BoteMan~888
mchammerdadafter fooling around with trying to figure out why T38 wasn't working, I did get this: My ATA is re-sending a SIP invite with the media set at T38. However my installation is rejecting that with: "488: Not Acceptable Here" how can I change or find whats wrong with my settings?
mchammerdadWhen I send a fax "from" my fax machine I'm not actually getting a T.38 SDP either though. I guess that means My ATA isn't configured correctly.
kitsunemchammerdad: Who's your carrier?
kitsuneIf you don't mind me asking
kitsuneTo answer your question, it depends on who is issuing the 488. FreeSWITCH doesn't always allow a call to Re-INVITE to T.38.
mchammerdadflowroute
cdhavalHow to check user is registered or not in C module? Can i get any value or status in variable in my C module?
mishehuseriously man. you might just want to hire somebody to write this stuff for you
mishehuall the C documentation is there on teh website or you can generate it yourself with doxygen
BeeBuuhello,all. I need a help on how to using playSIP to make a call to a client and record it...thanks.
clive-Hi. Is anyone here using astpp ?
stugsterhey all, is there an easy way to delete all voicemails for an extension through fs_cli?
stugsterI should have taken a number...
stugster~take-a-number is there an easy way to delete all voicemails for a given extension through fs_cli or alternative?
c88808:50 stugster, you are number 2
c88808:50 http://conference.freeswitch.org/number.jpg (autocommit)
stugster~popme
c88809:15 2 stugster: (02/05/18 08:50:02 is there an easy way to delete all voicemails for a given extension through fs_cli or alternative?)
c88809:15 (autocommit)
stugsterfreeswitch@newsip> vm_delete 301@mydomain
stugstersorted.
c888ACTION 09:22 The Conference is down!
c888ACTION 09:23 The Conference is back up!
c888ACTION 09:23 The Conference is down!
c888ACTION 09:23 The Conference is back up!
rossbcanTalking clock (mod_say_en) not working. Immediate hangup. mod_say_en enabled. libspeex and linspeexdsp installed. No SIGSEGV's. How to debug, see chain of events. Loglevel already debug..
MafooUKprobbably log of the call see - https://freeswitch.org/confluence/display/FREESWITCH/Creating+a+freeswitch.log+pastebin
rossbcanCall log extract: Action answer(); Action sleep(1000); Action say(${default_language} CURRENT_DATE_TIME pronounced ${strepoch()}); Action hangup()
rossbcanmod_say_en problem solved: wrong path (<X-PRE-PROCESS cmd="include" data="lang/en/*.xml"/>) should be <X-PRE-PROCESS cmd="include" data="language/en/*.xml"/> in freeswitch.xml
JustLearningFSHello dear FS Users
JustLearningFSCan anyone help me? I have 3 different VOip Providers each one have an specific dial plan prefix, how can i setting up an outbound route using the 3 different providers as a failover
JustLearningFSThanks in advanced
BoteManJustLearningFS: Perhaps this will help you. But beware that there are different ways for a call to fail, not all of which can be detected this way.
BoteManhttps://freeswitch.org/confluence/display/FREESWITCH/XML+Dialplan#XMLDialplan-Example7:Actionfailoveronfailedaction
JustLearningFSHi there, i had this:https://thepasteb.in/p/oYhlGP99PXxtZ and Boteman suggest me to do this: <action application="bridge" data="sofia/gateway/provider1/${prefix1}${dialstring}|sofia/gateway/provider2/${prefix2}${dialstring}"/> in order to solve my trouble, the thing is that i use FusionPBX to handle my server, can anyone to tellme how to setting u
JustLearningFSp an outbound route using diferent voip providers?, each one use a different outbound prefix