- FreeSWITCH-users - FreeSWITCH Mailing Lists

How to check whether incoming call is authenticated
by Artur Mega 20 Jan '16

20 Jan '16

In our external.xml: <domains> <domain name="all" alias="false" parse="true"/> </domains> <param name="auth-calls" value="false"/> It means, that everyone can make call from outside to our freeswitch. Is it possible to know if caller already authenticated in our freeswitch? (in a dialplan) Thanks. -- Arthur

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Call progress not being passed
by David Witham 20 Jan '16

20 Jan '16

Hi all, We're running the latest FREESwitch 1.4 (1.4.26) on debian7. We have a problem where we receive a SIP 183 from our upstream provider (via a sofia gateway) but the 183 does not get forwarded to the internal leg we bridge to. When we receive the 200 OK, that gets passed through and the call is established with media. We can add a ring_ready into the dialplan and force it to pass a 180 back to the A party but that just masks the problem. Has anyone else come across this behaviour? I haven't found any JIRAs that match this although perhaps FS-7714 is related. thanks, David

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Has anyone else experienced this issue - presence leaked / notifies for non subscribers
by Podrigal, Aron 19 Jan '16

19 Jan '16

https://freeswitch.org/jira/browse/FS-8753 -- Aron Podrigal - '1000001', '1110010', '1101111', '1101110' '1010000', '1101111', '1100100', '1110010', '1101001', '1100111', '1100001', '1101100' P: '2b', '31', '33', '34', '37', '34', '35', '38', '36', '30', '39', '39'

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hiredis module
by John Nash 19 Jan '16

19 Jan '16

I saw this module https://freeswitch.org/confluence/display/FREESWITCH/mod_hiredis I am looking forward to insert some data in REDIS from lua scripts and access it but I could not find any documentation on commands available. Above links only show how to configure freeswitch hiredis module. Is there any documentation in progress?

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UNALLOCATED_NUMBER
by Deepika Yadav 19 Jan '16

19 Jan '16

Hi, I am facing an issue where when I call a mobile phone number from a SIP account using Freeswitch via an external gateway(service provider) I get these logs and call to the recipient is not made: Logs : 2016-01-19 08:50:21.042365 [NOTICE] switch_channel.c:1055 New Channel sofia/external/919811806537 [ac043fa4-be89-11e5-b585-450e5c1ead46] 2016-01-19 08:50:21.182405 [NOTICE] sofia.c:7539 Hangup sofia/external/XXXXXXXX [CS_CONSUME_MEDIA] [UNALLOCATED_NUMBER] 2016-01-19 08:50:21.202390 [NOTICE] switch_core_session.c:1641 Session 6 (sofia/external/XXXXXXXX) Ended 2016-01-19 08:50:21.202390 [NOTICE] switch_core_session.c:1645 Close Channel sofia/external/919811806537 [CS_DESTROY] I also tried the following command : sofia profile external restart reloadxml but did not work I tried using some other gateway then it worked fine but how can I remove this issue so that I need not change my gateway service provider. Regards, Deepika

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Log jitter and package loss
by Michael Nielsen 19 Jan '16

19 Jan '16

Is it possible to see package loss and jitter on calls made in FreeSWITCH - and log the information (perhaps along the CDR's)?

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Call Return
by Blake Priddy 19 Jan '16

19 Jan '16

So I have this school that FS is installed at in Arkansas. They want this feature that they had before with their last install called call return? Basically from what I can interpret from him is that when a user transfers to another user and they do not answer, the receptionist wants the call to ring back to her so she can either take a message or direct the call elsewhere. Without the caller having to leave a voicemail, hang up, then call back to tell the receptionist that they were not available. Does anyone know the feature I am talking about and how can I implement this into FS. Thanks y'all! :) -- Blake Priddy Cogents Technologies, LLC. (501) 786-1434 blake(a)cogents.io http://cogents.io

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Seeking Real World Experience with Answering Machine Detection mod_com_amd
by Chris Brown 19 Jan '16

19 Jan '16

Greetings All, As the subject notes, looking for real world experience using the Commercial FreeSWITCH Answering Machine Detection module described here: https://freeswitch.org/confluence/display/FREESWITCH/mod_com_amd In particular, curious about the average percentage of successful answering machine detections and the number of simultaneous channels that can use the module on a single FreeSWITCH instance. Thanks much in advance! Chris

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Seeking Real World Experience with Answering Machine Detection mod_com_amd
by Chris Brown 19 Jan '16

19 Jan '16

Greetings All, As the subject notes, looking for real world experience using the Commercial FreeSWITCH Answering Machine Detection module described here: https://freeswitch.org/confluence/display/FREESWITCH/mod_com_amd In particular, curious about the average percentage of successful answering machine detections and the number of simultaneous channels that can use the module on a single FreeSWITCH instance. Thanks much in advance! Chris

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lost timezones
by Денис 19 Jan '16

19 Jan '16

After restart FS via service freeswitch restart we loose zones and all dialplans with time-intervals was broken. BUT reloadxml or simple reload spandsp (or other module who use timezones) helps. And sometimes we loose timezones in work 1) how to check timezones state in the work, via api/fs_cli 2) how to detect 'why we loose it' latest 1.4 FS, from off. yum repo, centos 6, 86-64, configs in DB, load via mod_curl_xml, in loader no errors, reload helps.

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