January 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_event_multicast bind interface question
by Борисов, Дмитрий / Dmitriy Borisov 26 Jan '15

26 Jan '15

I have freeswitch installed on server with two etehrnet interfaces. My profiles are started on eth1, but multicast must be sent from eth0. Is it possible? And how? -- with best regards, Dmitriy Borisov

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callcenter mod rtmp cant receive calls...
by jdayola＠spingine.com 26 Jan '15

26 Jan '15

Hi! Im using mod callcenter. I have 3 agents on queue 600 and assigned them extensions 601,602 and 603. Everything was working fine when I tested it using xlite and zoiper. But on production we have our own softphone based on flash so we need to use rtmp. I already enabled rtmp but I don't receive calls on my extensions using rtmp. Like if I assign 601 and 602 to xlite and zoiper and 603 on our flash client only 601 and 602 can receive calls. Do I need to make a new dial plan if I use RTMP? Im still new to voip and Im using fusionpbx as my freeswitch gui. Regards, John D.

1 0

Playing file using lua before bridge
by John Nash 25 Jan '15

25 Jan '15

I am trying to play some native files (Pre encoded) using lua script just after call is setup using Bridge (But before any answer). I need to use "Proxy_Media" (Want media to go through freeswitch but no codec translation). I tried using "bridge_pre_execute_aleg_app" but I do not hear sound files being played (Do not see any error message also). I have already tested lua script in some other test extension and it works fine. <extension name="main_route"> <condition field="destination_number" expression="^rb-(.+)$"> <action application="log" data="INFO progressing call..."/> <action application="set" data="proxy_media=true"/> <action application="set" data="bridge_pre_execute_aleg_app=lua"/> <action application="set" data="bridge_pre_execute_aleg_data=tt.lua ${credit_time}"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="execute_on_answer=sched_hangup +${credit_time}" /> <action application="bridge" data="sofia/$${domain}/${regex(${sip_req_uri}|^kb-(.+)$|%1)};fs_path=sip:${sip_received_ip}:${sip_received_port}"/> </condition> </extension>

2 1

voicemail silence_detection is always detecting silence
by Raphael Lechner 25 Jan '15

25 Jan '15

Hi, I configured an extension that first call for some seconds a phone and if nobody is picking up, the caller is hearing a playback and can press 1 for leaving a voicemail and 2 to get redirected to a mobile phone. The Problem is that in my test environment with a SIP Provider everything works fine after tuning the voicemail.conf.xml to <param name="record-silence-threshold" value="200"/> <param name="record-silence-hits" value="10”/> But on the production machine we are using a patton as a ISDN Gateway and after pressing 1 for the voicemail I always got "Message is less than minimum record length: 3” on the calling phone. After that I can record a message and that works as expected. Is there a way do disable the silence_detection or any hint what I can change? I tried changing the silence-threshold to 1,50 and silence-hits to 300,30000 but nothing has changed Debug Log https://pastebin.freeswitch.org/23851 <https://pastebin.freeswitch.org/23851> The called python script: def handler(session, args): voicemail = args.split(' ')[0] dtmf_pressed = args.split(' ')[1] forward_number = args.split(' ')[2] callerid = session.getVariable("caller_id_number") callername = session.getVariable("caller_id_name").lstrip() if dtmf_pressed == '1': send_sms('377XXXXXXX’,’New Voicemail from %s %s' % (callername, callerid)) session.execute("export", "skip_greeting=true") session.execute("export", "skip_instructions=true") session.execute("answer") session.execute("voicemail", "default 192.168.17.252 10”) #session.execute("bridge", "loopback/app=voicemail:default %s %s" % (conf['network']['ip'],voicemail)) elif dtmf_pressed == '2': consoleLog( "info", "Call is forwarded to %s\n" % forward_number) session.transfer(forward_number, "XML", "default") else: consoleLog( "info", "DTMF received is %s and not 1 or 2.Hangup Call\n" % (dtmf_pressed)) session.hangup() Thank you, Raphael

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Follow me implementation
by Brian Foster 25 Jan '15

25 Jan '15

I've been given an assignment. It's a little rough, and honestly I've been working on other projects and at the same time loosing my freeswitch-fu. So, here it goes. Company owner wants to be able to implement a follow me function. He's asking for the deskphones to begin ringing, then have cell phones ring N seconds later WHILE the deskphones continue to ring. The function has to be able to work using a couple different ways of dialing (we've got call groups implemented, http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#Groups) When the mobile phone is answered, we need to be able to get some feedback from the callee to figure out if they're human. We'll use AVMD to kill the call if it detects a voicemail beep. I've looked at several different examples on the wiki and mailing list, and the only way I can figure out how to do it while keeping the requirements in mind is to at some point resort to using Loopback (something i didnt want to do). Requirements are: 1. Use a custom IVR/menu/something to get a confirmation from the callee that they are human (while also keeping it available for customization he's wanting a way to blacklist numbers on that same menu). So that rules out group_confirm_file, etc. 2. Use AVMD to kill the call if we detect the call was picked up by voicemail. 3. The custom IVR/menu/something isn't used on the deskphones 4. Deskphones need to continue to ring after the external number leg is started. I can't timeout the call on the deskphone then call the cell phone, or call the deskphone, time it out, then call the deskphone and cell phone. 5. It has to work on any type of calling method (so basically, if the deskphone rings then eventually the cell phone needs to ring to if it's assigned). Has anyone done something similar, and if so, how did you do it? Thank you, Brian Foster Project Manager/Owner's Representative Davri Investments, Incorporated P: +1-317-787-2686 M: +1-317-600-9753 Indianapolis, Indiana

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spandsp_start_dtmf detecting first digit 2 times
by Babak Yakhchali 24 Jan '15

24 Jan '15

Hi I upgraded from 1.2.12 to 1.4.15+git~20150106T122408Z~f5145540b3~64bit and now with no change to configs ,after spandsp_start_dtmf in playAndGetDigits I see the first digit is detected twice: 2015-01-24 11:51:05.083629 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN DETECTED: [3] 2015-01-24 11:51:05.083629 [DEBUG] switch_channel.c:488 RECV DTMF 3:2000 2015-01-24 11:51:05.083629 [DEBUG] switch_ivr_play_say.c:1747 done playing file /usr/local/freeswitch/sounds/fa/ir/callie/applications/2/yazd_2030_long.wav 2015-01-24 11:51:05.083629 [DEBUG] switch_ivr_menu.c:410 digits '3' 2015-01-24 11:51:05.083629 [DEBUG] switch_ivr_menu.c:561 IVR action on menu 'ivr_10' matched '3' param 'lua lua/applications/call_script.lua 48 ' 2015-01-24 11:51:05.083629 [DEBUG] switch_ivr_menu.c:565 switch_ivr_menu_execute todo=[2] EXECUTE sofia/internal/87654321@172.16.90.3:5060 lua(lua/applications/call_script.lua 48 ) 2015-01-24 11:51:05.083629 [DEBUG] freeswitch_lua.cpp:360 DBH handle 0x7f36b80d0630 Connected. EXECUTE sofia/internal/87654321@172.16.90.3:5060 lua(/usr/local/freeswitch/scripts/gui/lua/yazd/yzd_phone_activation.lua ) 2015-01-24 11:51:05.083629 [DEBUG] switch_ivr_play_say.c:70 No language specified - Using [fa] 2015-01-24 11:51:05.083629 [DEBUG] switch_ivr_play_say.c:250 Handle play-file:[applications/11/yzd_act_cid_notice.wav] (fa:fa) 2015-01-24 11:51:05.083629 [DEBUG] switch_ivr_play_say.c:1314 Codec Activated L16@8000hz 1 channels 20ms 2015-01-24 11:51:05.123638 [DEBUG] mod_spandsp_dsp.c:385 DTMF END DETECTED: [3], duration = 38 ms 2015-01-24 11:51:05.163633 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN DETECTED: [3] 2015-01-24 11:51:05.163633 [DEBUG] switch_channel.c:488 RECV DTMF 3:2000 2015-01-24 11:51:05.343664 [DEBUG] mod_spandsp_dsp.c:385 DTMF END DETECTED: [3], duration = 178 ms 2015-01-24 11:51:05.863613 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN DETECTED: [5] 2015-01-24 11:51:05.863613 [DEBUG] switch_channel.c:488 RECV DTMF 5:2000 2015-01-24 11:51:06.123594 [DEBUG] mod_spandsp_dsp.c:385 DTMF END DETECTED: [5], duration = 255 ms 2015-01-24 11:51:06.403613 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN DETECTED: [6] 2015-01-24 11:51:06.403613 [DEBUG] switch_channel.c:488 RECV DTMF 6:2000 2015-01-24 11:51:06.643596 [DEBUG] mod_spandsp_dsp.c:385 DTMF END DETECTED: [6], duration = 242 ms 2015-01-24 11:51:07.083609 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN DETECTED: [9] 2015-01-24 11:51:07.083609 [DEBUG] switch_channel.c:488 RECV DTMF 9:2000 .... the digit 3 is detected twice! Thanks

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fail_on_single_reject and sequential bridge fail-over?
by Victor Chukalovskiy 24 Jan '15

24 Jan '15

Greetings, Just discovered that fail_on_single_reject does not work with "|" operator... Not sure if it was always the case or a recent "improvement" Is there something equivalent that can be used with sequential bridge, that is with | operator? The goal is that if I bridge to "sofia/gw1/5555555555|sofia/gw2/5555555555 and I get one of the rejection codes I expect over gw1 it should not attempt gw2 Thx! -Victor

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Expires in INVITE after challenge
by Spencer Thomason 24 Jan '15

24 Jan '15

Hello all, I’ve been reviewing a bunch of pcaps today during interop with a new ULC and I noticed that our FreeSWITCH instances send an Expires header in the INVITE that wasn’t present before the challenge. I’ve got <param name="enable-timer" value="false”/> set in the profile, am I missing something? Thanks! Spencer Trace excerpt below: # U A.A.A.A:5080 -> B.B.B.B:5060 INVITE sip:1NPANXX2222@B.B.B.B SIP/2.0. Via: SIP/2.0/UDP A.A.A.A:5080;rport;branch=z9hG4bKjp8H9Q9vUvmZp. Max-Forwards: 69. From: "DISPLAY NAME" <sip:1NPANXX1111@A.A.A.A>;tag=gFX13g0aS2Z8D. To: <sip:1NPANXX2222@B.B.B.B>. Call-ID: 9b82c095-1e10-1233-a19e-001851124686. CSeq: 70696564 INVITE. Contact: <sip:gw+GW-NAME@A.A.A.A:5080;transport=udp;gw=GW-NAME>. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE. Supported: path. Privacy: none. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 286. P-Asserted-Identity: "DISPLAY NAME" <sip:1NPANXX1111@A.A.A.A>. . v=0. o=WSC_PBX 1422043530 1422043531 IN IP4 A.A.A.A. s=SIP Media Capabilities. c=IN IP4 A.A.A.A. t=0 0. m=audio 21726 RTP/AVP 0 18 101. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. # U B.B.B.B:5060 -> A.A.A.A:5080 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP A.A.A.A:5080;rport=5080;branch=z9hG4bKjp8H9Q9vUvmZp. From: "DISPLAY NAME" <sip:1NPANXX1111@A.A.A.A>;tag=gFX13g0aS2Z8D. To: <sip:1NPANXX2222@B.B.B.B>;tag=45e7e4a19f9e2adbb8c0c696f8c31422.8a84. Call-ID: 9b82c095-1e10-1233-a19e-001851124686. CSeq: 70696564 INVITE. Proxy-Authenticate: Digest realm="A.A.A.A", nonce="VMRPz1TC/k8QLkFKz6mNTenFQ4hT8yIrSb4tyoA=", qop="auth". Content-Length: 0. . # U A.A.A.A:5080 -> B.B.B.B:5060 ACK sip:1NPANXX2222@B.B.B.B SIP/2.0. Via: SIP/2.0/UDP A.A.A.A:5080;rport;branch=z9hG4bKjp8H9Q9vUvmZp. Max-Forwards: 69. From: "DISPLAY NAME" <sip:1NPANXX1111@A.A.A.A>;tag=gFX13g0aS2Z8D. To: <sip:1NPANXX2222@B.B.B.B>;tag=45e7e4a19f9e2adbb8c0c696f8c31422.8a84. Call-ID: 9b82c095-1e10-1233-a19e-001851124686. CSeq: 70696564 ACK. Content-Length: 0. . # U A.A.A.A:5080 -> B.B.B.B:5060 INVITE sip:1NPANXX2222@B.B.B.B SIP/2.0. Via: SIP/2.0/UDP A.A.A.A:5080;rport;branch=z9hG4bKKZ1aBKt0r5ajj. Max-Forwards: 69. From: "DISPLAY NAME" <sip:1NPANXX1111@A.A.A.A>;tag=gFX13g0aS2Z8D. To: <sip:1NPANXX2222@B.B.B.B>. Call-ID: 9b82c095-1e10-1233-a19e-001851124686. CSeq: 70696565 INVITE. Contact: <sip:gw+GW-NAME@A.A.A.A:5080;transport=udp;gw=GW-NAME>. Expires: 90. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE. Supported: path. Proxy-Authorization: Digest username="8729701", realm="A.A.A.A", nonce="VMRPz1TC/k8QLkFKz6mNTenFQ4hT8yIrSb4tyoA=", cnonce="m4QpfR4QEjOeoQAYURJGhg", algorithm=MD5, uri="sip:1NPANXX2222@B.B.B.B", response="744aa6cda6803838bcf8c87d1810c6ed", qop=auth, nc=00000001. Privacy: none. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 286. P-Asserted-Identity: "DISPLAY NAME" <sip:1NPANXX1111@A.A.A.A>. . v=0. o=WSC_PBX 1422043530 1422043531 IN IP4 A.A.A.A. s=SIP Media Capabilities. c=IN IP4 A.A.A.A. t=0 0. m=audio 21726 RTP/AVP 0 18 101. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20.

1 1

Play native file in proxy_media true mode
by John Nash 24 Jan '15

24 Jan '15

Is it possible to play a native sound file in proxy_media mode?

2 2

Qualify a gateway and interact with it before using it..
by Gerry Hull 23 Jan '15

23 Jan '15

I want to do this with event socket... I'm just not sure how. - outbound call starting - Validate that a gateway is up - Perform a requred out-of-band operation (an application action probably) on the gateway and wait for it to complete. - Route call to that gateway If the gateway is not up, I'd repeat for secondary and tertiary gateways. Any recipe thoughts? Thanks, -Gerry

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