nblrdebbient_: first i provisioned it manually (basically while i was still exploring the pitfalls) nowadays the config is created by a software that also provides users a web ui.
nblrdebbient_: touchless bootstrapping was a bit tricky, as I needed to provision a TLS cert first in order to secure the rest of the provisioning process, but i solved that by delivering a very basic config that contains the CA cert and the vlan option, to bump the phone into the voice vlan after reboot
nblrdebbient_: in the voice vlan, the dhcpd serves an https provisioning url that serves the actual config (one generic config and one with per-phone line settings) - todo: deliver client certs in the first step to prevent fetching configs by unauthorized phones... but that would involve auth in the httpd... there are some other things on my to do before tackling that.
nblrdebbient: what really helped me was a look at http://voipt2.polycom.com/560/dirindex.php (and of course 3725-42644-002a-ucsoftware-administrator-guide-5-6-0.pdf which lists all available parameters)
t4nk724~take-a-number hey why gateway not work unless gateway is set to the domain name?
c88822:05 t4nk724, you are number 5
c88822:05 http://conference.freeswitch.org/number.jpg (autocommit)
t4nk724ok tAHNKS C888
t4nk724oops
rneesehey guys working on freeswitch for stretch and what is libyuv used for
rneeseas we have arm devices that need updating arm servers no little boxes
t4nk724dont no i have it working unless i cahnge the gateway to something that is not a domain it not work again
t4nk724rneese: do you no?
t4nk724Work around is to manually run: apt-get install libyuv
t4nk724i just want to name the gateway whatever i want like every example
t4nk724but if i set real and or proxy it will even break it
t4nk724s/real/realm
t4nk724so i dunno
c88801:34 t4nk724 left before we could help!
c88801:34 forgetting the question: 10/30/17 22:05:20 hey why gateway not work unless gateway is set to the domain name?
c88802:28 MatthewM left before we could help!
c88802:28 forgetting the question: 10/30/17 16:50:49 Has anyone seen SRTP scalability issues? All is good until ~150 SRTP (PCMU webrtc via kamailio) and then Freeswitch has issues seeing audio from Clients. RTCP packets has FS saying massive packet loss from Clients to FS but not in other direction. webrtc-internals shows RTT times jump from 100ms to 2-3 seconds. Clients connected using softphones and RTP have no issues. CPU not loaded.
Guest88045~take-a-number Is there an event I can monitor to see when a call is answered from the FIFO queue, ideally somethign i can use in the XML dialplan
c88805:35 Guest88045, you should change your name to something more descriptive, use /nick <desired nick>
c88805:35 Guest88045, you are number 4
c88805:35 http://conference.freeswitch.org/number.jpg (autocommit)
SwK~waiting
c88809:46 1 jaeckel: (10/19/17 11:58:35 I'm currently trying to get a FreeSWITCH/HylaFAX/gofaxip installation to run, but it looks like the T.38 call to my provider somehow isn't successful. I took the same installation&config and added our asterisk that also supports T.38, but there it works OOTB... any hints where I can start to debug the issue?)
c88809:46 2 pRiVi: (10/21/17 14:16:33 The bug https://freeswitch.org/jira/browse/FS-10655 is still open and not looked for monthes... My question about paying for resolving had no response, too. So it is impossible to get such big bugs resolved?)
c88809:46 3 Fred_: (10/30/17 07:17:28 when does api_reporting_hook fire,  is it always and only after hangup?)
c88809:46 4 Guest88045: (10/31/17 05:35:15 Is there an event I can monitor to see when a call is answered from the FIFO queue, ideally somethign i can use in the XML dialplan)
c88809:46
c88809:46 http://conference.freeswitch.org/number.jpg
SwK~pop 3
c88809:46 3 Fred_: (10/30/17 07:17:28 when does api_reporting_hook fire,  is it always and only after hangup?)
c88809:46 don't forget who helped you, check out SwK's wishlist:
c88809:46 http://amzn.com/w/38CG44Z1MZA87
c88809:46 (autocommit)
SwKwell fred's appearently fone
SwKs/fone/gone/
c88809:46 What SwK meant to say was... well fred's appearently gone
SwKpRiVi, hey still around?
SwKpRiVi, you can always email consulting@freeswitch.org for commercial support to get your problems looked at...
c888ACTION 09:48 E-Mail consulting@freeswitch.org or call +1-213-286-0400 for info about commercial consulting!
pRiViSwK: okay...
pRiVithanks
polysicshello!
polysics~take-a-number can anyone think of a reason why we are getting calls originated out, nothing for 3 minutes, then a 102 error code?
c88812:23 polysics, you are number 4
c88812:23 http://conference.freeswitch.org/number.jpg (autocommit)
polysicsit's 3 minutes almost on the second so it HAS to be something timing out
polysicsthere does not seem t be a 3 minute timeout anywhere
Oooohboy~take-a-number hello all, I'm working with a javascript script executed from the dialplan that uses session.execute to start an eavesdrop session. When the eavesdrop session ends, however, the script execution stops. Is there any way to pass control back to the javascript once the eavesdrop session is finished?
c88814:10 Oooohboy, you are number 5
c88814:10 http://conference.freeswitch.org/number.jpg (autocommit)
c88816:44 Oooohboy left before we could help!
c88816:44 forgetting the question: 10/31/17 14:10:26 hello all, I'm working with a javascript script executed from the dialplan that uses session.execute to start an eavesdrop session. When the eavesdrop session ends, however, the script execution stops. Is there any way to pass control back to the javascript once the eavesdrop session is finished?
cresl1nSo I'm guessing the new bridging code might cause ast_channel_connected(callee)->id.number.str to return something different between 11 and 13?
cresl1nosnap, wrong room :-)
cresl1nSorry guys
milan~take-a-number when making an outbound call using api, i am trying to set sip_h_History-Info header. Works fine. My problem is to add ;index=1 for example. How can i escape semicolon?
c88818:02 milan, you are number 5
c88818:02 http://conference.freeswitch.org/number.jpg (autocommit)
TylerPis anyone handy with DNS SRV? I have setup _sip._tcp dns srv records, but phones are registering with UDP every time instead of TCP
TylerPmy DNS provider doesn't offer NAPTR
c88819:13 milan left before we could help!
c88819:13 forgetting the question: 10/31/17 18:02:23 when making an outbound call using api, i am trying to set sip_h_History-Info header. Works fine. My problem is to add ;index=1 for example. How can i escape semicolon?
c88819:16 polysics left before we could help!
c88819:16 forgetting the question: 10/31/17 12:23:16 can anyone think of a reason why we are getting calls originated out, nothing for 3 minutes, then a 102 error code?