search for: videosupport

Hi All First time poster, long time reader. I just ran into something really bizarre. I've enabled videosupport and have been testing sip calls with Xten Eyebeam software, it works (mostly) However, when I have Videosupport=yes In the [general] section of my sip.conf, broadvoice calls fail w/ "We're sorry your call cannot be completed at this time" So... I've commented it out and trie...

Hi, I have a samba 4.6 member server joined to an 4.6 AD. There is a share named [videosupport2] with access rights for a user Videosupport. Now I’m able to connect to the share from an macOS client by share name videosupport2 AND just videosupport without the „2“ at the end …. Any hints why? Regards . Götz

Hi all, can Asterisk be used as a videoconference server or the like when enabling 'videosupport=yes' ? if so, how do I use it? is there any recommended SIP/Video-client for both Windows and Linux? Thanks, Martin

Hi all I was tryed to connect to mysip.ch scs_client by siemens that isn't works well. Does anyones knows to work H/W or S/W applictations in asterisk SIP videosupport? Regards mack_jpn

...= 5 passdb:5 auth:5 winbind:5 > > load printers = no > cups options = raw > > printing = bsd > printcap name = /dev/null > disable spoolss = yes > > [homes] > comment = Home Directories > browseable = no > writable = yes > This is where the 'videosupport' share is coming from i.e. 'videosupport's home directory. Which I have the sinking feeling is '/home/abteilungen/videosupport' > [videosupport2] > copy = template_01 > path = /home/abteilungen/videosupport > write list = videosupport > valid users = videosup...

Hello, I have trouble to enable a sip trunk with a CIRPAK. CIRPAK support answer that's there parameter are unvalid : a=silenceSupp:off - - - - is not standard and not working with cirpak - to be remove m=video 13072 RTP/AVP no video, how to remove it ? my extension.conf : exten => _6X.,1,Dial,SIP/${EXTEN:1}@x.x.x.x Regards, -- Arnaud Pignard (apignard@frontier.fr) Frontier Online -

...l allow=alaw:40 allowoverlap=yes allowsubscribe=yes callcounter=yes callingpres=allowed_passed_screen canreinvite=no context=inputinterior.se directmedia=no dtmfmode=rfc2833 faxdetect=no host=dynamic language=se nat=yes qualify=yes sendrpid=pai t38pt_udptl=no transport=udp trustrpid=yes type=friend videosupport=no [0851711201] secret=xyz callerid=Input Interior Stockholm (fax) disallow=all allow=alaw:40 allowoverlap=yes allowsubscribe=yes callcounter=yes callingpres=allowed_passed_screen canreinvite=yes context=inputinterior.se directmedia=yes dtmfmode=rfc2833 faxdetect=no host=dynamic language=se nat=n...

...llWeaver and am currently using the very same MYSQL tables (and columns) with Asterisk 1.4.11 and things are working well. The questions I have are, since new configuration variables have been added into Asterisk 1.4, can I simply add columns in my MySQL sippeers table for things such as "videosupport" "mohinterpret", etc.? When a user upgrades, how would one know if all of the possible user/peer/friend variables listed in sip.conf can in fact be pulled from Realtime? I have consulted http://www.voip-info.org/wiki-Asterisk+RealTime+Sip but that table seems to be out of date....

Hi all, I would like to connect in sip mode an Eyebeam client to a messenger via Asterisk. I want to use video. Nat is not an issue as vpn connections will be used. Is this a difficult tasks, can someone give me some pointers to get started... Have a good week-end, Francois Random Thought: --------------- Wanna buy a duck?

Does IAX2 support a video call ? Jerry

...out, trying again (Attempt #n) I looked at the sip debug stuff, and all I can see is my asterisk sending the registration packets, but no answer is received. Here's the relevant parts of my sip.conf: [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes tos=0x18 checkmwi=10 videosupport=yes allow=all relaxdtmf=yes rtptimeout=60 rtpholdtimeout=300 register => XXXXX:pass@sipgate.de/XXXXX ;XXXXX == sipgateid [XXXXX] type=friend insecure=very nat=yes username=XXXXX fromuser=XXXXX fromdomain=sipgate.de secret=pass host=sipgate.de qualify=yes -- Michiel van Baak michiel@vanbaak....

...t not vice versa. Can someone tell me where is the problem? TIA! Here's part of my configurations: ---------- sip.conf ---------- ; 113 is the Grandstream phone [113] type=friend username=113 secret=secret context=default dtmfmode = rfc2833 host = dynamic qualify = yes allow = h263 video=yes videosupport=yes ; 112 is the X-Lite phone [112] type=friend host=dynamic user=112 username=112 secret=secret allow=all nat=no ------------- extensions.conf ------------- exten => 112,1,Dial(SIP/112) exten => 112,2,Playback(vm-nobodyavail) exten => 112,102,Playback(tt-allbusy) exten => 112,103,Vo...

...able slow, pedantic checking for Pingtel ;tos=lowdelay ;tos=184 ;maxexpirey=3600 ; Max length of incoming registration we allow ;defaultexpirey=120 ; Default length of incoming/outoing registration ;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY ;videosupport=yes ; Turn on support for SIP video disallow=all ; Disallow all codecs allow=ulaw ; Allow codecs in order of preference allow=ilbc [1001] type=friend secret=1001 host=dynamic username=1001 mailbox=1001 context=local nat=no [1006] type=friend s...

...ity issues with my SIP accounts. Unauthorized people have been able to access the server (bots) and they have been able to make calls (in today's case to Cuba). Here's a copy (slightly modified) of my sip.conf: [general] context=default ; Default context for incoming calls videosupport=yes rtcachefriends=yes autocreatepeer=no t38pt_udptl=yes allowoverlap=no udpbindaddr=0.0.0.0 srvlookup=yes ;pedantic=yes disallow=all allow=alaw allow=ulaw allow=speex [1001] type=friend username=1001 secret=blah subscribecontext=default regexten=1001 callerid=&...

...phone hanging off of an TDM the audio works great. This is using the CVS code from 11/7/04. I'm trying to do video conference Friday and everything from making this work to H.323 (which is a pain anyway) isn't working out so well. Thanks for any help, Rob >From sip.conf: [general] videosupport=yes [3005] type=friend secret=XXXXXX mailbox=3005 host=dynamic nat=yes canreinvite=no disallow=all allow=gsm allow=ulaw allow=h261 allow=h263 [3007] type=friend regexten=3007 mailbox=3007 secret=XXXXXX host=dynamic nat=yes canreinvite=no disallow=all allow=gsm allow=ulaw allow=h261 allow=h263

I have success register polycom in to asterisk and it can called by other extension. But why it can't calling other extension ? and i have warning from asterisk chan.sip.c:3602 process_sdp: Unknown SDP media type in offer: application 49200 RTP/AVP 100 anyone undertand this warning ?

...192.168.1.0/255.255.255.0 externhost=pbx.DOMAIN.com (Set your external hostname name here) externrefresh=10 fromdomain=DOMAIN.com (Set your external domain name here) nat=yes qualify=yes canreinvite=no Add extra codecs to /etc/asterisk/sip_custom.conf allow=gsm allow=h261 allow=h263 allow=h263p videosupport=yes _________________________________________________________________ Windows Live?: Keep your life in sync. http://windowslive.com/explore?ocid=PID23384::T:WLMTAGL:ON:WL:en-US:NF_BR_sync:082009 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/...

...y it shows up as regg'ed nnn=is my iconnect here number xxx is my secret Thank you Jeremy [general] qualify=no register=NNNNNNNNNNN:XXXX@iconnecthere/NNNNNNNNN context=default bind = 0.0.0.0 port=5060 bindaddr=192.168.215.5 ; IP address to bind to (0.0.0.0 binds to all) srvlookup=no videosupport=no disallow=all ; First disallow all codecs allow=ulaw relaxdtmf=yes nat=yes ; NAT settings externip = 24.172.122.XXX localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks [iconnecthere] type=friend secret=pppp username=uuuuuuuu I e...

...dvoice] type=friend username=PHONE authuser=PHONE fromuser=PHONE secret=secret host=sip.broadvoice.com port=5060 context=default fromdomain=sip.broadvoice.com canreinvite=no dtmfmode=inband insecure=very permit=sip.broadvoice.com qualify=yes disallow=all allow=ulaw maxexpirey=180 defaultexpirey=160 videosupport=no exten => _9XXXXXXX,1,Dial(SIP/Broadvoice/${EXTEN:1},${SMVOICE_DIAL_LONG_TIMEOUT},mtT) exten => _91XXXXXXXXXX,1,Dial(SIP/Broadvoice/${EXTEN:1},${SMVOICE_DIAL_LONG_TIMEOUT},mtT)

...PU.BL.IC.IP tcpenable = yes limitonpeers = yes rtcachefriends = yes rtupdate=yes tos_sip=cs3 tos_audio=ef realm = testers.com autodomain=no domain=testers.com allowexternaldomains=no allowguest=no ;avpf=yes ; encryption=yes transport=ws,wss,udp icesupport=yes srvlookup=yes nat=force_rport,comedia videosupport=yes directmedia=no And here's the way I've defined my websocket peer to my sippeers table: id: 4 name: 660 ipaddr: PU.BL.IC.IP port: 5060 regseconds: 1407744248 defaultuser: 660 fullcontact: sip:660 at PU.BL.IC.IP:5060 regserver:...