search for: recordhistory

...context=default allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes domain=proxy.myhostname disallow=all allow=alaw sipdebug = yes recordhistory=yes dumphistory=yes register => <authstuff>@sip.externalpeer.com externhost=proxy.myhostname localnet=192.168.0.0/255.255.0.0 localnet=10.0.0.0/255.0.0.0 localnet=172.16.0.0/12 localnet=169.254.0.0/255.255.0.0 nat=never canreinvite...

...d route it according to what I've specified there, i.e Least Cost Routing, etc. I've set-up something similar to what I've found online, but it doesn't work! Asterisk doesn't pick up the call at all..... :( The files I used: sip.conf (for the DID) [general] context=default recordhistory=yes port=5060 bindaddr=0.0.0.0 srvlookup=yes tos=lowdelay maxexpirey=3600 defaultexpirey=120 allow=ulaw allow=alaw musicclass=default language=en relaxdtmf=yes rtptimeout=60 trustrpid = no progressinband=yes useragent=Asterisk PBX promiscredir = no [incoming] ; For incoming calls only. type=user u...

...nd a 404 NOT FOUND answer. I can't understand why asterisk doesn't found the users if they are registred... It's making a "Scheduling Call Destruction". My config files are : sip.conf : [general] >>context=default ; Default context for incoming calls >>recordhistory=yes ; Record SIP history by default >>port=5060 ; UDP Port to bind to (SIP standard port is 5060) >>bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) >>srvlookup=yes >> >>[1111] >>;Turn off silence suppression in X-Lite (&qu...

...larityswitch=yes callprogress=yes callwaiting=yes relaxdtmf=no progzone=uk useincomingcalleridonzaptransfer = yes usecallerid=no callerid=asreceived cidsignalling=v23 cidstart=polarity ukcallerid=yes channel => 4 # sip.conf [general] allow=ulaw allow=alaw allow=gsm allow=g723.1 context=incoming recordhistory=yes port=5060 bindaddr=0.0.0.0 srvlookup=yes tos=lowdelay defaultexpirey=120 nat=no localnet=192.168.0.0/255.255.252.0 [ronan] regextension=ronan regcontext=4L mailbox=100@default callerid=Ronan Mullally <100> restrictcid=no callgroup=1,2 pickupgroup=1,2 host=dynamic language=en type=friend...

...tory = no externip = 216.26.109.22 g726nonstandard = no jbenable = yes jbforce = no jblog = no localnet = internal subnet maxcallbitrate = 384 maxexpiry = 3600 minexpiry = 60 mohinterpret = default nat = yes notifyringing = yes pedantic = no progressinband = never promiscredir = no realm = asterisk recordhistory = no registerattempts = 0 registertimeout = 20 relaxdtmf = no sendrpid = no sipdebug = no t1min = 100 t38pt_udptl = no tos_audio = none tos_sip = none tos_video = none trustrpid = no useragent = Asterisk PBX usereqphone = no videosupport = no disallow = all allow = ulaw,gsm subscribecontext = devic...

...or congestion tone. it CANT be a password-problem as it only happens if a mobile gets called and rejects the call. What can I do to change this ? ------------------sip.conf----------------------- [general] disallow=all allow=ulaw allow=alaw context = from_sip defaultexpirey=160 tos=reliability recordhistory=yes realm=pbx.exse.net localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks localnet=10.0.0.0/255.0.0.0 ; Also RFC1918 register => XXXXXXXXXXXXXX@sipgate.de/1724173 register => XXXXXXXX@sip.broadvoice.com:YYYYYYYY:XXXXXXXXX@sip.broadvoice.com/XXXXXXXXX [out_sipgat...

hello, All AreskiCC users: I faced some problems in using AreskiCC. one is when I reload the asterisk server, the system display some errors such as execution 30 .. second one is there is no data display for admin added before. Does anyone know how to solve the problems, Please tell me! thanks in advance!

...et.net:3478) with no luck at all. The only option available on the market for me at the moment is Broadvoice because of an unlimited international plan with a flat rate, so I'm stuck here. Please, send me ideas. Thanks in advance, -Dhennys Pestana ### sip.conf [general] context=default recordhistory=yes realm=voip.server.com port=5060 bindaddr=0.0.0.0 srvlookup=yes tos=lowdelay maxexpirey=3600 defaultexpirey=900 videosupport=yes disallow=all allow=ulaw musicclass=default language=en useragent=Asterisk PBX canreinvite=yes nat=no [sipphone] type=peer username=6462050505 secret=secret host=sip.b...

...to happen is the call to go out through Asterisk - to SER (as SER knows where the SIP extension is) - and then onto the extension of the person to call. In my sip.conf I have the following: [general] context=sip-incoming ; Default context for incoming calls autocreatepeer=yes recordhistory=yes ; Record SIP history by default ; (see sip history / sip no history) ;realm=fedcore2.eicon.com ; Realm for digest authentication ; defaults to "asterisk" ; Realms MUST b...

...0 bindaddr=0.0.0.0 srvlookup=yes ;domain=mydomain.tld ;domain=mydomain.tld,mydomain-incoming ;domain=1.2.3.4 ;allowexternalinvites=no ;autodomain=yes ;pedantic=yes ;tos=184 ;tos=lowdelay ;maxexpiry=3600 ;defaultexpiry=120 ;notifymimetype=text/plain ;checkmwi=10 ;vmexten=voicemail ;videosupport=yes ;recordhistory=yes disallow=all allow=g729 allow=gsm allow=ulaw jitterbuffer=yes maxjitterbuffer=1500 ;allow=ilbc ;musicclass=default ;language=en ;relaxdtmf=yes rtptimeout=60 ;rtpholdtimeout=300 ;trustrpid = no ;sendrpid = yes ;progressinband=never ;useragent=Asterisk PBX ;promiscredir = no ;usereqphone = no dtm...

Can you also post information such as: Type of phone (model Number would be idela) How is it conencted, SIP, ZAP, IAX, Channel Bank. Corresponding config files would also help. Help us help you. >>-----Original Message----- >>From: asterisk-users-bounces@lists.digium.com >>[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of >>Paul A Brown >>Sent:

Hello again! I just got a SIP account and it seems - from a config on the net -, that I've configured it correctly. But I get no call to the outside. Registration was OK. I tried: channel originate sip/1/echo at iptel.org Application ... I see the channel active for a while, but no call gets established. In my config I have defined the section [iptel] for the outgoing call and I

...Voice, but it doesn't work, either. I've probably screwed my configs to hell trying to get this to work, but here they are. Any suggestions would be appreciated. Here are my configs, decrufted... sip.conf ------------------------------------------------------------ [general] context=sip recordhistory=yes port = 5060 bindaddr = 0.0.0.0 allow=gsm allow=alaw allow=ulaw allow=adpcm allow=speex allow=ilbc allow=slinear [general] nat=yes register => 2129999999:<password>:2129999999@147.135.8.128:5060 register => 2129999999:<password>:2129999999@147.135.0.128:5060 externip = 208.5...

...to happen is the call to go out through Asterisk - to SER (as SER knows where the SIP extension is) - and then onto the extension of the person to call. In my sip.conf I have the following: [general] context=sip-incoming ; Default context for incoming calls autocreatepeer=yes recordhistory=yes ; Record SIP history by default ; (see sip history / sip no history) ;realm=fedcore2.eicon.com ; Realm for digest authentication ; defaults to "asterisk" ; Realms MUST b...

Hi list, My need is to append a site specific parameter to the Contact: header on all INVITEs exiting * via a SIP trunk. I'd like it to look something like this: Contact: <bob:3125551212 at 10.10.10.10;SITE-ID=us.here> where SITE-ID=us.here is set in a config file that * parses on startup. Or in a Dial() command option? Or I don't care exactly how. :-) It is possible to

...go out through Asterisk - to SER (as > SER knows where the SIP extension is) - and then onto the extension of the > person to call. > > In my sip.conf I have the following: > [general] context=sip-incoming ; Default context for incoming calls autocreatepeer=yes recordhistory=yes ; Record SIP history by default ;realm=fedcore2.eicon.com ; Realm for digest authentication port=5061 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) srvlookup=no...

...e=text/plain ; Allow overriding of mime type in MWI NOTIFY > ;checkmwi=10 ; Default time between mailbox checks for peers > ;vmexten=voicemail ; dialplan extension to reach mailbox sets the > ;videosupport=yes ; Turn on support for SIP video > ;recordhistory=yes ; Record SIP history by default > disallow=all ; First disallow all codecs > allow=ulaw ; Allow codecs in order of preference > allow=gsm ; > musicclass=default ; Sets the default music on hold c...

...ck peers/users: ; sip show peers Show all SIP peers (including friends) ; sip show users Show all SIP users (including friends) ; sip show registry Show status of hosts we register with ; ; sip debug Show all SIP messages ; [general] context=sip ; Default context for incoming calls ;recordhistory=yes ; Record SIP history by default ; (see sip history / sip no history) realm=home.net ; Realm for digest authentication ; defaults to "asterisk" ; Realms MUST be globally unique according to RFC 3261 ; Set this to your host name or domain name port=5060 ; UDP Port...

...; dialplan extension to reach mailbox sets the ; Message-Account in the MWI notify message ; defaults to "asterisk" ;videosupport=yes ; Turn on support for SIP video ;recordhistory=yes ; Record SIP history by default ; (see sip history / sip no history) ;disallow=all ; First disallow all codecs ;allow=ulaw ; Allow codecs in order of preference ;allow=ilbc ; ;musicclass=defa...

Hi all, I registered a SIP account at budgetphone.nl/talkin2ya.nl Receiving calls works like a charm, I even redirected my normal PSTN number to the number I got from them so everything ends up in my * server. Before I ask them to take over my normal phone number I wanted to test all of it, so I ordered some calling minutes to test. Now I cannot get outbound calling to work with them. Anyone here