search for: useclientcode

...p= X.Y.Z.250 [general] port = 5060 bindaddr = 0.0.0.0 context = default allowoverlap=no srvlookup = yes : externip = externip = x.y.z.250 localnet=10.202.17.0/255.255.255.0 qualify=yes nat=yes register = xxxxxxx:SipServer/xxxxxxxx limitonpeers=yes allowsubscribe=yes notifyringing=yes notifyhold=yes useclientcode=yes canreinvite=no I have pfsense setup to forward ports 5060 and RTP ports over UDP back to the internal asterisk server. And a firewall rule to allow this traffic from only my ITSP SipServer. I can make a call from any phone on the local phones network to the outside world via the SIP proxy wit...

...NULL directmedia: NULL nat: force_rport,comedia callgroup: NULL pickupgroup: NULL language: NULL disallow: NULL allow: NULL insecure: NULL trustrpid: NULL progressinband: NULL promiscredir: NULL useclientcode: NULL accountcode: NULL setvar: NULL callerid: NULL amaflags: NULL callcounter: NULL busylevel: NULL allowoverlap: NULL allowsubscribe: NULL videosupport: NULL maxcallbitrate: NULL rfc2833compensate: NULL mailbox...

...n a big change in sip.conf or codec handling which cause this problem, cause i used the same sip.conf just adding notifyringing=yes, limitonpeers=yes and allowsubscribe=yes. Here is my sip.conf with one client: [general] 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 subscribeconte...

...* The 'X-ClientCode' header within a SIP INFO message when using chan_sip and the 'useclientcode' option is enabled (note, it's disabled by default). * The CDR dialplan function executed from AMI when setting...

...a specific context for SUBSCRIBE requests notifyringing = yes ; Notify subscriptions on RINGING state (default: no) notifyhold = yes ; Notify subscriptions on HOLD state (default: no) limitonpeers = yes ; Apply call limits on peers only. This will improve useclientcode=yes When more information is needed, please ask..............

...* The 'X-ClientCode' header within a SIP INFO message when using chan_sip and the 'useclientcode' option is enabled (note, it's disabled by default). * The CDR dialplan function executed from AMI when setting...

...m status: sip.conf: [1050] ; THAT'S A SOFTPHONE type=friend host=dynamic callerid=Softphone 1050 secret=xxxx context=call-center disallow=all allow=alaw allow=ulaw dtmfmode=rfc2833 canreinvite=yes nat=no qualify=yes call-limit=1 allowtransfer=yes insecure=no promiscredir=no useclientcode=no videosupport=no [7600] ; THAT'S PSTN CONNECTION username=7600 type=friend secret=xxxx qualify=no port=5060 nat=yes mailbox=7600 at default host=dynamic dtmfmode=rfc2833 context=out canreinvite=no callerid=7600 disallow=all allow=g729 [sipgvt] ; THAT'S PSTN CONNECTION...

...but by pressing the blinking light I could pickup the call. This is a part of my sip.conf for a snom phone: [404] type=friend context=sip callerid=?Stephan Seitz? <404> username=404 secret=<secret> host=dynamic defaultip=10.10.30.103 canreinvite=no mailbox=404 vmexten=1404 call-limit=1 useclientcode=yes subscribecontext = sip notifyringing = yes notifyhold=yes dtmfmode = rfc2833 Any hints how to solve the problem are welcome. Shade and sweet water! Stephan -- | Stephan Seitz E-Mail: Nur-Ab-Sal at gmx.de | | PGP Public Keys: http://fsing.rootsland.net/~stse/pgp.html | -...

You are reading the instructions for the STABLE 1.0 version of asterisk and are using the CVS version. Goto the wiki and read the instructions for RealTime. -Matthew ----- Original Message ----- From: "Muhammad Rizwan Khan" <rizwan@advcomm.net> To: <Asterisk-Dev@lists.digium.com> Sent: Wednesday, January 05, 2005 12:42 PM Subject: [Asterisk-Dev] Asterisk with MySQL >