bkw__~vg
c888ACTION 20:07 Invalid >=|
c88820:59 momoterr_ left before we could help!
c88820:59 forgetting the question: 06/29/17 17:35:52 i have a sip solution based on kamailio already and just need dynamic conferencing like on demand conference room for my users to join and talk to each other any idea on the approach i should take
c88800:46 mahalel left before we could help!
c88800:46 forgetting the question: 06/29/17 17:11:09 hi all, i am trying to get FS to use pgsql instead of sqlite for mod_callcenter but i don't think i'm having luck can anyone give me some pointers please?
niekniekhello! I'm researching some VoIP problems, probably not related to any of the VoIP systems but rather the network
P-NuTHi all. I'm an asterisk refugee who's come to free switch after reading the freeswitch 1.6 cookbook. I love the amount of stuff you get out of the box and the licensing is far more appealing also.
P-NuTDo you guys know of any appliances that use free switch under the hood?
niekniekSo i was thinking how SIP handles packetloss in a dialog, anyone got experience in that area?
niekniekP-Nut: sanoma uses it
niekniekp-nut: what do you want the appliance to do? be a pbx? a router? a voice recorder? etc?
niekniekP-nut: *sangoma
P-NuTniekniek: Anything, I'm just wondering where it's used.
niekniekhow do i know a bridge is transcoding? is the parameter for this?
niekniekP-Nut: sipxcom uses it
Sneliusniekniek: when ur legs have different codecs => transcoding.
SneliusP-NuT: https://caw.me for example, https://ion.team/phone.html too. Both FS based
niekniekis there any way to know when a transfer is completed?
niekniekor a way to know to debug which dialplan instruction is being executed?
MafooUKit depends how you are monitoring teh call, ESL, hook events ?
MafooUKIf you are debuggign calls, generalyl speaking you shodu lbe in fs_cli with log level 7 and sofia global siptrace on, also starting fs_cli with -S helps
niekniekwell i would really like to execute a dialplan instruction after the transfer
niekniekuuid_media_reneg to be exact
MafooUKmaybe look at teh execute_on_* varibles
niekniekit probably is possible with events, but then it's needed to constantly watch everything that is going on to know when to run it
MafooUKyou probbably want exec_after_bridge_*
MafooUKi take it you are trying to force a renegotiation to get a better codec
niekniekMafooUK: interesting suggestion, tried it, but can't find the right one
elbow~take-a-number How to get FS to generate "inband ringback" audio when bridging an already answered A side to a new call to a SIP endpoint that sends 180 but no early-media.
c88806:41 elbow, you are number 2
c88806:41 http://conference.freeswitch.org/number.jpg (autocommit)
niekniekelbow: https://wiki.freeswitch.org/wiki/180_vs._183_vs._Early_Media might do the trick
niekniekelbow: If you set ringback var and ignore_early_media, both 180 and 183 will trigger your fake ringing.
niekniekdon't know if it'll work if you're in call already
niekniekquestion: how does Freeswitch know so quickly a host does not respond to sip?
niekniekwhat will happen if the host is up but not that quick to respond?
elbowthanks @niekniek. what I'm looking for of course is 180 received on B channel -> play fake ringback to A channel / 183wSDP -> Play the B side early media through to the A side
elbowso ignore_early_media isn't excactly what I'm after
MafooUKset transfer_ringback and ringback to a tonestream you want to play during teh bridge operation
niekniekelbow: ok, i was thinking you could respond in the same way when receiving 180 & 183; just play fake media
elbow@niekniek 183 early media can be announcements etc that I want the caller to hear.
niekniekelbow ah ok
elbow@mafoouk thanks for the pointer - I'll look into those.
niekniekmafoouk: got any idea how long freeswitch waits for a trying after an invite? it's pretty damn short :)
MafooUKa lot of tonestreams are preset in vars.xml so you can do like $${uk-ring}
MafooUKniekniek no idea taht woudl be down to timeouts configured by the endpoint protocol (sofia/wss)
niekniekyeah, it's sofia
nieknieki was thinking originate_timeout but that's before we get media (wether its early media, ringing or answer)
niekniekehm progress_timeout
niekniekwhat if the sip endpoint is slow to respond, then the call would fail?
niekniekMafooUK: I learned through loglevel 9 that i actually get an icmp response that the port is unresponsive so quick that it seemed instant :)
niekniekit's close to that, it's in the 0.0000x seconds range
c88810:59 elbow left before we could help!
c88810:59 forgetting the question: 06/30/17 06:41:09 How to get FS to generate "inband ringback" audio when bridging an already answered A side to a new call to a SIP endpoint that sends 180 but no early-media.
playnet~take-a-number Why sip_rh_X-aaa in 200 OK sets in two legs, not in leg from invite?
c88811:05 playnet, you are number 2
c88811:05 http://conference.freeswitch.org/number.jpg (autocommit)
playnetsip_copy_custom_headers not set
c888ACTION 12:28 The Conference is down!
c888ACTION 12:28 The Conference is back up!
playnetwith sip_copy_custom_headers=false and sip_h headers all OK, but ph and rh broken too