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Event Multicast vs. ODBC Core
by Jody Rudolph 14 Sep '10

14 Sep '10

For the purpose of clustering, is using using the event-socket multicast between servers capable of providing all of the same usable information that sharing a core ODBC DSN? Thanks, Jody Rudolph

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[CRIT] sofia_glue.c:743 No RTP ports available!
by Jan Riedinger 14 Sep '10

14 Sep '10

Hello all together! I started to operate FreeSwitch with live traffic for the first time yesterday. There is nothing special with my setup beside I used mod_h323. After some hours of operation the termination for all calls failed. In my CDR I find: hangup_cause_q850="27", hangup_cause="DESTINATION_OUT_OF_ORDER", sip_hangup_disposition="send_refuse". In freeswitch.log I find a lot of entries like: 2010-09-13 23:45:59.916855 [DEBUG] switch_core_state_machine.c:338 (sofia/external/9999#24107363687(a)xx.xx.xx.xx) State INIT 2010-09-13 23:45:59.916855 [DEBUG] mod_sofia.c:83 sofia/external/9999#24107363687(a)xx.xx.xx SOFIA INIT 2010-09-13 23:45:59.916855 [CRIT] sofia_glue.c:743 No RTP ports available! 2010-09-13 23:45:59.916855 [ERR] sofia_glue.c:1646 Port Error! 2010-09-13 23:45:59.916855 [DEBUG] switch_channel.c:2322 (sofia/external/9999#24107363687(a)xx.xx.xx) Callstate Change DOWN -> HANGUP and 2010-09-13 23:48:05.038734 [ERR] mod_h323.cpp:2013 h323/12424669543 Unsupported ptime of 2 on write Audio codec G.729{sw} for connection [0x2aaac01d2500] 2010-09-13 23:48:05.038734 [DEBUG] mod_h323.cpp:2017 h323/12424669543 initialise write codec Audio for connection [0x2aaab00e9d18] 2010-09-13 23:48:05.038734 [DEBUG] mod_h323.cpp:2072 Set write Audio codec to G.729{sw} for connection [0x2aaac01d2500] 2010-09-13 23:48:05.038734 [DEBUG] mod_h323.cpp:2078 ------------------->tech_pvt->rtp_session = [(nil)] 2010-09-13 23:48:05.038734 [DEBUG] mod_h323.cpp:2079 ------------------->samples_per_packet = 160 2010-09-13 23:48:05.038734 [DEBUG] mod_h323.cpp:2080 ------------------->actual_samples_per_second = 8000 2010-09-13 23:48:05.038734 [DEBUG] mod_h323.cpp:2127 ------------------------->tech_pvt->rtp_session = (nil) 2010-09-13 23:48:05.038734 [ERR] mod_h323.cpp:2132 AUDIO RTP REPORTS ERROR: [Missing local port] 2010-09-13 23:48:05.038734 [DEBUG] switch_channel.c:2322 (h323/12424669543) Callstate Change DOWN -> HANGUP 2010-09-13 23:48:05.038734 [NOTICE] mod_h323.cpp:2133 Hangup h323/12424669543 [CS_INIT] [DESTINATION_OUT_OF_ORDER] After a restart of FreeSwitch, it is working again. Does anyone have an idea, what could has caused the problem? Thank you in advance Jan -- Jan Riedinger Phone : +49-30-39 73 19 66 Dipl.-Inf. | Managing Director Fax : +49-30-39 73 19 64 E-Mail: riedinger(a)sns.eu SNS Consult GmbH ICQ : 163-237-041 Südwestkorso 49a MSN : jan(a)sns-consult.de 14197 Berlin GERMANY Skype : Jan Riedinger AG Charlottenburg - HRB 71973

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How do I play media to a bridged call
by Antony Martin 14 Sep '10

14 Sep '10

Hi I am working on mod_woomera and have a dial plan question. I am connecting a inbound caller into a video conference hosted on the Woomera Server using the dial plan below. <extension name="woomera-controlled-conf-controller"> <condition field="destination_number" expression="^611$"> <action application="set" data="hangup_after_bridge=true"/> <action application="answer"/> <action application="bridge" data="woomera/conf:1/role=controller:tiles=9"/> </condition> </extension> When connected I would like to play a video clip to the video conference using the WoomeraPlayback application that is supported in mod_woomera. <action application="WoomeraPlayback" data="conference_demo/diastar-video-conf"/> How do I execute this on a bridged call? Many Thanks Antony

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issue in playing background music while carrying out external commands
by Chaitanya Bhatt // Viva 14 Sep '10

14 Sep '10

Hii We are trying to play some hold music to that caller in freeswitch while carrying out some external task like lengthy database query. We are using ESL to handle incoming calls(using outbound socket to a perl script). We had tried bgapi with start_local_stream but call hangs up immediately. $conn->execute("event", "plain all"); $conn->bgapi("start_local_stream","default"); while(1) { my $e = $conn->recvEventTimed(100); $e = $conn->recvEvent(); my $event_name; if ($e) { my $h = $e->firstHeader(); while ($h) { if($h eq "Event-Name") { $event_name = $e->getHeader($h); } #log header $h = $e->nextHeader(); } } } We had tried valet_park but getting required result. Can someone please guide us how to hold call with music or can provide some code snippet ? Incase of any further queries, Please feel free to mail me or contact me on the numbers provided below. Thanks & Regards, Chaitanya Bhatt Software Engineer. Viva Infomedia Pvt. Ltd. 242, Oshiwara Industrial Centre, New Link Road, Opp. Oshiwara Bus Depot, Goregaon West, Mumbai 400104. Direct: +91.22.40310356 Board: +91.22.40310310 Email : chaitanya(a)vivainfomedia.com Viva Infomedia: Awarded as Best SME (E-Commerce) at CNBC Emerging India Awards 2009

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mod_skypopen - performence and capacity.
by Shamun toha md 14 Sep '10

14 Sep '10

Hi, Lets say 1 wants, Enterprise production plan with mod_skypopen, not a target of 20 channels only. lets assume more channels (T1) sort of target: # Question operating system should be CentOS or FreeBSD or OpenSolaris?, to win a little more CPU space for this Skype processor monster. Then what server hardware platform should it be, to provide support physically too: 1. What is the CPU should be target to handle more skype channels? ex: 8 CPU in 1 hardware ( 20 skype instance x 8 = 160 channels? ), which server or os support this even? 2. What is the RAM should be? ex: 20GB? which hardware or os? 3. What server it should be? Cloud or Private co-located? 4. What is the bandwidth plan? ex: 10GB or 2MB? 5. L16 codec, will it be using inbound also for outbound? What will be the Kilo bytes per seconds? There is no reference what CPU with What unit of Ghz used? ex: http://wiki.freeswitch.org/wiki/Skypiax Thanks for your feedback

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CDR Behaviour (b_only)
by Ross McKillop 14 Sep '10

14 Sep '10

Hi, I'm curious how others handle CDRs from FreeSWITCH. I need them to end up in a MySQL database and the easiest way I've found of achieving this is to use the mod_xml_cdr module to call a PHP script that inserts the row. Is there a preferred way of achieving this? Also, I only need ONE cdr entry per call, for billing purposes, so what I end up doing is having process_cdr as "b_only" which works great for successful calls. However, a call that is not successfully bridged results in two HTTP POST calls to my script one with is_outbound = true (as I'd expect for the b leg) but also what appears to be the a leg of the CDR (despite requesting b_only) Is this the intended behaviour of "b_only" Regards, Ross

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Invalid Application hash
by Peder 13 Sep '10

13 Sep '10

Does anybody know what this means: 2010-09-13 11:01:50.353264 [ERR] switch_core_session.c:1791 Invalid Application hash Is it saying the application named hash is invalid? Or there is an invalid hash in an application?

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polycoms and alert_info=<http://...>
by Frank Carmickle 13 Sep '10

13 Sep '10

Hello all I am trying to get the polycom 335/650 to play the file specified by alert_info. I see from my http log that it is dowloading the file each time a call is placed to the phone with the alert_info channel variable set. It doesn't actually play them. I have tried with 8k pcm wave file, a 16k pcm wave and a 8k ulaw file and no joy. Anyone know what I may be missing in my polycom configs? Thanks --Frank

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How to check user in directory
by Boris Kovalenko 13 Sep '10

13 Sep '10

Hello! I have a set of local users registered to freeswitch. So, when user is not registered with switch I get USER_NOT_REGISTERED when route call to it. But is there a way to know is user present in directory and not registered, or even isn't in directory? I want to call voicemail in first case and play 'invalid number' message in second. Is this possible? -- Regards, Boris

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SPA942 - SLA - Inbound calls
by Aloysius Lloyd 13 Sep '10

13 Sep '10

Hello All, I have difficulty with Inbound SLA. Outbound SLA working without any issue. Inbound SLA not working. FreeSWITCH Version - FreeSWITCH Version 1.0.head (git-6875a97 2010-09-08 16-33-50 -0500) Linksys SPA942 - 6.1.5(a) Once the inbound call answered the following SQL Error appear on the debug log. Please let me know how to fix this issue. 2010-09-08 23:06:25.086299 [ERR] sofia_presence.c:687 PROCESS PRESENCE EVENT 2010-09-08 23:06:25.086299 [ERR] sofia_presence.c:1827 PRES SQL select sip_from_user,sip_from_host,call_info,call_info_state,uuid from sip_dialogs where hostname='li176-12' and sip_from_user='200' and sip_from_host=' mydomain.com' 2010-09-08 23:06:25.087545 [ERR] sofia_presence.c:1848 PRES SQL select call_id,expires,sub_to_user,sub_to_host,event from sip_subscriptions where expires > -1 and hostname='li176-12' and sub_to_user='200' and sub_to_host=' mydomain.com' and (event='call-info' or event='line-seize') 2010-09-08 23:06:25.087545 [ERR] sofia_presence.c:1675 SLA3: 0 [call_id]=[ 5c0f4043-aaa7bbd0(a)192.168.240.104] 2010-09-08 23:06:25.087545 [ERR] sofia_presence.c:1675 SLA3: 1 [expires]=[1284008633] 2010-09-08 23:06:25.087545 [ERR] sofia_presence.c:1675 SLA3: 2 [sub_to_user]=[200] 2010-09-08 23:06:25.087545 [ERR] sofia_presence.c:1675 SLA3: 3 [sub_to_host]=[mydomain.com] 2010-09-08 23:06:25.087545 [ERR] sofia_presence.c:1675 SLA3: 4 [event]=[call-info] 2010-09-08 23:06:25.087545 [ERR] sofia_presence.c:1675 SLA3: 0 [call_id]=[ 696f8b47-50fa4cba(a)192.168.240.110] 2010-09-08 23:06:25.087545 [ERR] sofia_presence.c:1675 SLA3: 1 [expires]=[1284008691] 2010-09-08 23:06:25.087545 [ERR] sofia_presence.c:1675 SLA3: 2 [sub_to_user]=[200] 2010-09-08 23:06:25.087545 [ERR] sofia_presence.c:1675 SLA3: 3 [sub_to_host]=[mydomain.com] 2010-09-08 23:06:25.087545 [ERR] sofia_presence.c:1675 SLA3: 4 [event]=[call-info] 2010-09-08 23:06:25.087545 [ERR] sofia_presence.c:1863 CLEAR SQL delete from sip_dialogs where sip_from_user='200' and sip_from_host='mydomain.com' and call_info_state='seized' 2010-09-08 23:06:25.088541 [ERR] sofia_presence.c:680 STATE SQL update sip_dialogs set call_info='appearance-index=1',call_info_state='active' where hostname='li176-12' and uuid='00f17b14-2e0b-4711-b082-a7c5a90621d8' 2010-09-08 23:06:25.088541 [ERR] sofia_presence.c:687 PROCESS PRESENCE EVENT 2010-09-08 23:06:25.088541 [ERR] sofia_presence.c:1827 PRES SQL select sip_from_user,sip_from_host,call_info,call_info_state,uuid from sip_dialogs where hostname='li176-12' and sip_from_user='200' and sip_from_host=' mydomain.com' 2010-09-08 23:06:25.088541 [ERR] sofia_presence.c:1848 PRES SQL select call_id,expires,sub_to_user,sub_to_host,event from sip_subscriptions where expires > -1 and hostname='li176-12' and sub_to_user='200' and sub_to_host=' mydomain.com' and (event='call-info' or event='line-seize') 2010-09-08 23:06:25.089546 [ERR] sofia_presence.c:1863 CLEAR SQL delete from sip_dialogs where sip_from_user='200' and sip_from_host='mydomain.com' and call_info_state='seized' Thanks Lloyd

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