search for: 8888

...ve set a test extension to get the input dtmf key and say the digit out. They are getting correctly on the IVR but when using n-way conferencing, it's not taking it. Here is the output of testing DTMF with IVR. v103*CLI> v103*CLI> -- Executing [88 at from-internal:1] Read("SIP/8888-b6807538", "digito||10") in new stack -- Accepting a maximum of 10 digits. * DTMF-relay event received: 8 * DTMF-relay event received: 5 * DTMF-relay event received: 2 -- User entered '852' -- Executing [88 at from-internal:2] SayDigits("SIP/8888-b6807538&quo...

...downloaded from here: http://downloads.asterisk.org/pub/telephony/sounds/ ============ find /var/lib/asterisk/sounds/ -name "manolo_camp-mor*" /var/lib/asterisk/sounds/fr/manolo_camp-morning_coffee.ulaw ============ extensions.conf ;just in case, but shouldn't be needed exten => 8888,1,Set(CHANNEL(language)=fr) exten => 8888,n,Wait(2) exten => 8888,n,Answer() exten => 8888,n,Playback(manolo_camp-morning_coffee) exten => 8888,n,Hangup ============ cat asterisk.conf ... [options] nocolor = yes ; Disable console colors languageprefix = yes ; Use the new sound prefix p...

Dear, I need to create a custom device extension in order to dial a local queue. Suppose my queue number is 8888, how can fill the Dial field from the custom extension ??? Because if I put just 8888 or Local/8888, I don't succeed. Thanks a lot

...istener and connector, and starting with an empty firewall with everything set to ACCEPT: $ id -u 1000 Test 1a - Works on both 3.2 and 3.10 ------------------------------------ Verify connection without any firewall rules. Start a listener on port 8889 and start a connection from source port 8888. Connection established and data flows. $ nc -l -p 8889 | nc -p 8888 127.0.0.1 8889 | I can send data! I can send data! | | ^C Test 1b - Works on both 3.2 and 3.10 ------------------------------------ Verify connection without...

...he landline is back to idle after a call, but for some reason, Asterisk doesn't detect that the callee has hung up after listening to MoH for a few seconds: ========== extensions.conf ;Play MoH for a few seconds, hang up, and ;check ChanIsAvail() able to detect when line idle again exten => 8888,1,Answer() exten => 8888,n,Playback(/var/tmp/manolo_camp-morning_coffee) exten => 8888,n,Hangup ========== CLI ;keep line engaged for a few seconds, and hang up from remote end originate Zap/1/5551234 extension 8888 at internal ========== extensions.conf ;call from XLite to check line stat...

Good day, I have tried to forward port 8888 on the firewall to port 80 on an internal IP of 10.40.0.202. Please advise which rule to add to the rules file to achieve this. I have tried to add DNAT net loc:10.40.0.202:80 tcp 8888 But still no luck TIA for you assistance Quentin

can someone help me with an .htaccess rewrite? i''m trying to convert all requests to "/?spot=8888" to be "/locations/ 888". thanks. --~--~---------~--~----~------------~-------~--~----~ You received this message because you are subscribed to the Google Groups "Ruby on Rails: Talk" group. To post to this group, send email to rubyonrails-talk-/JYPxA39Uh5TLH3MbocFFw@publi...

Hello I''m trying to setup tinyproxy and shorewall on a LEAF Bering firewall. What I''d like to do is block all HTTP connections to the internet on port 80 and 8080 and force users to use port 8888. So in shorewall/rules I have ACCEPT loc fw tcp 8888 DROP loc fw tcp 80,8080 The ACCEPT works fine but the DROP does not seem to work. If I don''t specify a port in a web browser, I can still surf. Any clues as to what I can try to figure it out? T...

...to keep (any word, letter, or number). However, after strsplit and unlist, I know how to select specific elements with "[ ]", but I want to select instead all elements that are not missing. That is, do not select any elements that are simply "". output <- c("a b 8888 6") gsub(" ","",unlist(strsplit(output," ",fixed=TRUE))) [1] "a" "" "b" "8888" "" "" "6" Thanks! Scott Chamberlain [[alternative HTML version deleted]]

Hi, I'm having some trouble trying to run tinc over GNU httptunnel. On server A I have: /etc/tinc/netname/tinc-up: #!/bin/sh ifconfig $INTERFACE 10.3.0.1 netmask 255.255.0.0 /usr/bin/hts -F localhost:655 8888 /etc/tinc/netname/hosts/A: Address = foo.com Subnet = 10.3.0.0/24 TCPonly = yes /etc/tinc/netname/hosts/B: ConnectTo = A Subnet = 10.4.0.0/24 TCPonly = yes /etc/tinc/netname/tinc.conf: Name = A Device = /dev/net/tun Hostnames = no PrivateKeyFile = /etc/tinc/apa/rsa_key.priv This setup worked fi...

...SIP configuration access to the phone. i.e. let say I want user 132 (that uses extension 132 in our system) to be able to lock his phone (located in a publicly accessible office). Could he dial an special extension (i.e. 9999) and Asterisk will drop any call until another special extension (i.e. 8888) is dialed? Suggestions? - -- Andres Jimenez -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.3 (GNU/Linux) Comment: http://firegpg.tuxfamily.org iD8DBQFGxX6A8SZxpGYWwpYRAtojAJ4yKE77nv9rpkoXXr1i4SOiLPb7JACgug+7 64yg8fCDRdnmeZFmmpGynwQ= =eCWP -----END PGP SIGNATURE-----

...w peers. But, my local dial rules does not work. I can call from realtime user to static users(the ones in users.conf) and if they are not available voicemail activates etc. But when i call a realtime user which is already on peer list i got chan_sip.c:20152 handle_request_invite: Call from '8888' to extension '8888' rejected because extension not found in context 'DLPN_WorldcallDial'. And this is when i call a static user (works normal) Executing [6000 at DLPN_WorldcallDial:1] Macro("SIP/8888-0000001e", "stdexten,6000,SIP/6000") in new stack T...

...ay public ip 132.1.1.3. It is connected on tun1 with server 1 - IP 10.2.0.1 (on server3) an IP 10.2.0.6 (on server 1) I change the ip rule to ip rule from 123.1.1.1 fwmark 0x1 lookup vpn_to_server3 Exactly the same settings as above with default gateway on tun1. I connect with with plink.exe -D 8888 me at server1.com on windows xp. I can connect and surf with the ip of server 3. Fine! Setup works! But ... I connect from arch (ssh -v => OpenSSH_6.0p1, OpenSSL 1.0.1a 19 Apr 2012) to server 1 with ssh -vvv -D 8888 me at server1.com and when I try to surf a page, I get these errors: debug1: C...

Just deployed samba print server. Print jobs are going thru but getting: Mar 11 08:11:53 printsrv smbd[2017]: [2004/03/11 08:11:53, 0] smbd/connection.c: register_message_flags(220) Mar 11 08:11:53 printsrv smbd[2017]: register_message_flags: tdb_fetch failed Mail list archive thread said to use tdbbackup. What good will that do? What should I look for in tdbbackup -v? Should I delete

...er - PoE (power over ethernet), so I can power it off the central switch - cheap enough to rip apart Any ideas? Also, is there anyway in the Asterisk config, to ensure anything dialed from the phone in turn just dials a specific extension? What I'm thinking: Door bell button rings extension 8888, which rings all phones. Caller ID is "Front Door Entry" or whatever, and the Cisco's Dial Plan has a doorbell chime as the ring tone for this specific extension filter (so 8888 has it's own ring tone). There is a time out of 5 to 10 seconds, which represents the time it takes...

...isk goes crazy and continues to loop forever. Please have a look and tell me where I went so wrong. Func_odbc.conf looks like this: [PIN] dsn=BRIDGE mode=multirow readsql=SELECT pin from users WHERE confid='${SQL_ESC(${CONF_ID})}' extensions.conf section: [infromhost] ;Host dials 8888 over SIP trunk exten=8888,1,Answer exten=8888,n,Background(conf-getconfno) exten=8888,n,WaitExten(10) exten=8888,n,Hangup exten=_XXXXXX,1,Set(GLOBAL(CONF_ID)=${EXTEN}) exten=_XXXXXX,n,GoTo(rooms,${EXTEN},1) ; [rooms] exten=_XXXXXX,1,Set(CONF_ID=${EXTEN}) exten=_XXXXXX,n,Background(conf-get...

...IP address in NameVirtualHost and worker.ajp13Worker.host directives. Shall I provide DNS name? Thankx again. Vaneet Vaneet Sharma executive manager iDeasTank Limited an iwg business dolphins' court po 388 valletta, m-malta/europe mobile: +356 9943 8263 skype: CALLVANEET fax: +356 9952 8888 phone: +356 9942 8888 vaneet at iwg.info call me on www.skype.com - my ID is CALLVANEET Want a signature like this? - www.plaxo.com\signature iwg is a global e-mobile company creating, building and growing new businesses. iwg founders are pioneers in creating multi-billion dollar mobile and In...

Just a reminder to keep testing. The latest builds should be rock solid. The url is: http://i.cantcode.com:8888/stats.xml You can now get stats ouptut by pointing a browser to: http://i.cantcode.com:8888/stats.xml We're looking for more win32 freeamp reports, and any other clients besdies the standard ones (sonique, winamp, xmms, etc). Oh, and I stuck some silly station ids in between some tracks :)...

Just a reminder to keep testing. The latest builds should be rock solid. The url is: http://i.cantcode.com:8888/stats.xml You can now get stats ouptut by pointing a browser to: http://i.cantcode.com:8888/stats.xml We're looking for more win32 freeamp reports, and any other clients besdies the standard ones (sonique, winamp, xmms, etc). Oh, and I stuck some silly station ids in between some tracks :)...

Good Day all, I have a Fractional PRI connected to my Asterisk Box via a T100P card. When I initiate a call out to phone number 123-8888 the call sounds great no echo what so ever. If the person at 123-8888 hangs up and calls me right back (same handset on both sides) same trunk line The call always has echo on it. The Asterisk sip extension hears them selves echoing. The remote party does not notice any difference. I have tr...