October 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Conference floor given to muted member
by Duvid Rottenberg 10 Oct '11

10 Oct '11

I am setting up a conference to use for paging similar to http://wiki.freeswitch.org/wiki/Lua_Intercom and I noticed that the first 2 seconds of audio are not being sent. After reviewing the logs it appears that the first member to enter the conference is automatically given the floor event thought it was entered with the mute flag. I set it up so that the originator enters the conference with the moderator flag and is also unmuted, however it appears that he doesn't get the floor right away, even though he's the only unmuted member. It takes about 2 seconds of speech for the moderator to get the floor. I have the following settings for the conference profile I use: <param name="energy-level" value="0"/> <param name="conference-flags" value="wait-mod"/> <param name="moh-sound" value="silence"/> <param name="moh-sound" value="silence"/> I tried setting <param name="member-flags" value="waste"/> but it didn't help, for now I changed it so that the moderator enters the conference with the wast flag. Are there any other settings I can use to either avoid giving a muted member the floor, or to speed up the time it takes to give the floor to the moderator? Thank You, Duvid Rottenberg

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Antw.: SCA Hunt Group
by Weigel, Stefan 10 Oct '11

10 Oct '11

Hi Spencer, just use mod_limit to limit one call per extension. http://wiki.freeswitch.org/wiki/Mod_limit Best regards, Stefan ----- Reply message ----- Von: "Spencer Thomason" <spencer(a)5ninesolutions.com> Datum: Mo., Okt. 10, 2011 19:29 Betreff: [Freeswitch-users] SCA Hunt Group An: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> Hello all, I'm working on replacing a legacy key system with FreeSWITCH and a few Cisco SPA-509G phones. I have SCA/SLA working correctly but I was wondering if anyone has a suggestion for a good way to make the inbound calls to the shared extensions "roll over" like a traditional phone system. I have a simple lua script that does a sequential dial and checks for a busy response from the phones but the problem is that these Cisco phones have a minimum of 2 calls per line. This results in the second call going to the same "line" as the first. What I'd like to do is have an inbound call pick the next available shared extension. Does anyone have any suggestions? Is there some variable I could check with a script that would show whether the shared extension is in use? Thanks, Spencer 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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Inband SPS/PPS missing because of private/public IP.
by RuiQiang Huang 10 Oct '11

10 Oct '11

Hi, I tried to get video working between one Grandstream and Polycom IP phone with H.264. I'm planning to make freeswitch as SBC for my internal network. However, both proxy media a d pass-thru media are getting one way video. That is Polycom VVX can see Grandstream, not the other way. I sniffer and found out that Polycom is doing inband SPS/PPS, so the first packet or so is for SPS/PPS, without this, the other-side cannot decode the video. But freeswitch media proxy doesn't forward packets to right public address port until it receives packets from the other stream, which will trigger a RTP auto adjustment. I just wondering is there any config to fix this or I have to modify the code? Thanks.

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SCA Hunt Group
by Spencer Thomason 10 Oct '11

10 Oct '11

Hello all, I'm working on replacing a legacy key system with FreeSWITCH and a few Cisco SPA-509G phones. I have SCA/SLA working correctly but I was wondering if anyone has a suggestion for a good way to make the inbound calls to the shared extensions "roll over" like a traditional phone system. I have a simple lua script that does a sequential dial and checks for a busy response from the phones but the problem is that these Cisco phones have a minimum of 2 calls per line. This results in the second call going to the same "line" as the first. What I'd like to do is have an inbound call pick the next available shared extension. Does anyone have any suggestions? Is there some variable I could check with a script that would show whether the shared extension is in use? Thanks, Spencer

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Break command not working
by Chris Martineau 10 Oct '11

10 Oct '11

Hi, I have the following scenario... application='preanswer' application='set' data='execute_on_answer=break' application='sched_broadcast' data='+1 /path/file' application='bridge' data='sofia...' This works fine in that it pre answers the call and plays back a welcome message prior to normal early media ringtone. The problem I have is that if the call is answered quickly the playback just continues until finished. I would have thought that application='set' data='execute_on_answer=break' should stop the playback on answer but the break command doesn't have any effect. I have tried break all to no effect. Should this work? Any ideas on how to make this work? Many thanks Chris

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Sending DTMF on B-leg
by Wayne 10 Oct '11

10 Oct '11

Hello All, I need to call out on a SIP trunk and when the call is answered send DTMF tones. I need to send the DTMF only on the outbound leg. Does anyone have a dialplan that will do that? Is it possible. I have only found one thread on it and did get much out of it. Thanks Wayne

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Half the call recording
by Hynek Cihlar 10 Oct '11

10 Oct '11

I cannot find information on recording half the call, only a page which states, that split recording of the caller/callee is possible with freeswitch. Reading the documentation about recording API commands I couldn't find any info, whether freeswittch can record only one leg of a two-legged called or even how to do it. Any help is appreciated. Hynek

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There is no voice in Two party call
by sirvan paraste 10 Oct '11

10 Oct '11

Dear all After regard for your attention to me I am student how want work on freeswitch. I have read freeSwitch documents (introduction, how to install, use default confuguration,...). for start of using freeswitch, I make three computer network together with a hub, after that,I install freeswitch on one of them and X-Lite on another computers, network checked with sharing file and ping issue. After registering of two X-lite hone with No. 1001,1002, they can call each other but there is no voice !!! :( i have checked and did some work but the problem is not solved. Can you help me? thank you.

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Re: [Freeswitch-users] mod_sofia@
by Mi Ke 09 Oct '11

09 Oct '11

found it on wiki - it's sip_contact_user variable ----- Original Message ----- From: Mi Ke Sent: 10/08/11 09:33 PM To: Freeswitch Users Subject: [Freeswitch-users] mod_sofia@ Hi ! I originate call from lua script by the following command: local ob_session = freeswitch.Session("{originate_timeout=120,return_ring_ready=true,origination_caller_id_number=" .. calling_id .. ",called_id=" .. called_id .. ",accountcode=" .. accountcode .. ",credit_time=" .. out_credit_time .. ",execute_on_answer=lua b_answer.lua}" .. out_called_id,session) My iface IP is 2.2.2.2, origination_caller_id_number is my calling ID (55555) and out_called_id = sofia/internal/1212121212(a)1.1.1.1 Leg B setup and bridging goes normally, when remote party disconnects, we receive: tport_wakeup_pri(0x81c1a0): events IN tport_recv_event(0x81c1a0) tport_recv_iovec(0x81c1a0) msg 0x7f28386348a0 from (udp/2.2.2.2:5060 http://78.46.44.138:5060/ ) has 425 bytes, veclen = 1 tport_deliver(0x81c1a0): msg 0x7f28386348a0 (425 bytes) from udp/1.1.1.1:5060/sip http://193.104.107.7:5060/sip next=(nil) nta: received BYE sip:mod_sofia@2.2.2.2: http://sip:mod_sofia@78.46.44.138:5060/ 5060 http://sip:mod_sofia@78.46.44.138:5060/ SIP/2.0 (CSeq 48765) nta: canonizing sip:mod_sofia@2.2.2.2:5060 http://sip:mod_sofia@78.46.44.138:5060/ with contact nta: BYE (48765) going to existing leg nua: nua_stack_process_request: entering nua(0x94c4a0): event i_bye 100 Trying nua: nua_application_event: entering 2011-10-08 14:54:48.866309 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/ 841266482056(a)193.104.107.7 .1.1.1 [BREAK] nua: nua_handle_magic: entering 2011-10-08 14:54:48.886307 [DEBUG] switch_channel.c:2800 (sofia/internal/ 841266482056(a)193.104.107.7 .1.1.1) Callstate Change ACTIVE -> HANGUP (normal bye handshake follows...) Should remote party see me as sip:55555@2.2.2.2 and not as sip:mod_sofia@2.2.2.2 ? What info should I include into session originate to acheive that? Thanks / Mike

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Failed to get callee number during the call
by fieldpeak 09 Oct '11

09 Oct '11

Hello Friends, i would like to get the callee number during the call, i used the function 'switch_channel_get_variable', however, when i display it to the console, it shows 'null', i tried call this func before and after answering the call, both failed, could anyone help advise if i used the correct function or how to realzize it, thanks. source code as below, const char* in_calledstation; in_calledstation = switch_channel_get_variable(channel, "sip_callee_id_number"); switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "in_username: %s, in_calledstation: %s, in_uniqueid: %s, sessiontime: %d \n", in_username, in_calledstation, in_uniqueid, sessiontime); -- Regards, Charles

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