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PDD Math
by Andre 28 Nov '13

28 Nov '13

How do I calculate PDD in the freeswitch CDR's? Per http://wiki.freeswitch.org/wiki/PDD In Seconds: * progress_mediasec * progresssec In milliseconds (1000 = 1 second): * progress_mediamsec if there was media * progressmsec if there was no media In MicroSeconds (1000 = 1 millisecond) * progress_mediausec * progressusec but most of my calls that are not answered are 0 on both A and B legs.

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rtp-timer-name param
by Ashwin Jain 28 Nov '13

28 Nov '13

What is "rtp-timer-name"? -- Thanks and Regards, Ashwin Jain Director (Engineering) MetroGuild

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disastery plan
by Nikita Tarasov 28 Nov '13

28 Nov '13

Hi guys. We have loadbalancer(opensips) and a huge amount of incoming calls going to ONE sip server, which is unscalable and priopretary. I need to create some kind of disastery plan for this server service is down or on high load. The main idea of this plan is, when disastery happends, to balance all incoming calls to freeswitch and keep them with music, while main server is down. On recovery calls should be forwarded to main server with the same destination number as they was. I have tryed to make it with mod_callcenter. It was done by placing calls in queue and when main server is on and re-registered on freeswitch, calls was forwarded. But without destination number. And I can't copy destination number to forwarded calls. Can you help me?

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Sound files for mod_say_he
by Zev Spitz 28 Nov '13

28 Nov '13

According to the link at http://wiki.freeswitch.org/wiki/Mod_say_he, the sound files for mod_say_he will be available soon. The page was last updated 4 April 2011. Has anything happened since? Thanks in advance, Zev Spitz

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LOL - What language should I use?
by James Mortensen 27 Nov '13

27 Nov '13

All joking aside, it's clear that the author in the FreeSWITCH docs prefers Lua. It sounds like, hands down, Lua is the answer based on information found here: http://wiki.freeswitch.org/wiki/Which_scripting_language_should_I_use%3F But why isn't Java in the list? Is Lua the fastest scripting language when only compared to Perl, Python, and JavaScript? What types of results have folks compiled from tests with mod_java? I ask because this seems to be a bit of a hole in the docs. Java is by far one of the most widely used languages (although I personally prefer something other than Java) so it seems that this should be covered too. Hope this helps and thank you! James

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pablio.h - MAX_IO_CHANNELS hardcoded to 2
by Erik M. Devane - Comms Guy 27 Nov '13

27 Nov '13

Dear all, I've been fighting an issue with PortAudio on CentOS for a few days now: I have two Delta 1010 cards which I'm using with mod_portaudio streams. On Windows, the cards appear as eight two-channel devices, and FreeSWITCH can use them. On CentOS, the cards appear as two eight-channel devices (actually 10 out/12 in) and I'm unable to configure more than two channels per card. Looking at the code, it looks like pablio.h's MAX_IO_CHANNELS is hardcoded to 2, and this is the limit of mod_portaudio. As it's a ring-buffer, I can see why it might need to be a power of 2, but does anyone know why it is set to 2? How dangerous would it be to increase this value? Any thoughts? Regards, Erik

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fail_on_single_reject ignored with LCR endpoint
by Victor Chukalovskiy 27 Nov '13

27 Nov '13

Hi, I rely on fail_on_single_reject = INCOMPATIBLE_DESTINATION to make sure calls don't fail-over to other routes. INCOMPATIBLE_DESTINATION is returned as the first mod_lcr route. Call rejection works this way: <action application="set" data="fail_on_single_reject=INCOMPATIBLE_DESTINATION"/> <action application="lcr" data="2411765512 test_profile"/> <action application="bridge" data="${lcr_auto_route}"/> However, when call is bridged to lcr endpoint, fail_on_single_reject is ignored. Call gets failed-over to the second route after hitting INCOMPATIBLE_DESTINATION: <action application="set" data="fail_on_single_reject=INCOMPATIBLE_DESTINATION"/> <action application="bridge" data="lcr/test_profile/2411765512"/> Is this a bug or something by-design??? Here is an example of LCR routes involved: | Digit Match | Carrier | Rate | Codec | CID Regexp | Limit | Dialstring | | 241 | 488_REJECTION | 0.00001 | | | ld_carriers 488_REJECTION -1 | [lcr_carrier=488_REJECTION,lcr_rate=0.00001]error/INCOMPATIBLE_DESTINATION | | 241 | CARRIER_1 | 0.40240 | | | ld_carriers CARRIER_1 -1 | [lcr_carrier=CARRIER_1,lcr_rate=0.40240]sofia/gateway/CARRIER_1_GW_1/+2411765512 | Thank you, Victor

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ClueCon Weekly Nov 27th , Thanksgiving Edition...
by Ken Rice 27 Nov '13

27 Nov '13

Hey GUys, Join us on 888 (sip:888@conference.freeswitch.org)... Today's topic is pretty light hearted... General Community Scrum, Thanks to all the contributors over the years, and Open Discussion... IF you havent Joined us in the past, join us today and see what its all about! IF you need a PSTN dialin see http://fs0.us/call888 Ken http://freeswitch.org

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Cannot Blind Transfer 1 Legged calls
by assgeo8＠gmail.com 27 Nov '13

27 Nov '13

Hi, I've be trying to get the attended transfers to work without success. Freeswitch setup is 1.2-stable with the default configuration. I have 3 yealink phones 1001-1002-1003. I call from 1001 to 1002 - OK I initiate the attended transfer from 1002 to 1003 - OK (it's just like another call & 1001 gets moh) Then I transfer 1001 to 1003 but it fails, returning the call to 1002 with the message 'transfer failed" Looking at the logs i found this error : Cannot Blind Transfer 1 Legged calls prior to Processing Extension 1001 <1001>->1003 in context public Why this error? Maybe it's because the processing goes to context public? Regards, Jeff Here's the logs from the CLI : 2013-11-27 09:31:30.033871 [NOTICE] switch_channel.c:1034 New Channel sofia/internal/1001@<FS_IP> [5071607c-573e-11e3-a1ae-5b62f41fde95] 2013-11-27 09:31:30.153836 [INFO] mod_dialplan_xml.c:558 Processing 1001 <1001>->1002 in context default ... 2013-11-27 09:31:30.173818 [NOTICE] switch_channel.c:1034 New Channel sofia/internal/sip:1002@<MY_IP>:5062 [5085d84a-573e-11e3-a1cd-5b62f41fde95] 2013-11-27 09:31:30.373828 [NOTICE] sofia.c:5847 Ring-Ready sofia/internal/sip:1002@<MY_IP>:5062! 2013-11-27 09:31:30.373828 [INFO] switch_ivr_originate.c:1190 Sending early media 2013-11-27 09:31:30.373828 [NOTICE] sofia_glue.c:4351 Pre-Answer sofia/internal/1001@<FS_IP>! 2013-11-27 09:31:31.833844 [NOTICE] sofia.c:6508 Channel [sofia/internal/sip:1002@<MY_IP>:5062] has been answered 2013-11-27 09:31:31.853914 [NOTICE] switch_ivr_originate.c:3437 Channel [sofia/internal/1001@<FS_IP>] has been answered 2013-11-27 09:31:38.173834 [NOTICE] switch_channel.c:1034 New Channel sofia/internal/1002@<FS_IP> [554aecbc-573e-11e3-a1d6-5b62f41fde95] 2013-11-27 09:31:38.353860 [INFO] mod_dialplan_xml.c:558 Processing 1002 <1002>->1003 in context default ... 2013-11-27 09:31:38.373825 [NOTICE] switch_channel.c:1034 New Channel sofia/internal/sip:1003@<MY_IP>:5063 [55688d1c-573e-11e3-a1f5-5b62f41fde95] 2013-11-27 09:31:38.673887 [NOTICE] sofia.c:5847 Ring-Ready sofia/internal/sip:1003@<MY_IP>:5063! 2013-11-27 09:31:38.693829 [INFO] switch_ivr_originate.c:1190 Sending early media 2013-11-27 09:31:38.693829 [NOTICE] sofia_glue.c:4351 Pre-Answer sofia/internal/1002@<FS_IP>! 2013-11-27 09:31:42.033830 [NOTICE] sofia.c:6508 Channel [sofia/internal/sip:1003@<MY_IP>:5063] has been answered 2013-11-27 09:31:42.053816 [NOTICE] switch_ivr_originate.c:3437 Channel [sofia/internal/1002@<FS_IP>] has been answered 2013-11-27 09:31:45.493836 [NOTICE] switch_channel.c:1034 New Channel sofia/internal/sip:1003@<FS_IP> [59a6f1a2-573e-11e3-a1ff-5b62f41fde95] 2013-11-27 09:31:45.493836 [NOTICE] switch_channel.c:1034 New Channel sofia/internal/1001@<FS_IP> [59a71ad8-573e-11e3-a201-5b62f41fde95] 2013-11-27 09:31:45.513831 [INFO] mod_dialplan_xml.c:558 Processing Extension 1001 <1001>->1003 in context public 2013-11-27 09:31:45.513831 [ERR] sofia.c:7281 Cannot Blind Transfer 1 Legged calls 2013-11-27 09:31:45.613825 [NOTICE] mod_sofia.c:2649 Hangup sofia/internal/1001@<FS_IP> [CS_EXECUTE] [BLIND_TRANSFER] 2013-11-27 09:31:45.613825 [NOTICE] sofia.c:6563 Hangup sofia/internal/sip:1003@<FS_IP> [CS_CONSUME_MEDIA] [NO_USER_RESPONSE] 2013-11-27 09:31:45.613825 [NOTICE] switch_core_session.c:1564 Session 48 (sofia/internal/1001@<FS_IP>) Ended 2013-11-27 09:31:45.613825 [NOTICE] switch_core_session.c:1568 Close Channel sofia/internal/1001@<FS_IP> [CS_DESTROY] 2013-11-27 09:31:45.633835 [NOTICE] switch_core_session.c:1564 Session 47 (sofia/internal/sip:1003@<FS_IP>) Ended 2013-11-27 09:31:45.633835 [NOTICE] switch_core_session.c:1568 Close Channel sofia/internal/sip:1003@<FS_IP> [CS_DESTROY] 2013-11-27 09:32:02.373827 [NOTICE] sofia.c:716 Hangup sofia/internal/sip:1003@<MY_IP>:5063 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2013-11-27 09:32:02.393821 [NOTICE] switch_ivr_bridge.c:1569 Hangup sofia/internal/1002@<FS_IP> [CS_EXECUTE] [NORMAL_CLEARING] 2013-11-27 09:32:02.393821 [NOTICE] switch_core_session.c:1564 Session 46 (sofia/internal/sip:1003@<MY_IP>:5063) Ended 2013-11-27 09:32:02.393821 [NOTICE] switch_core_session.c:1568 Close Channel sofia/internal/sip:1003@<MY_IP>:5063 [CS_DESTROY] 2013-11-27 09:32:02.393821 [NOTICE] switch_core_session.c:1564 Session 45 (sofia/internal/1002@<FS_IP>) Ended 2013-11-27 09:32:02.393821 [NOTICE] switch_core_session.c:1568 Close Channel sofia/internal/1002@<FS_IP> [CS_DESTROY] 2013-11-27 09:32:04.033828 [NOTICE] sofia.c:716 Hangup sofia/internal/1001@<FS_IP> [CS_EXECUTE] [NORMAL_CLEARING] 2013-11-27 09:32:04.053845 [NOTICE] switch_ivr_bridge.c:721 Hangup sofia/internal/sip:1002@<MY_IP>:5062 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2013-11-27 09:32:04.053845 [NOTICE] switch_core_session.c:1564 Session 44 (sofia/internal/sip:1002@<MY_IP>:5062) Ended 2013-11-27 09:32:04.053845 [NOTICE] switch_core_session.c:1568 Close Channel sofia/internal/sip:1002@<MY_IP>:5062 [CS_DESTROY] 2013-11-27 09:32:04.053845 [NOTICE] switch_core_session.c:1564 Session 43 (sofia/internal/1001@<FS_IP>) Ended 2013-11-27 09:32:04.053845 [NOTICE] switch_core_session.c:1568 Close Channel sofia/internal/1001@<FS_IP> [CS_DESTROY]

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siprec question
by gk 26 Nov '13

26 Nov '13

Is it possible to configure FreeSwitch so that sip calls can be recorded with an external recording application? I've read about OrkAudio, but that seems to use passive recording via span port mirroring. I'm looking to perform active sip recording through a connected socket/port. That would require FreeSwitch to establish a socket connection with the recorder, but I'm not sure if that's something FreeSwitch can do. Thanks. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/siprec-question-tp7596002.html Sent from the freeswitch-users mailing list archive at Nabble.com.

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