search for: grandstream1

...; Console interface for demo IAXINFO=guest ; IAXtel username/password TRUNK=Zap/1 ; Trunk interface TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0) [default] include => incoming [incoming] exten => s,1,Answer() exten => s,2,NoOp(${CALLERID}) exten => s,3,Dial(SIP/Grandstream1) exten => 123,1,Answer exten => 123,2,Dial(SIP/Grandstream1) exten => 321,1,Answer exten => 321,2,Dial(Zap/1) ~~~~~~~~~~~~~~~~~~~~~~~~~~~~ When I use the Analog phone to dial "123" The Grandstream1 rings and answers and works fine. But, when I pickup the Gr...

Using svn trunk, I was trying to see what the astdb entry in the sip.conf file does. Nothing :) I presume that it's meant to create an entry in the astdb. so, I have astdb=chan2ext/SIP/grandstream1=1234 in sip.conf But database show only gives *CLI> database show /SIP/Registry/706 : 192.168.0.200:5060:3600:706:sip:706@192.168.0.200:5060 /SIP/Registry/731 : 192.168.6.156:5060:3600:731:sip:731@192.168.6.156:5060 /dundi/secr...

...romuser=5083021425 secret=password-for-2nd-BV-account dtmfmode=inband context=sip canreinvite=no insecure=very and here's the output from sip show peers and sip show registry *CLI> sip show peers Name/username Host Dyn Nat ACL Mask Port Status grandstream1/grandstream1 (Unspecified) D 255.255.255.255 0 Unmonitored phone2/phone2 (Unspecified) D 255.255.255.255 0 Unmonitored phone1/phone1 192.168.1.108 D 255.255.255.255 5060 Unmonitored simpleconnect-sip/wlee179...

...; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls disallow=all ; Disallow all codecs allow=g729 [cisco] context=in type=friend insecure=yes host=<removed> dtmfmode=rfc2833 [grandstream1] type=friend secret=grandstream1 host=dynamic context=class1 dtmfmode=rfc2833 [grandstream2] type=friend secret=grandstream2 nat=yes host=dynamic context=class1 dtmfmode=rfc2833 Asterisk ver: Asterisk CVS-01/22/04-18:13:23 Grandstream ver: Program--1.0.3.81 Bootloader--1.0.0.7 HTML--1.0.0....

...to bind to ;externip =3D 200.201.202.203 ; Address that we're going to put in = SIP messages if we're behind a NAT tos=3Dlowdelay disallow=3Dall ; Disallow all codecs allow=3Dulaw ; Allow codecs in order of preference dtmfmode=3Dinfo [grandstream1] type=3Dfriend host=3Ddynamic secret=3Dmysecret context=3Doutgoing nat=3Dyes reinvite=3Dno canreinvite=3Dno qualify=3D2000 has anyone done this before? chandra -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-us...

...coming calls only. Example: FWD (Free World Dialup) ;type=user ;context=from-fwd ;[sip_proxy-out] ;type=peer ; we only want to call out, not be called ;secret=guessit ;username=yourusername ;fromuser=yourusername ; Many SIP providers require this! ;host=box.provider.com ;[grandstream1] ;type=friend ; either "friend" (peer+user), "peer" or "user" ;context=from-sip ;username=grandstream1 ; usually matches the [section] title ;fromuser=grandstream1 ; overrides the callerid, e.g. required by FWD ;callerid=John Doe <1...

...=user ;context=from-fwd ;[sip_proxy-out] ;type=peer ; we only want to call out, not be called ;secret=guessit ;username=yourusername ; Authentication user for outbound proxies ;fromuser=yourusername ; Many SIP providers require this! ;host=box.provider.com ;[grandstream1] ;type=friend ; either "friend" (peer+user), "peer" or "user" ;context=from-sip ;fromuser=grandstream1 ; overrides the callerid, e.g. required by FWD ;callerid=John Doe <1234> ;host=192.168.0.23 ; we have a static but pr...

...=user ;context=from-fwd ;[sip_proxy-out] ;type=peer ; we only want to call out, not be called ;secret=guessit ;username=yourusername ; Authentication user for outbound proxies ;fromuser=yourusername ; Many SIP providers require this! ;host=box.provider.com ;[grandstream1] ;type=friend ; either "friend" (peer+user), "peer" or "user" ;context=from-sip ;fromuser=grandstream1 ; overrides the callerid, e.g. required by FWD ;callerid=John Doe <1234> ;host=192.168.0.23 ; we have a static but priv...

I am trying to setup a couple of virtual pbx's off of my one may asterisk box. So far I have been able to segment most everything via the Dial plan. My only question/problem has to do with the # Transfer function. I had set up # Transfers prior to segmenting the dial plan, and I cannot remember how I was able to specify which context to use when the user presses #. I haven't been able

...; ;type=user > ;context=from-fwd > > ;[sip_proxy-out] > ;type=peer ; we only want to call out, not be called > ;secret=guessit > ;username=yourusername > ;fromuser=yourusername ; Many SIP providers require this! > ;host=box.provider.com > > ;[grandstream1] > ;type=friend ; either "friend" (peer+user), "peer" or "user" > ;context=from-sip > ;username=grandstream1 ; usually matches the [section] title > ;fromuser=grandstream1 ; overrides the callerid, e.g. required by FWD >...

...end username=2010 secret=1945 nat=1 host=dynamic dtmfmode=rfc2833 canreinvite=yes reinvite=yes qualify=200 allow=all [2011] ; ip-phone no brand type=friend username=2011 secret=1945 nat=1 host=dynamic dtmfmode=rfc2833 canreinvite=yes reinvite=yes qualify=yes allow=all [2012] ;grandstream1 type=friend username=2012 secret=1945 nat=1 host=dynamic dtmfmode=rfc2833 canreinvite=yes reinvite=yes qualify=yes allow=all ***************************** and with this extensions.conf file: [general] static=yes writeprotect=yes autofallthrough=yes [globals] CONSOLE=Console/dsp...

...of the slave, but iax.conf on the slave indicates to register. So here we sit. I pick up the SIP phone and dial "22" and on the slave (to which the SIP phone is connected) I get: ---------------------------------------------------------------- *CLI> -- Executing Dial("SIP/grandstream1-c62b", "IAX2/asterisk:lilbuddy@192.168.0.250/22@internal") in new stack -- Called asterisk:lilbuddy@192.168.0.250/22@internal Aug 27 11:40:30 WARNING[131080]: chan_iax2.c:5352 socket_read: Call rejected by 192.168.0.250: No authority found -- Hungup 'IAX2/192.168.0.250:45...

I've just installed an X100P, built the kernel module, and tried to use it to make an outgoing call (via a phone connected to an ATA-186). However, I just get a reorder tone and see this on the console: -- Executing Dial("SIP/ata1-4409", "Zap/1/5551212") in new stack NOTICE[1200825920]: File app_dial.c, Line 481 (dial_exec): Unable to create channel of type