- FreeSWITCH-users - FreeSWITCH Mailing Lists

DETECT ACTIVE CHANNELS
by Shan Randhawa 26 Jan '16

26 Jan '16

Hi, i need some help. Is there a way in scripting like in lua like using freeswitch.API() through which i can get active channels . I can get active channels in fs cli by running command "active channels".But how to do it in scripts. Thanks. Regards, Shan randhawa

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documents for mod_verto
by Siju Nair 26 Jan '16

26 Jan '16

Hi Team, i am trying mod_verto, i didn't find more documents regarding this, can i have a link or pdf file where i can see in depth details of mod_verto. Thanks, Siju Nair

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Control Success with rxfax
by Adam Sherman 26 Jan '16

26 Jan '16

Good Morning, Is it possible to control whether the rxfax application sends a success code back to the far end? I would like to have a setup where the calling fax machine only thinks there was success if my mod_python script successfully submits the file to our mail server. Thanks, A. -- Adam Sherman | Versature | CTO | 613-317-1212 | asherman(a)versature.com | www.versature.com <http://www.versature.com?utm_source=Signature&utm_medium=Email&utm_campaign…> <https://www.linkedin.com/company/versature> <https://www.facebook.com/Versature> <https://twitter.com/Versature/> <http://inside.versature.com> <https://plus.google.com/+Versature> VERSATURE LATEST NEWS : Improving Business Operations with Detailed Call Analytics from Versature <http://www.versature.com/conversation-blog/improving-business-operations-wi…> Versature Gives Back - Year in Review <http://www.versature.com/conversation-blog/versature-gives-back/?utm_source…> Configuring your Business Communications for the Holidays <http://www.versature.com/conversation-blog/business-communications-for-the-…>

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Random calls failing with WRONG_CALL_STATe when using TLS
by Emrah 26 Jan '16

26 Jan '16

Hi all, I am facing a very frustrating issue. I often have to dial twice when using my Yealink phone with TLS because the first attempt times out. The logs on the Yealink indicate that the first invite is successfully received, to which my FS sends a 100 trying and 407 proxy auth required. It is subsequently when my phone sends back the invite that the connection crashes with the following error: SSL ERROR SYSCALL Is this something common? Why does the SSL connection crashes when the phone attempts to send the second invite? My phone is behind NAT. It is going to be a crazy expedition to collect the logs and Pastebin them, so I am tempting my luck on the list first to see if you have any pointers. As a last piece, my Bria on my iPHone, among other clients, never had this issue. I did experience it from time to time with Blink on Mac OS X. Any help appreciated. Emrah

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Freeswitch is crashing often.
by Brian Beebe 25 Jan '16

25 Jan '16

We have Freeswitch setup with bypass-media and we have 2 HA pairs. Since we have been in production we have had a few segfaults. Jan 25 09:05:09 freeswitch-pl1a kernel: freeswitch[30906]: segfault at 30 ip 00007ff5f770cdc0 sp 00007ff5ea7073a8 error 4 in libfreeswitch.so.1.0.0[7ff5f769d000+1bf000] Jan 21 23:13:45 freeswitch-pl1a kernel: freeswitch[18879]: segfault at 30 ip 00007f19ad23fdc0 sp 00007f19a42833a8 error 4 in libfreeswitch.so.1.0.0[7f19ad1d0000+1bf000] Jan 21 13:09:42 freeswitch-pl2b kernel: fs_cli[3863]: segfault at 3134264fc0 ip 00007f6270c4dd1c sp 00007ffd19738690 error 4 in libc-2.17.so[7f6270bd0000+1b6000] Jan 21 12:32:29 freeswitch-pl2a kernel: freeswitch[2594]: segfault at 30 ip 00007f50e1e4edc0 sp 00007f50d8e573a8 error 4 in libfreeswitch.so.1.0.0[7f50e1ddf000+1bf000] Jan 21 09:10:44 freeswitch-pl2a kernel: fs_cli[9629]: segfault at 10 ip 00007f0710166bb0 sp 00007f07107abcf8 error 4 in libpthread-2.17.so[7f071015c000+16000] Jan 19 14:30:54 freeswitch-pl2a kernel: fs_cli[19996]: segfault at 10 ip 00007f29026e7bb0 sp 00007f2902d2ccf8 error 4 in libpthread-2.17.so[7f29026dd000+16000] Jan 19 09:51:33 freeswitch-pl1a kernel: freeswitch[24917]: segfault at 30 ip 00007fc974829dc0 sp 00007fc95f80d3a8 error 4 in libfreeswitch.so.1.0.0[7fc9747ba000+1bf000] The latest was this morning and we were able to get a crash file. The fscore_pb output is located at http://pastebin.freeswitch.org/24538 It seems to be happening in the switch_core_session_get_channel call. Any help would be appreciated. Thanks, Brian

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Profile won't start after stop, doesn't properly release IP/Port
by Luiz Fernando Softov 25 Jan '16

25 Jan '16

Hi all, i'm trying to do a stop then a start in a profile. After profile stop, the IP/Port doesn't release I Google it and found this JIRA https://freeswitch.org/jira/browse/FS-6603 But this is when i'm making a unload mod_sofia, the port release So, only after this i can start the profile I'm using FreeBRS, and in my other programs I use SO_REUSEPORT There is a function (su_setreuseaddr) in libs/sofia-sip/libsofia-sip-ua/su/su.c where i found a #define SO_REUSEPORT My question is, there a correct place to define SO_REUSEPORT, or i just add it to make of the system and the SO_REUSEPORT will work? Or this isn't related with the fact that the port isn't released by the module when i make a stop? If not, how can i make to stop and start profile without problem?

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subevents from CHANNEL_EXECUTE
by Денис 25 Jan '16

25 Jan '16

I need make some checks for callcenter and better event - CHANNEL_EXECUTE. Can i set another fields in lua bind? Now in call logs - many 2016-01-25 19:35:49.890541 [DEBUG] mod_lua.cpp:459 lua event hook: execute 'hook/event-CHANNEL_EXECUTE-for-callcenter-recording.lua' 2016-01-25 19:35:49.890541 [DEBUG] mod_lua.cpp:459 lua event hook: execute 'hook/event-CHANNEL_EXECUTE-for-callcenter-recording.lua' 2016-01-25 19:35:49.890541 [DEBUG] mod_lua.cpp:459 lua event hook: execute 'hook/event-CHANNEL_EXECUTE-for-callcenter-recording.lua' ... How can i reduce this out? (i need run recording on queue before any agent answers, based on some fields in A leg, without callcenter record property) from event - i need fields -- variable_current_application: callcenter -- variable_current_application_data: 000*111%40domain -- Application: callcenter -- Application-Data: 000*111%40domain -- Application-UUID: 8270b92a-bdc5-11e5-9a68-b50922e2fa07

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mod_cdr_csv vs mod_xml_cdr
by John Nash 25 Jan '16

25 Jan '16

Hello, I am currently using mod_cdr_csv and a perl script (every 5 min) to insert into DB. I tested with live traffic of around 500 CDR per hour but every hour I see a difference of 4-5 CDR with my carrier. These CDR are of various durations ranging from 1 second to 600 seconds. I dont know how to debug this. As many suggested in mailing list that mod_xml_cdr is a better choice? I understand that XML cdr is good option where freeswitch instances are in many locations but Can the issue I am facing be related to mod_cdr_csv? Regards, John

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zrtp HS80 not supported?
by Peter Villeneuve 24 Jan '16

24 Jan '16

Hi guys, I'm testing out FS' zrtp implementation with a zrtp enabled softphone and I found the following in the FS logs: 2016-01-24 18:59:31.761752 [DEBUG] switch_rtp.c:1370 [zrtp respond]: Preparse incoming COMMIT. Remote peer wants DH3k:2 mode lic=1 peer M=0. 2016-01-24 18:59:31.761752 [DEBUG] switch_rtp.c:1370 [zrtp respond]: INFO: Authtag HS80 isn't supported by profile. ID=0 2016-01-24 18:59:31.761752 [DEBUG] switch_rtp.c:1370 [ zrtp engine]: Enter InitiatingError State with ERROR:, notification Enabled. ID=0 Looking through the source at zrtp_crypto_atl.c I see that HS80 should be supported as well as the more common HS32: atl80->base.id= ZRTP_ATL_HS80; Anyone know what I'm doing wrong? I would expect FS to detect the HS_80 hash length and negotiate zrtp with the endpoint but it doesn't seem to be working. Is there any config file for zrtp in FS that I should be looking at? Any clues? Cheers, Peter

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Re: [Freeswitch-users] Learning FreeSWITCH - Can't call into test installation
by Kim Culhan 24 Jan '16

24 Jan '16

> I am new to freeSWITCH, and working trough the examples from the > FreeSWITCH (1.2) Book. > > I have the system built, and running, and everything looks OK. The > system registers with my VoIP provider (voip.ms), an IP phone registers > with FreeSWITCH, and I can call out to the POTS. However, when I try to > call into the IP phone, I can see the incoming call information at > fs_cli, but it instantly hangs up, and never goes through to the > intended destination set up in dialplan/public/01_voipms.xml. The > message that I get is > > [NOTICE] switch_core_state_machine.c:315 > sofia/external/613XXXXXXX(a)158.85.70.XXX has executed the last dialplan > instruction, hanging up Suggest you take a look at: https://wiki.freeswitch.org/wiki/Dialplan_XML#Example_9:_Routing_DID_to_an_… Since is has 'executed the last dialplan instruction' it has failed to find a match within any of the dialplan instructions and it hangs up. Take a look back through the lines appearing in your fs_cli and try to find the line containing a reference to '01_voipms.xml' and you will probably notice the problem a few characters after: 'Regex (FAIL)' This will be where your incoming voip.ms phone number fails to match what you have in the dialplan intended destination setup. Hope this helps. -kim

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