zrtp implementations (libzrtp and zrtpcpp). >> >> Here's what I've determined so far: >> 1) ZRTP support is compiled and functioning. I can establish ZRTP sessions between two iPhones running Groundwire. >> 2) linphone + ZRTP works correctly if I make a one legged call to a Freeswitch echo extension >> 3) linphone + ZRTP works correctly if I turn on bypass-media >> 4) linphone gets a CRC error when trying to verify the ZRTP packet on any two-legged call >> 5) disabling the CRC check results in a successful ZRTP session, but the audio is extremely garbled >> >> Is Freeswitch modifying the ZRTP packets when running in proxy mode? If so, how is it that some clients are able to handle this situation?

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Connecting Internal IP phones to PSTN
by Ebrahim Bararian 09 Apr '13

09 Apr '13

Hi, I have some soft IP phones in an internal local network. I want to connect these IP phones to PSTN via 2 PSTN lines. How many ways are there to solve this problem? Which hardwares should I buy?

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