April 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Manipulating audio streams at sample level
by Oivvio Polite 05 Apr '17

05 Apr '17

I have a scenario where I'd like to be able to manipulate audio streams coming into FS on a sample level. This is a 3-party SIP call where 2 parties (A and B) are in the same room but using individual handsets. This creates an echo effect for the third party (C) since the audio streams from A and B will arrive at her location slightly out of sync. So at the start of a call I want to examine the audio streams coming from A B as they come into FS to see how out of sync they are, then add an appropriate delay to the one that's ahead and put that delayed audio back into FS. Oivvio

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BEHAVIOR CHANGE (YES PAY ATTENTION TO THIS IF YOU USE FAXING)
by Brian West 05 Apr '17

05 Apr '17

FreeSWITCHers, In 1.6.14 we did a slight behavior change related to T.38 re-invite acceptance. Before 1.6.14 we would by default accept the re-invite even if t38 wasn't enabled and switch to UDPTL mode. This was an incorrect behavior to do by default. But one caveat in 1.6.14, We by default 488 the fax but switch the media mode to UDPTL which is also a bad behavior fixed in (FS-9943) slated for 1.6.15. If you wish to retain the previous behavior you can set fax_enable_t38=true in your vars.xml, You can also not set that and just set refuse_t38=true if you wish to never do t.38. These default behaviors are fixed in our next release 1.6.15. Thanks, -- *Brian West* brian(a)freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com https://www.gofundme.com/freeswitch_ubuntu Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest

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Grandstream GXP1625 not working with Freeswitch Version 1.4.26~64bit
by Uday kumar 05 Apr '17

05 Apr '17

Hi, I have multiple Grandstream IP phone in my network. All IP phones are working but only one IP phone is not workint i.e GXP1625 with Freeswitch. I am able to register extension and also incoming call is working but when making outgoing call then only its not working. I tried same IP phone with Asterisk and its working in incoming and outgoing both. I am using this only for extension to extension calls so incoming and outgoing calls only between extensions. Is Grandstream GXP1625 supported with Freeswitch or not? *When I am dialing outgoing call then on CLI getting below logs:-* [NOTICE] switch_channel.c:1077 New Channel sofia/internal/2005101@Server_IP [ab84501a-345e-445c-8d4a-52d4d6594e21] [DEBUG] switch_core_session.c:1062 Send signal sofia/internal/2005101@Server_IP [BREAK] [DEBUG] switch_core_session.c:1062 Send signal sofia/internal/2005101@Server_IP [BREAK] [DEBUG] switch_core_state_machine.c:472 (sofia/internal/2005101@Server_IP) Running State Change CS_NEW [DEBUG] sofia.c:8957 sofia/internal/2005101@Server_IP receiving invite from XX.XX.XX.XX:65476 version: 1.4.26 64bit [DEBUG] sofia.c:9124 IP XX.XX.XX.XX Rejected by acl "domains". Falling back to Digest auth. [DEBUG] switch_core_state_machine.c:491 (sofia/internal/2005101@Server_IP) State NEW [DEBUG] switch_core_session.c:1062 Send signal sofia/internal/2005101@Server_IP [BREAK] [DEBUG] sofia.c:2065 detaching session ab84501a-345e-445c-8d4a-52d4d6594e21 *Output of sofia status profile internal reg 2005101:-* Registrations: ================================================================================================= Call-ID: 926642231-5060-1(a)GG.CCG.HD.CAA User: 2005101@Server_IP Contact: "" <sip:2005101@XX.XX.XX.XX:65476> Agent: Grandstream GXP1625 1.0.4.6 Status: Registered(UDP)(unknown) EXP(2017-03-16 12:25:52) EXPSECS(2409) Ping-Status: Reachable Ping-Time: 0.00 Host: Server_IP IP: XX.XX.XX.XX Port: 65476 Auth-User: 2005101 Auth-Realm: Server_IP MWI-Account: 2005101@Server_IP -- Thanks & Regard Uday Site:- www.shareyourknowledge.in Mobile:- +91-9377579349

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Clearmode
by Stefan Kainz 04 Apr '17

04 Apr '17

Hello, im having troubles with clearmode / proxy_media. Im getting an inbound call, with just CLEARMODE/8000 in the SDP Header: *m=audio 12600 RTP/AVP 56** **a=rtpmap:56 CLEARMODE/8000** **a=sendrecv** **a=ptime:20* proxy_media is set to true, but i still get: *2017-03-29 10:57:32.547851 [ERR] switch_core_codec.c:666 Invalid codec CLEARMODE!** **2017-03-29 10:57:32.547851 [ERR] switch_core_media.c:2414 Can't load codec?* Also inbound-late-negotiation is set to true on external profile. I've found a mod_clearmode in jira, do i need this to get this to work? Or is there anything else i need to do? thank you in andvance! regards,

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100% CPU usage
by Grant Bagdasarian 04 Apr '17

04 Apr '17

Hi, We're currently experiencing a weird issue with one of our freeswitch servers where the CPU just goes berserk and stays at 100%. There are no active calls at that moment, but FS is still using 100% CPU. I was unable to get a debug using gdb, since freeswitch does not crash. We had this issue on 1.6.13 and then I upgraded to 1.6.15 this morning, but the problem still persists. There is currently only one active application running on this box which is using LUA. Maybe that is causing the issue? We checked the code but there are no weird loops etc. Perhaps we are missing something? Any ideas how to trace this? Regards, Grant

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Grandstream DP750 or Yealink W56P
by Rajil Saraswat 04 Apr '17

04 Apr '17

Hello, I am looking to buy a SIP/DECT phone to replace my Linksys SP3102. It seems that new unit from Cisco to replace this is SPA232D which is also at its end of life. My quick market survey shows two products Yealink W56P and Grandstream DP750 as viable options. Unfortunately, both of these don't have an FXO interface. Yealink includes an Openvpn client while Grandstream has Opus support. The latter is cheaper as well. Is there any recommendation on which is a better product? My plan is to use the FXO interface of the SPA3102 along with one of these SIP/DECT systems from above. Thanks PS: I looked for Gigaset devices but Amazon USA did not seem to carry them

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Call Center Behaviour
by Jai Rangi 03 Apr '17

03 Apr '17

We are using call center module. We noticed one thing, if the agent change status to not available or on break and then after a while come back to change the status to available. His stats (num of calls answered or talk_time) reset to 0. Example, callcenter_config queue list agents sales@domain name|system|uuid|type|contact|status|state|max_no_answer|wrap_up_time|reject_delay_time|busy_delay_time|no_answer_delay_time|last_bridge_start|last_bridge_end|last_offered_call|last_status_change|no_answer_count|calls_answered|talk_time|ready_time 101@domain|single_box||callback|[call_timeout=30]user/130@domain |Available|Waiting|0|0|0|0|0|0|0|1489009000|1488960061|1|0|0|0 102@domain|single_box||callback|[call_timeout=30]user/111@jdomain |Available|Waiting|0|0|60|60|0|1489009008|1489009016|1489009000|1488960061 *|0|1|8|0* Here I see 102, had one call for 8 seconds, Change the agent status to break. session:execute("set", "res=${callcenter_config(agent set status ${caller_id_number}@${domain_name} 'On Break')}") 101@domain|single_box||callback|[call_timeout=30]user/101@domain |Available|Waiting|0|0|0|0|0|0|0|1489009000|1488960061|1|0|0|0 102@domain|single_box||callback|[call_timeout=30]user/101@domain|On Break|Waiting|0|0|60|60|0|1489009008|1489009016|1489009000|1489009163| *0|1|8|0* Change the agent state to Available On Demand, stats are still there. session:execute("set", "res=${callcenter_config(agent set status ${caller_id_number}@${domain_name} 'Available (On Demand)')}") 101@domain|single_box||callback|[call_timeout=30]user/101@domain |Available|Waiting|0|0|0|0|0|0|0|1489009000|1488960061|1|0|0|0 102@domain|single_box||callback|[call_timeout=30]user/102@domain|Available (On Demand)|Waiting|0|0|60|60|0|1489009008|1489009016|1489009000|1489009252| *0|1|8|0* Now changing to Available and stats are reset ed to 0,0, session:execute("set", "res=${callcenter_config(agent set status ${caller_id_number}@${domain_name} 'Available')}") 101@domain|single_box||callback|[call_timeout=30]user/101@domain |Available|Waiting|0|0|0|0|0|0|0|1489009000|1488960061|1|0|0|0 102@domain|single_box||callback|[call_timeout=30]user/102@domain |Available|Waiting|0|0|60|60|0|1489009008|1489009016|1489009000|1489009415|0|0|0|0 Is there way to control this behavior. So basically we can to control the call stats.

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DTMF detection error
by Hamit Türkü Kaya 03 Apr '17

03 Apr '17

Hi I’m using FreeSWITCH Version 1.7.0+git~20151120T222441Z~3ac546fbf1~64bit (git 3ac546f 2015-11-20 22:24:41Z 64bit) on our media server with mod_pyhton. When i try to detect dtmf answers with playAndGetDigits, it behaves weird. For instance i’m accepting 3 - 4 digits, when i press 42 it plays error sound as usual, but after then when i press 4256, it continues to play error sound. When i checked detected tones i get strange results. Sometimes it detects 4256 as 426 or 456 with skipping digits. Sometimes it merges with failed digits like 12425 what can be possibly cause this. I couldn’t find any solution. My related python script codes are below. gather_result = session.playAndGetDigits( int(min_digits), int(max_digits), int(max_attempts) gather_timeout, "#", ask_sound, error_sound, "\\d+" ) if gather_result: session.setVariable(variable_name), gather_result) session_vars_list.append(variable_name + "=" + session.getVariable("appvar_"+str(arguments["variable_name"]))) Regards. — Hamit Türkü KAYA, Geliştirici BULUTFON AŞ | www.bulutfon.com | 0 (850) 885 0000

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park_after_bridge don't working for busy destination!
by Alexander Haugg 03 Apr '17

03 Apr '17

Hi, Ich try to catch the leg A after the destination is hangup with the cause busy to signalling a busy tone stream to the caller (leg A) But leg A will dropped too. Here is the test dialplan: <context name="H3KSip_SP_outbound"> <extension name="outbound_softphone"> <condition field="${channel_name}" expression="^sofia/H3KSip_B2Bua/.*$" break="on-false"> <action application="set" data="ringback=%(1500,3000,425)" /> <action application="set" data="instant_ringback=true" /> <action application="multiset" data="ignore_early_media=true hangup_after_bridge=false park_after_bridge=true" /> <action application="export" data="nolocal:absolute_codec_string=G722,PCMU,PCMA" /> <action application="bridge" data="{ignore_early_media=true}sofia/gateway/GW_H3KSip/${destination_number}" /> <action application="log" data="ERR last_bridge_hangup_cause=${last_bridge_hangup_cause}" /> <action application="log" data="ERR bridge_hangup_cause=${bridge_hangup_cause}" /> <action application="sleep" data="1000" /> <action application="playback" data="tone_stream://%(480,480,425);loops=3" /> <action application="hangup" data="USER_BUSY"/> </condition> </extension> </context> Whats wrong? Thanks a lot!

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VPX on windows?
by Alexander Haugg 01 Apr '17

01 Apr '17

Hi, in the newest main brunch I can see the vpxlib project, but after the build I did not found the libvpx.dll or a mod_vpx.dll. Is the VPX video support on windows available? Thanks a lot!

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