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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

1 2

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

6 9

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

2 1

[ERR] switch_core_io.c:187 sofia/SIPRouter/1917xxxxxxx has no read codec
by Ali Pey 23 Jan '15

23 Jan '15

Hello, I get this error time to time to certain numbers. There is an incoming call and freeswitch is trying to do a bridge to this number. Codec on all calls is only G.711. When I see this error, I see freeswitch sends an Invite message and then Cancel in less than one second after it receives 100 Trying. Any suggestion what I should look for or why this pops up. It results to outbound call failure and hangup cause is 88: INCOMPATIBLE_DESTINATION Any suggestion or advice would be greatly appreciated. Thanks, Ali Pey

2 3

In call Applications - drop_dtmf
by Magnus Kelly 23 Jan '15

23 Jan '15

Hello list, I wonder if someone could give me a hint on how I go about working out when applications can be used after bridging A & B legs? I have spent a fair amount of time googling the list, but not yet stumbled into how it is determined if an application can be invoked mid call. Some can and some can¹t but I have yet to work out the logic that decides this? The application in question is ³drop_dtmf² but the question is more general. Also how does one work out if it should work with SIP-Info instead of 2833? Thanks Magnus

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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 103, Issue 247
by Ahmed Habiba 23 Jan '15

23 Jan '15

Thank you really Michael, I go through logs and I find the below, the question why could lead to that, the machine is 16 core?? 2015-01-23 02:06:59.593497 [CRIT] switch_core_session.c:1780 Thread Failure! 2015-01-23 02:06:59.593497 [CRIT] switch_core_session.c:1713 LUKE: I'm hit, but not bad. 2015-01-23 02:06:59.593497 [CRIT] switch_core_session.c:1714 LUKE'S VOICE: Artoo, see what you can do with it. Hang on back there.... Green laserfire moves past the beeping little robot as his head turns. After a few beeps and a twist of his mechanical arm, Artoo reduces the max sessions to 346 thus, saving the switch from certain doom. From: Michael Jerris <mike(a)jerris.com <mailto:mike@jerris.com>> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>> Date: January 23, 2015 at 2:51:18 AM GMT+3 Reply-To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>> Subject: Re: [Freeswitch-users] Session Limit look at the logs, if you extend beyond your resources and freeswitch can't start threads among other things, it will lower your session limit. > On Jan 22, 2015, at 6:47 PM, Ahmed Habiba <ahabiba(a)gmail.com <mailto:ahabiba@gmail.com>> wrote: > > Dears, > > kindly Im facing some issue with concurrent session limit, below is the status showing the maximum sessions of 355, however I set maximum sessions to 10000, you kind help to identify why freeswitch use lower limit than the configured one will be appreciated. > > /etc/freeswitch# fs_cli -x "status" > UP 0 years, 0 days, 0 hours, 19 minutes, 2 seconds, 490 milliseconds, 264 microseconds > FreeSWITCH (Version 1.5.8b+git git 35f2bcc 2014-02-12 23:33:16Z 64bit) is ready > 345 session(s) since startup > 345 session(s) - peak 345, last 5min 345 > 0 session(s) per Sec out of max 1000, peak 66, last 5min 0 > 335 session(s) max > min idle cpu 0.00/62.00 > Current Stack Size/Max 240K/8192K > > switch.conf.xml: > > <param name="max-sessions" value="10000"/> >  > <param name="sessions-per-second" value="1000”/> From: Muhammad Naseer Bhatti <nbhatti(a)gmail.com <mailto:nbhatti@gmail.com>> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>>, Michael Jerris <mike(a)jerris.com <mailto:mike@jerris.com>> Date: January 23, 2015 at 2:58:31 AM GMT+3 Reply-To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>> Subject: Re: [Freeswitch-users] Unable to access channel vars after session terminates evens with set_zombie_exec() Setting inline will have performance penalty? — Thanks, Muhammad Naseer Bhatti From: Michael Jerris <mike(a)jerris.com> <mailto:mike@jerris.com> Reply: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org>> <mailto:freeswitch-users@lists.freeswitch.org> Date: January 23, 2015 at 2:11:48 AM To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org>> <mailto:freeswitch-users@lists.freeswitch.org> Subject: Re: [Freeswitch-users] Unable to access channel vars after session terminates evens with set_zombie_exec() > if you set them using inline they will be set during routing state, before it goes to execute. Once it is in execute state, it checks status between every action. > >> On Jan 22, 2015, at 3:53 PM, Muhammad Naseer Bhatti <nbhatti(a)gmail.com <mailto:nbhatti@gmail.com>> wrote: >> >> >> I need to access some channel variables being set in the dialplan, but the channel is already hanged up too quick, ORIGINATOR_CANCEL. Since the session was still setting up the channel variables. I have tried app, set_zombie_exec but still not able to see any channel vars set in the dial plan. Am i doing something wrong or if there is a better way to see the channel vars if the session terminates too soon? >> >> With: git a067a49 >> >> Dialplan: sofia/internal/9401404(a)10.211.55.26 <mailto:sofia/internal/9401404@10.211.55.26> Regex (PASS) [internal] destination_number(1786866) =~ /$/ break=on-false >> Dialplan: sofia/internal/9401404(a)10.211.55.26 <mailto:sofia/internal/9401404@10.211.55.26> Action set_zombie_exec() >> Dialplan: sofia/internal/9401404(a)10.211.55.26 <mailto:sofia/internal/9401404@10.211.55.26> Action set(hangup_after_bridge=true) >> Dialplan: sofia/internal/9401404(a)10.211.55.26 <mailto:sofia/internal/9401404@10.211.55.26> Action set(continue_on_fail=true) >> Dialplan: sofia/internal/9401404(a)10.211.55.26 <mailto:sofia/internal/9401404@10.211.55.26> Action set(fail_on_single_reject=USER_BUSY,NO_ANSWER,NO_USER_RESPONSE,ORIGINATOR_CANCEL) >> Dialplan: sofia/internal/9401404(a)10.211.55.26 <mailto:sofia/internal/9401404@10.211.55.26> Action set(disable_hold=true) >> Dialplan: sofia/internal/9401404(a)10.211.55.26 <mailto:sofia/internal/9401404@10.211.55.26> Action set(failed_json_cdr_prefix=failed_cdr_index) >> Dialplan: sofia/internal/9401404(a)10.211.55.26 <mailto:sofia/internal/9401404@10.211.55.26> Action set(debug_cdr=0) >> Dialplan: sofia/internal/9401404(a)10.211.55.26 <mailto:sofia/internal/9401404@10.211.55.26> Action set(debug_cdr_sql=1) >> Dialplan: sofia/internal/9401404(a)10.211.55.26 <mailto:sofia/internal/9401404@10.211.55.26> Action set(cust_default_lrn=intra) >> Dialplan: sofia/internal/9401404(a)10.211.55.26 <mailto:sofia/internal/9401404@10.211.55.26> Action set(cust_lrn_dip_cost=0.01817) >> Dialplan: sofia/internal/9401404(a)10.211.55.26 <mailto:sofia/internal/9401404@10.211.55.26> Action set(cust_jurisdiction=INTRASTATE) >> Dialplan: sofia/internal/9401404(a)10.211.55.26 <mailto:sofia/internal/9401404@10.211.55.26> Action limit(hash random_xgw 0.0.0.0 10/1 !FACILITY_REJECTED) >> Dialplan: sofia/internal/9401404(a)10.211.55.26 <mailto:sofia/internal/9401404@10.211.55.26> Action limit_execute(hash outbound random_xgw 20 bridge [enable_heartbeat_events=5,nibble_rate=0.2192,nibble_increment=5,nibble_account=AB8KA191,carrier_switch=random_xgw,carrier_switch_id=1,carrier_ratecard_id=2,carrier_rate_rev=1,carrier_rate_type=lrn,carrier_id=1,carrier_connection_cost=0,carrier_rate=0.0119,carrier_interstate_cost=0.0119,carrier_intrastate_cost=0.0119,carrier_enable_billing=t,carrier_call_increment=1,carrier_min_duration=5,carrier_balance=17099.88924]sofia/gateway/random_xgw/1786866) >> Dialplan: sofia/internal/9401404(a)10.211.55.26 <mailto:sofia/internal/9401404@10.211.55.26> Action limit(hash switch02 0.0.0.0 10/1 !FACILITY_REJECTED) >> Dialplan: sofia/internal/9401404(a)10.211.55.26 <mailto:sofia/internal/9401404@10.211.55.26> Action limit_execute(hash outbound switch02 20 bridge [enable_heartbeat_events=5,nibble_rate=0.2192,nibble_increment=5,nibble_account=AB8KA191,carrier_switch=switch02,carrier_switch_id=3,carrier_ratecard_id=2,carrier_rate_rev=1,carrier_rate_type=lrn,carrier_id=1,carrier_connection_cost=0,carrier_rate=0.0119,carrier_interstate_cost=0.0119,carrier_intrastate_cost=0.0119,carrier_enable_billing=t,carrier_call_increment=1,carrier_min_duration=5,carrier_balance=17099.88924]sofia/gateway/switch02/1786866) >> Dialplan: sofia/internal/9401404(a)10.211.55.26 <mailto:sofia/internal/9401404@10.211.55.26> Action hangup() >> 2015-01-22 11:47:14.856013 [DEBUG] switch_core_state_machine.c:216 (sofia/internal/9401404(a)10.211.55.26 <mailto:sofia/internal/9401404@10.211.55.26>) State Change CS_ROUTING -> CS_EXECUTE >> 2015-01-22 11:47:14.856013 [DEBUG] switch_core_session.c:1388 Send signal sofia/internal/9401404(a)10.211.55.26 <mailto:sofia/internal/9401404@10.211.55.26> [BREAK] >> 2015-01-22 11:47:14.856013 [DEBUG] switch_core_state_machine.c:528 (sofia/internal/9401404(a)10.211.55.26 <mailto:sofia/internal/9401404@10.211.55.26>) State ROUTING going to sleep >> 2015-01-22 11:47:14.856013 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/9401404(a)10.211.55.26 <mailto:sofia/internal/9401404@10.211.55.26>) Running State Change CS_EXECUTE >> 2015-01-22 11:47:14.856013 [DEBUG] sofia.c:6614 Channel sofia/internal/9401404(a)10.211.55.26 <mailto:sofia/internal/9401404@10.211.55.26> entering state [terminated][487] >> 2015-01-22 11:47:14.856013 [NOTICE] sofia.c:7530 Hangup sofia/internal/9401404(a)10.211.55.26 <mailto:sofia/internal/9401404@10.211.55.26> [CS_EXECUTE] [ORIGINATOR_CANCEL] >> 2015-01-22 11:47:14.856013 [DEBUG] switch_channel.c:3222 Send signal sofia/internal/9401404(a)10.211.55.26 <mailto:sofia/internal/9401404@10.211.55.26> [KILL] >> 2015-01-22 11:47:14.856013 [DEBUG] switch_core_session.c:1388 Send signal sofia/internal/9401404(a)10.211.55.26 <mailto:sofia/internal/9401404@10.211.55.26> [BREAK] >> 2015-01-22 11:47:14.856013 [DEBUG] switch_core_state_machine.c:535 (sofia/internal/9401404(a)10.211.55.26 <mailto:sofia/internal/9401404@10.211.55.26>) State EXECUTE >> 2015-01-22 11:47:14.856013 [DEBUG] mod_sofia.c:178 sofia/internal/9401404(a)10.211.55.26 <mailto:sofia/internal/9401404@10.211.55.26> SOFIA EXECUTE >> 2015-01-22 11:47:14.856013 [DEBUG] switch_core_state_machine.c:535 (sofia/internal/9401404(a)10.211.55.26 <mailto:sofia/internal/9401404@10.211.55.26>) State EXECUTE going to sleep >> > ______________________________________

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FreeSWITCH Friday FreeForAll Reminder!
by Ken Rice 23 Jan '15

23 Jan '15

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888@conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon

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