search for: rtupdat

Hi all, Is there any native way to save endpoint statuses to database? I use asterisk 13 with pjsip and realtime, and didn't found proper way. I read that there is config parameter in sip.conf: rtupdate=yes. But how can I do that with pjsip? Or I should use sip.conf with pjsip simultaneously. Or is there any kind of hooks, which allows make custom action on endpoint status change. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/p...

...an see packets between Asterisk and both sides (SPA112 and PSTN fax) but it seems that faxes can't agree how to send image. == sip.conf: [general] tcpenable=yes videosupport=yes transport=udp,tcp dtmfmode=rfc2833 qualify=yes directmedia=no allowguest=no alwaysauthreject=yes rtcachefriends=yes rtupdate=no callcounter=yes t38pt_udptl=yes,redundancy,maxdatagram=200 t38pt_rtp=no t38pt_tcp=no ignoresdpversion=yes disallow=all allow=alaw allow=ulaw externip=82.200.7.184 localnet=192.168.0.0/255.255.0.0 [mtt] type=peer host=80.75.130.136 fromuser=74957777777 disallow=all allow=alaw,ulaw directmedia=n...

...it will register. but after registration expires and its time to re-register the same thing will happen, so i have to update the port settings again just to make it work which is troublesome. i'm using Asterisk 1.4.31 with the following realtime config: rtcachefriends=yes rtsavesysname=yes rtupdate=yes rtautoclear=no one thing i noticed is that it only seems to happen on linksys devices e.g. PAP2 and SPA's. another phone i'm using is yealink and so far no client has complain about it. hope anyone can help. thank you. regards Ron

Hello Asterisk community, I'm trying to use BLF with Asterisk Realtime, i've been searching for some info but nothing seems to be clear, can anyone help me eith some ideas to make this work ok? I'va my dialplan with Realtime Thanks in advance -- Saludos Danny Dias SkypeID: danny.dias1

...disallow=all ; First disallow all codecs allow=g729 allow=gsm allow=alaw ; Allow codecs in order of preference dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833 rtcachefriends=yes realm=vtxvoip useragent=VTX SIP rtupdate=yes language=en tos=184 notifyringing=yes t38pt_udptl=yes I have the definition of the phone in DB. voip-test-01*CLI> sip show peer 0625037998 voip-test-01*CLI> * Name : 0625037998 Realtime peer: No Secret : <Set> MD5Secret : <Not set> C...

...bind to (SIP standard port is 5060) tos_sip=cs3 tos_audio=ef ; bindport is the local UDP port that Asterisk will ; listen on bindaddr=192.168.xx.xx ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls limitonpeers=yes notifyringing=yes rtupdate=yes[authentication] [104] type=peer context=phones host=dynamic fromuser=104 secret=xxxxxx canreinvite=update directrtpsetup=no call-limit=3 nat=yes qualify=yes register=no session-timers=accept session-expires=90 session-minse=120 session-refresher=uac register => 104:xxxxx...

...register on that one. So far I considered the following for this project: - Moving all SIP extensions from individual sip.confs to one MySQL database, and point all servers to that one - Configure sip.conf on each machine like this: regcontext=dundi-internal rtcachefriends=yes rtsavesysname=yes rtupdate=no rtautoclear=yes ignoreregexpire=no That way each time an extension registers, Asterisk would add an extension to the dundi-internal context, which as you guessed, is the one being mapped to the other servers. So instead of mapping extensions using wildcards, the extensions will be mapped indiv...

Hi, list. We have a problem with loss peers after 'sip reload', our configuration: Asterisk 11.6-cert1, SIP realtime peers, sip.conf: - rtcachefriends=yes - rtsavesysname=yes - rtupdate=yes - rtautoclear=yes When we do 'sip reload' , peers are removing from available. Before `sip reload` : srv-pbx2*CLI> sip show peers Name/username Host Dyn Forcerport ACL Port Status Description Realtime 303411/303411...

...9;m not sure if this is relevant but I checked that Asterisk was successfully compiled with res_srtp module. Here's my sip.conf contents: bindport = 5070 ; using this since Kamailio is at 5060 bindaddr = PU.BL.IC.IP tcpenable = yes ;no limitonpeers = yes rtcachefriends = yes ; for realtime rtupdate=yes tos_sip=cs3 tos_audio=ef useragent=MyAsterisk realm = myrealm.com autodomain=no domain=PU.BL.IC.IP domain=testers.com allowexternaldomains=no allowguest=no avpf=yes encryption=yes transport=ws,udp icesupport=yes srvlookup=yes And here's an example of a ws client in my realtime peer ta...

Hi, Is there something wrong with REALTIME (ARA) when used with rtcachefriends parameter? In my sip.conf (Asterisk 1.2.0): rtcachefriends=yes rtupdate=yes rtautoclear=yes Desired configuration is realtime configuration (via odbc) for SIP phones + MWI. Realtime means the following: when I make changes to db they should apply with no extra commands executed in CLI. In order to use MWI with REALTIME it is necessary to use rtcachefriends parameter...

...mode, etc). However, the IP address field is coming back as "unspecified". Now, naturally I ensured server time was acurate with NTP for starters. This was to avoid the registration appearing as EXPIRED on ServerB. Sip.conf snippet (same on ServerA and ServerB) ... rtcachefriends=yes rtupdate=yes rtautoclear=yes rtignoreexpire=yes So, as you can all see it's most importantly caching the peer and updating MySQL. And yes if you do a 'realtime load sipusers name PEERNAME', it shows the ipaddr field with the correct value. Is there something obvious I am missing? I googled th...

Anyone know the command that tells * to load a sipfriend from the realtime db rather than saying no such host? I've tried various combinations of the rt commands: rtcachefriends=yes; ;rtcache=yes ;rtAutoClear=yes ;rtautoreg=yes ;rtIgnoreRegExpire=yes ;rtupdate=yes rtfromcontact=yes Basically I have a group of 4 * servers all routing calls, but only two are hearing the phones registration. I'd like the other two to load the sipfriends entry from mysql when a channel for that sipfriends is requested. Any ideas? -------------- next part -...

Hi all, Is there a way to keep track in Asterisk of which phones are online in realtime using some MySQL DB table for exemple, much like "sip show peers" does in the CLI? Regards, Ricardo.

...context=default allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes domain=srvcentral.meudominio.com.br tos_sip=cs3 tos_audio=ef tos_video=af41 language=pt_BR rtptimeout=60 rtpholdtimeout=300 notifyringing = no notifyhold = no limitonpeers = yes nat=yes rtcachefriends=yes rtsavesysname=yes rtupdate=yes Registration is working fine, the only problem I can see is qualify. Anybody can help me ? Marcelo H. Terres mhterres at gmail.com **************************************** ICQ: 6649932 MSN: mhterres at hotmail.com Jabber: mhterres at jabber.org http://twitter.com/mhterres http://mhterres.j...

...context=incoming realm=softpbx bindport=5060 bindaddr=0.0.0.0 srvlookup=yes useclientcode=yes defaultexpirey=3600 vmexten=voicemail disallow=all allow=alaw allow=ulaw allow=gsm ;qualify=no ;canreinvite=no musicclass=default language=de useragent=ipanlage callevents=yes nat=yes rtcachefriends=no rtupdate=no rtautoclear=no ignoreregexpire=yes amaflags=omit canreinvite=no subscribecontext=outcust limitonpeers=yes allowsubscribe=yes notifyringing=yes [xxx] type=friend context=outcust nat=yes qualify=yes secret=yyyy username=xxx callerid="bla bla" accountcode=xxx disallow=all allow=alaw all...

...at=yes pickupgroup= callgroup= qualify=2000 secret=cyx2mo type=friend username=8250 subscribecontext=pbx9 call-limit=100 disallow=all allow=g729 allow=g723 allow=ulaw allow=alaw allow=ilbc allow=gsm allow=h263 allow=h263p allow=h264 ***sip.conf*** [general] . . . . notifyringing=yes notifyhold=no rtupdate=yes rtcachefriends=yes ***extensions.conf*** [pbx9] exten => 8340,hint,SIP/8340 include => pruebas switch => Realtime/pbx9 at extensions In the Asterisk CLI i could see this message: [Jul 15 18:43:14] WARNING[15867]: chan_sip.c:15766 handle_request_subscribe: SUBSCRIBE failure: unreco...

...at=yes pickupgroup= callgroup= qualify=2000 secret=cyx2mo type=friend username=8250 subscribecontext=pbx9 call-limit=100 disallow=all allow=g729 allow=g723 allow=ulaw allow=alaw allow=ilbc allow=gsm allow=h263 allow=h263p allow=h264 ***sip.conf*** [general] . . . . notifyringing=yes notifyhold=no rtupdate=yes rtcachefriends=yes ***extensions.conf*** [pbx9] exten => 8340,hint,SIP/8340 include => pruebas switch => Realtime/pbx9 at extensions In the Asterisk CLI i could see this message: [Jul 15 18:43:14] WARNING[15867]: chan_sip.c:15766 handle_request_subscribe: SUBSCRIBE failure: unreco...

...in the clients to a very small value to avoid becoming 'UNREACHABLE' for a long time, but his doesn't seem to be the solution. Is there any specific SIP settings which needs to be made in 1.6 to avoid this problem? I am using realtime sip. Some of my sip settings are rtcachefriends=yes rtupdate=no qualify=yes canreinvite=yes nat=yes Thanks

...------------------------<br> ; For additional information on ARA, the Asterisk Realtime Architecture,<br> ; please read realtime.txt and extconfig.txt in the /doc directory of the<br> ; source code.<br> ;<br> rtcachefriends=yes <br> ;rtsavesysname=yes<br> ;rtupdate=yes<br> ;rtautoclear=yes<br> ;ignoreregexpire=yes<br> <br> <br> <br> Kind regards,<br> <br> Jonas.<br> </font></font> </body> </html>

...owdelay maxexpirey=3600 defaultexpirey=1000 allow=all musicclass=default language=fr insecure=very allowguest=yes rtptimeout=60 rtpholdtimeout=300 useragent=PBX dtmfmode = rfc2833 checkmwi=20 promiscredir=no nat=yes autodomain=no domain=nxs.yi.org,sip allowexternalinvites=yes rtcachefriends=yes rtupdate=yes rtautoclear=yes ignoreregexpire=yes and extensions.conf [general] static=yes writeprotect=no autofallthrough=yes ////////////////////////////////////////////////////// [globals] [mainmenu] exten => s,1,Answer() exten => s,n,GotoIfTime(09:30-21:00|mon-sun|*|*?day,s,1) exten => s,n...