search for: textsupport

Hi, i have try to send IM from Client A (Ekiga) to Client B (Ekiga). I have enable the textsupport in the sip.conf. I used this "How to": http://www.voip-info.org/wiki/view/Asterisk+Documentation+1.6.1+realtimetext.txt sip.conf [general] [...] disallow=all allow=ulaw allow = alaw allow=t140 allow=t140red textsupport = yes videosupport = yes...

...other, sound comes through just fine. So my hunch is that is something to do with the audio supplied by the server. Do I need to have alsa installed?? Any hint? sip.conf: [general] context = unauthenticated bindport = 5060 bindaddr = 0.0.0.0 tcpbindaddr = 0.0.0.0 tcpenable = yes videosupport = no textsupport=yes alwaysauthreject=yes allowguest=no [1001] ; grandstream 1 context = home type = friend callerid = One <1001> secret = XYZ host = dynamic mailbox = 1001 disallow = all allow = ulaw transport = udp dtmfmode=auto ; accept touch-tones from the devices, negotiated automatically nat=forc...

...ltip: NULL rtptimeout: NULL rtpholdtimeout: NULL sendrpid: NULL outboundproxy: PU.BL.IC.IP timert1: NULL timerb: NULL qualifyfreq: NULL constantssrc: NULL contactpermit: NULL contactdeny: NULL usereqphone: NULL textsupport: NULL faxdetect: NULL buggymwi: NULL auth: NULL fullname: NULL trunkname: NULL cid_number: NULL callingpres: NULL mohinterpret: NULL mohsuggest: NULL parkinglot: NULL hasvoicemail: NULL subscribemwi...

...output? The response has no Eventlist: start Ej. Response: Success Eventlist: start Message: Peer status list will follow Event: PeerEntry Channeltype: SIP ObjectName: 1001 ChanObjectType: peer IPaddress: 192.168.175.1 IPport: 63772 Dynamic: yes Natsupport: no VideoSupport: no TextSupport: no ACL: no Status: Unmonitored RealtimeDevice: yes Event: PeerlistComplete EventList: Complete ListItems: 1 Response: Success Message: Peer status list will follow Event: PeerEntry Channeltype: IAX2 ChanObjectType: peer ObjectName: 1001/1001 IPaddress: -none- IPport: 0 Dynamic: yes...

Hello list, I have in sip.conf : /maxexpiry=60 ; Maximum allowed time of incoming registrations ; and subscriptions (seconds) minexpiry=60 ; Minimum length of registrations/subscriptions (default 60) defaultexpiry=120 ; Default length of incoming/outgoing registration ;-----------------------------------------

...out your participation. Thank you! The following are the issues resolved in this release: Bugs fixed in this release: ----------------------------------- * ASTERISK-22790 - check_modem_rate() may return incorrect rate for V.27 (Reported by Paolo Compagnini) * ASTERISK-23061 - [Patch] 'textsupport' setting not mentioned in sip.conf.sample (Reported by Eugene) * ASTERISK-23028 - [patch] Asterisk man pages contains unquoted minus signs (Reported by Jeremy Lain??) * ASTERISK-23046 - Custom CDR fields set during a GoSUB called from app_queue are not inserted (Reported by...

...out your participation. Thank you! The following are the issues resolved in this release: Bugs fixed in this release: ----------------------------------- * ASTERISK-22790 - check_modem_rate() may return incorrect rate for V.27 (Reported by Paolo Compagnini) * ASTERISK-23061 - [Patch] 'textsupport' setting not mentioned in sip.conf.sample (Reported by Eugene) * ASTERISK-23028 - [patch] Asterisk man pages contains unquoted minus signs (Reported by Jeremy Lain??) * ASTERISK-23046 - Custom CDR fields set during a GoSUB called from app_queue are not inserted (Reported by...

HI ALL got small question i use call-limit=1 on peers but call limit is not working if user is not registered on PBX and making calls so the main question is -- how to Disallow CALLS without registering on PBX -- Best regards Antony tel. +380669197533 tel2. +380636564340 Paypal http://paypal.me/Satskiy

...at_cap allocation (Reported by Corey Farrell) * ASTERISK-23062 - res_pjsip AOR config option qualify_frequency is inconsistently respected (Reported by Rusty Newton) * ASTERISK-23071 - pjsip: mailboxes documentation is lacking (Reported by Matt Jordan) * ASTERISK-23061 - [Patch] 'textsupport' setting not mentioned in sip.conf.sample (Reported by Eugene) * ASTERISK-23028 - [patch] Asterisk man pages contains unquoted minus signs (Reported by Jeremy Lain??) * ASTERISK-23046 - Custom CDR fields set during a GoSUB called from app_queue are not inserted (Reported by...

...at_cap allocation (Reported by Corey Farrell) * ASTERISK-23062 - res_pjsip AOR config option qualify_frequency is inconsistently respected (Reported by Rusty Newton) * ASTERISK-23071 - pjsip: mailboxes documentation is lacking (Reported by Matt Jordan) * ASTERISK-23061 - [Patch] 'textsupport' setting not mentioned in sip.conf.sample (Reported by Eugene) * ASTERISK-23028 - [patch] Asterisk man pages contains unquoted minus signs (Reported by Jeremy Lain??) * ASTERISK-23046 - Custom CDR fields set during a GoSUB called from app_queue are not inserted (Reported by...

...------------------------- * ASTERISK-22790 - check_modem_rate() may return incorrect rate for V.27 (Reported by Paolo Compagnini) * ASTERISK-23034 - [patch] manager Originate doesn't abort on failed format_cap allocation (Reported by Corey Farrell) * ASTERISK-23061 - [Patch] 'textsupport' setting not mentioned in sip.conf.sample (Reported by Eugene) * ASTERISK-23028 - [patch] Asterisk man pages contains unquoted minus signs (Reported by Jeremy Lain??) * ASTERISK-23046 - Custom CDR fields set during a GoSUB called from app_queue are not inserted (Reported by...

...------------------------- * ASTERISK-22790 - check_modem_rate() may return incorrect rate for V.27 (Reported by Paolo Compagnini) * ASTERISK-23034 - [patch] manager Originate doesn't abort on failed format_cap allocation (Reported by Corey Farrell) * ASTERISK-23061 - [Patch] 'textsupport' setting not mentioned in sip.conf.sample (Reported by Eugene) * ASTERISK-23028 - [patch] Asterisk man pages contains unquoted minus signs (Reported by Jeremy Lain??) * ASTERISK-23046 - Custom CDR fields set during a GoSUB called from app_queue are not inserted (Reported by...

...sion Asterisk 1.8.3.3-1digium1~squeeze built by pbuilder @ nighthawk on a x86_64 running Linux on 2011-04-22 17:50:44 UTC *CLI> sip show settings Global Settings: ---------------- UDP Bindaddress: 0.0.0.0:5060 TCP SIP Bindaddress: 0.0.0.0:5060 TLS SIP Bindaddress: 0.0.0.0:5061 Videosupport: No Textsupport: No Ignore SDP sess. ver.: No AutoCreate Peer: No Match Auth Username: No Allow unknown access: No Allow subscriptions: Yes Allow overlap dialing: Yes Allow promsic. redir: No Enable call counters: No SIP domain support: Yes Realm. auth: No Our auth realm pbx.domain.com Use domains as realms: No Ca...

CentOS-6.5 (FreePBX-2.6) Asterisk-11.14.2 (FreePBX) snom870-SIP 8.7.3.25.5 I am having a very difficult time attempting to get TLS and SRTP working with Asterisk and anything else. At the moment I am trying to get TLS functioning with our Snom870 desk-sets. And I am not having much luck. Since this is an extraordinarily (to me) Byzantine environemnt I am going to ask if any of you have gotten

...managed to get a headache... I have tried following best practices, worst practices, and still nothing works. My sip.conf looks like this: [general] context = default bindport = 5060 bindaddr = 0.0.0.0 srvlookup = yes allow = all ;allow = t140red textsupport = yes videosupport = yes ;allow = h263 maxcallbitrate = 384 register => sip-in?<phone number>:<secret>@sip.internode.on.net/<phone number> externip = <my static ip> localnet = <my local subnet> canreinvite = no hasvoice...