March 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Flooded with Stun Errors
by Naveen Khanna 06 Mar '15

06 Mar '15

Hi, I am getting flood of following messages. 2015-03-05 13:49:25.766474 [ALERT] switch_rtp.c:875 STUN PACKET TYPE: BINDING_RESPONSE 2015-03-05 13:49:25.766474 [ALERT] switch_rtp.c:878 |---: STUN ATTR INVALID 2015-03-05 13:49:25.926473 [ALERT] switch_rtp.c:1678 sofia/internal/sip:2002@df7jal23ls0d.invalid audio stat 98.00 2012/2042 flaws: 30 mos: 4.48 v: 23.30 9.52/400.00 2015-03-05 13:49:25.946476 [ALERT] switch_rtp.c:5440 sofia/internal/sip:2002@df7jal23ls0d.invalid syncing 1 audio packet(s) 2015-03-05 13:49:25.986475 [ALERT] switch_rtp.c:1678 sofia/internal/sip:2003@df7jal23ls0d.invalid audio stat 99.00 241/242 flaws: 1 mos: 4.49 v: 55.49 10.00/400.00 2015-03-05 13:49:26.146475 [ALERT] switch_rtp.c:1678 sofia/internal/sip:2002@df7jal23ls0d.invalid audio stat 98.00 2012/2043 flaws: 31 mos: 4.48 v: 23.29 9.52/400.00 2015-03-05 13:49:26.166475 [ALERT] switch_rtp.c:5440 sofia/internal/sip:2002@df7jal23ls0d.invalid syncing 1 audio packet(s) 2015-03-05 13:49:26.286473 [ALERT] switch_rtp.c:875 STUN PACKET TYPE: BINDING_RESPONSE 2015-03-05 13:49:26.286473 [ALERT] switch_rtp.c:878 |---: STUN ATTR INVALID 2015-03-05 13:49:26.366474 [ALERT] switch_rtp.c:1678 sofia/internal/sip:2002@df7jal23ls0d.invalid audio stat 98.00 2012/2044 flaws: 32 mos: 4.48 v: 23.28 9.52/400.00 2015-03-05 13:49:26.426473 [ALERT] switch_rtp.c:875 STUN PACKET TYPE: BINDING_RESPONSE 2015-03-05 13:49:26.426473 [ALERT] switch_rtp.c:878 |---: STUN ATTR INVALID 2015-03-05 13:49:26.486474 [ALERT] switch_rtp.c:5440 sofia/internal/sip:2002@df7jal23ls0d.invalid syncing 1 audio packet(s) 2015-03-05 13:49:26.666475 [ALERT] switch_rtp.c:875 STUN PACKET TYPE: BINDING_RESPONSE 2015-03-05 13:49:26.666475 [ALERT] switch_rtp.c:878 |---: STUN ATTR INVALID Requesting an insight what could have been causing this. Regards, Naveen Khanna

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FreeSWITCH Friday FreeForAll Reminder!
by Ken Rice 06 Mar '15

06 Mar '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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Does Sangoma Analog card support callerid in dtmf format?
by james.zhu 06 Mar '15

06 Mar '15

Hello: My customer installed Sangoma A200 with FreeTDM. A200 can not get the callerid in dtmf format, But if the callerid send by FSK format, the callerid can be received. I also take a loot at the code for callerid process in freetdm, it seems lack of supporting the callerid in dtmf format. Mr.Moises also mentioned that three year ago: http://freeswitch-users.2379917.n2.nabble.com/Caller-id-on-incoming-FXO-wit… I am not very sure the problem is still there. I post this issue to with logs, please help me check that: https://freeswitch.org/jira/browse/OPENZAP-235?jql=project%20%3D%20OPENZAP%… I think callerid in dtmf/fsk format should be a standard feature, hope it can be resolved soon.

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freeswitch support
by Antony Aguirre Morales 06 Mar '15

06 Mar '15

I have configured my fs with xml_curl module in the directory part, may serve 100,000 extensions from a single FS or is there any limitation as to the software? IN the hardware I have a server with the following specifications: Brand: DELL RAM: 4GB CPU: Intel (R) Xeon (R) CPU E5405 @ 2.00GHz cores: 4 DD: 50GB regards.

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Stale Channels build up
by Naveen Khanna 06 Mar '15

06 Mar '15

Hi, I am getting Stale Channels build up in my Freeswitch installation. We have a deployed a Dialer solution which is based on mod_callcenter, and a continuous stream of call are generated using Originate api command. As time progress the hung channels build increase and moment it reaches a threshold of 420+ of stale channels the entire site goes down. The Sip connections / extensions get disconnect and there is no means that they can get connected again. However, switch continues receive incoming calls, but extensions / agents sitting behind mod_callcenter are unable to answer calls. The only solution that I have is to restart the switch. I am using : Freeswitch 1.4 enabled with WebRTC support, SipmML5 based Sip Client and postgres as database. The solution is deployed on CentOS 6.6 64 bit. Regards, Naveen Khanna

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Conference Announce Count Inline - Not working
by T Fred Farmington 05 Mar '15

05 Mar '15

I am following the instructions on: "This example is a very quick and dirty dialplan and conference config that lets you hear how many callers are in conference." https://wiki.freeswitch.org/wiki/Conference_Announce_Count_Inline Within my existing conf\autoload_configs\conference.conf.xml I added the new caller controls: <group name="plain"> as shown in the referenced page. Additionally within the same file I edited the <profile name="default"> to use the new: <param name="caller-controls" value="plain"/> I then created a new XML file: conf/dialplan/default/01_Announce_Conf_Count.xml containing the code shown in the referenced page. Lastly I edited this file to change the last 'application' line beginning with: application="say" to the following <action application="phrase" data="callers_in_conf,${conf_count}"/> so as to play the existing macro phrase which exists in: conf\lang\en\ivr\sounds.xml It should be playing existing wav files and, therefore not need, TTS But when I enter the conference.. 1. The first caller in gets a 'voice' message played indicating: "You are currently the only person in this conference" 2. But subsequent callers get nothing but a tone upon entry into the conference. The intended macro phrase is not playing. What needs to change to make the 'Announce Conference Participant Count" work. Any assistance would be greatly appreciated. Thanks

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Re: [Freeswitch-users] Real-time billing application for the, FreeSWITCH (mod_lua, mod_perl or ESL)
by DanB 05 Mar '15

05 Mar '15

One more thing to consider when you build your real-time billing application is call authorize before connect. If you only treat answer and hangup you can end up with the call without balance going through and eating some seconds at each connect (which in case of DoS requests can result in quite serious amounts for you). In CGRateS we use park application in dialplan to put the call on hold before being authorized by the same CHANNEL_PARK event and loop the call through dialplan back when we are done checking it. Just my two cents, DanB On 05.03.2015 18:36, freeswitch-users-request(a)lists.freeswitch.org wrote: > 015-03-04 17:17 GMT-05:00 jorgemariodlc<jorgemariodlc(a)gmail.com>: > >> >I actually working in it, I found it yesterday you need to add those lines >> >(/autoload_configs/lua.conf.xml): >> > >> ><configuration name="lua.conf" description="LUA Configuration"> >> > <settings> >> > >> > <hook event="CHANNEL_ANSWER" script="answer.lua"/> >> > <hook event="CHANNEL_HANGUP_COMPLETE" script="hangup.lua"/> >> > >> > </settings> >> ></configuration> >> > >> >Check this link to know what information is in each event, because it's >> >important to handle the direction-call (Outbound, Inbound) >> >https://wiki.freeswitch.org/wiki/Event_List

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How can i create dynamic dialplan in freeswitch.
by Aqs Younas 05 Mar '15

05 Mar '15

Hi, user. After working for more than 3 months while writing my dialplan in static xml file,but now wants to know how can i effectively create dynamic dialplan in freeswitch. Thanks.

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What does cause 'INCOMPATIBLE_DESTINATION' a hangup cause?
by Aqs Younas 05 Mar '15

05 Mar '15

Hi, list. I see my freeswitch hanging a lot of calls with INCOMPATIBLE_DESTINATION as hangup cause in my cdr though the DID they are hitting is a proper number. Could someone please tells me why freeswitch is hanging calls with this reason. Thanks,

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(no subject)
by Sergey Safarov 05 Mar '15

05 Mar '15

I want to receive events from the erlang module and for that execute the following commands [root@fs1 xml]# erl -sname test -setcookie ClueCon Erlang/OTP 17 [erts-6.2.1] [source] [64-bit] [smp:2:2] [async-threads:10] [hipe] [kernel-poll:false] Eshell V6.2.1 (abort with ^G) (test@fs1)1> {foo, fs1@fs1} ! {event, 'CHANNEL_CREATE'}, receive Y -> Y after 1000 -> timeout end. ok (test@fs1)2> And second way [root@fs1 xml]# erl -sname test -setcookie ClueCon Erlang/OTP 17 [erts-6.2.1] [source] [64-bit] [smp:2:2] [async-threads:10] [hipe] [kernel-poll:false] Eshell V6.2.1 (abort with ^G) (test@fs1)1> {foo, fs1@fs1} ! {event, 'ALL'}. {event,'ALL'} (test@fs1)2> receive Y -> Y after 1000 -> timeout end. ok (test@fs1)3> And I can not get events. What am I doing wrong?

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