search for: mysecret

...fully it will work later, when Deutsche Telekom > > changes my ISDN to VoIP... > > Don't worry, Asterisk works very well with Deutsche Telekom and there > new ip-based connections ... That's a good news... Currenty I configured my sip.conf so: register => 00493511111111:MYSECRET at pbxluca/00493511111111 register => 00493512222222:MYSECRET at pbxfax/00493512222222 register => 00493513333333:MYSECRET at pbxanika/00493513333333 register => 4444444444:MYVERYSECRET at messagenet/4444444444 [pbxluca] type=peer defaultuser=00493511111111 secret= MYSECRET dtmfmode=rfc28...

Darryl Moore <darryl at moores.ca> schrieb: > Ahh. Seen that before! That suggests to me that you don't have your > sip.conf records setup right. > > What's your sip.conf look like? Well, here what I wrote in my sip.conf: register => 00493511111111:MYSECRET at pbxluca/00493511111111 register => 00493512222222:MYSECRET at pbxfax/00493512222222 register => 00493513333333:MYSECRET at pbxanika/00493513333333 register => 4444444444:MYSECRET at messagenet/4444444444 [pbxluca] type=peer defaultuser=00493511111111 secret= MYSECRET dtmfmode=rfc2833 h...

...libvirt supports encrypted snapshots,Here are my versions of libvirt and qemu [root@xx ~]# libvirtd -V libvirtd (libvirt) 4.5.0 [root@xx ~]# qemu-img -V qemu-img version 2.12.0 (qemu-kvm-ev-2.12.0-33.1.el7_7.4) Copyright (c) 2003-2017 Fabrice Bellard and the QEMU Project developers 1. assign $MYSECRET to libvirt secret using the secret-define and secret-set-value commands,and $MYSECRET is in base64 format MYSECRET=`printf %s "123456" | base64` 2. created a disk encrypted in luks format qemu-img create --object secret,id=sec0,data=$MYSECRET,format=base64 -f qcow2 -o encrypt.format=lu...

...lt;darryl at moores.ca> schrieb: > >> Ahh. Seen that before! That suggests to me that you don't have your >> sip.conf records setup right. >> >> What's your sip.conf look like? > Well, here what I wrote in my sip.conf: > > register => 00493511111111:MYSECRET at pbxluca/00493511111111 > register => 00493512222222:MYSECRET at pbxfax/00493512222222 > register => 00493513333333:MYSECRET at pbxanika/00493513333333 > register => 4444444444:MYSECRET at messagenet/4444444444 > > [pbxluca] > type=peer > defaultuser=00493511111111 &...

...on a small pbx. There is an IVR to select the extension. The DTMF tones are not being sensed so the IVR does not work and incoming calls are not being answered. I have listed my sip.conf entries. Is there any solution to this? ;Sipura units [101] type=friend host=dynamic context=default secret=mysecret mailbox=101 dtmfmode=inband disallow=all allow=ulaw [3200] type=friend host=dynamic context=pstn-in secret=mysecret qualify=yes dtmfmode=inband disallow=all allow=ulaw insecure=very [pstn-spa3k1] type=peer auth=md5 host=192.168.101.11 port=5061 secret=mysecret username=asterisk fromuser=...

...kCLI: == Using SIP RTP CoS mark 5 [May 29 19:42:13] NOTICE[2526]: chan_sip.c:20163 handle_request_invite: Call from '00493511111111' to extension '00493513333333' rejected because extension not found. users.conf on Ubuntu-PBX: [00493511111111] fullname = 00493511111111 secret = MYSECRET dahdichan = 1 hassip = yes hasiax = no hash323 = no hasmanager = no callwaiting = no context = default host = dynamic dtmfmode=rfc2833 canreinvite=no sendrpid=pai type=friend nat=force_rport,comedia qualify=yes qualifyfreq=60 transport=Auto avpf=no force_avp=no icesupport=no encryption=no callgroup...

Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb: > The phone you gave your wife is really old. Are you sure it supports SIP > OPTIONS? Can you make a call in or out to it? If you can, it is more > likely that it just doesn't support that and you can't use a qualify > statement. No, I'm not sure. And no, I can't make any call, right now... At least,

...wing error on the cli WARNING[7751]: app_dial.c:1081 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is busy/congested at this time (1:0/0/1) here is my configuration SERVER A sip.conf ~~~~~~~~~~ [general] register=>fromtrixbox:mysecret@192.168.0.7 [receive] ;this should receive call from server B type=friend ;host=192.168.0.7 host = dynamic secret=mysecret context=rizwan ;trunk=yes insecure = very [backtorazi] ;this is used to throw call to server B (works fine) type=friend host=dynamic secret=mysecret conte...

...rk starting from libvirt-5.10 (but I strongly suggest using at least 6.1) > > [root@xx ~]# qemu-img -V > > qemu-img version 2.12.0 (qemu-kvm-ev-2.12.0-33.1.el7_7.4) And qemu-4.2 > > Copyright (c) 2003-2017 Fabrice Bellard and the QEMU Project developers > > 1. assign $MYSECRET to libvirt secret using the secret-define and secret-set-value commands,and $MYSECRET is in base64 format > > MYSECRET=`printf %s "123456" | base64` > > 2. created a disk encrypted in luks format > > qemu-img create --object secret,id=sec0,data=$MYSECRET,format=base64...

...nfigure 2 parallel sip trunks between 2 boxes. However I seem to have the problem that when making a call from Box 2 to Box 1, it sometimes says authentication failed because it is using the username of the other trunk. Here's my configuration: Box 1: [dp-dp2] type=peer username=dp-dp2 secret=mysecret qualify=yes host=box.2.ip.address context=from-internal [e911-dp2] context=from-pstn host=box.2.ip.address qualify=yes secret=mysecret2 type=peer username=e911-dp2 Box 2: [dp-dp2] host=box.1.ip.address qualify=yes type=peer username=dp-dp2 secret=mysecret context=from-pstn [e911-dp2] context=f...

Good day all I've tried my iax conf and I'm struggling.So I want to know If someone else got this working and if they can pleas send my their configs I have to asterisk server,in different tows,both offices connected wit a direct line so both servers are on the same network running SIP.Each town got different extension register to each sever.Town A=100+ town B=200+ How do I get town A

....20) to my real server (e.g. nasir.server.com) which has abc as user configured same as on system1 (192.168.0.20), call goes to [default] instead of going to [payasyougo] context and is treated as incoming call... when we use register string calls works ok on real server too. I also tried SIP/abc:mysecret/${EXTEN} and SIP/${EXTEN}@abc:mysecret but nothing seems to work. there is another problem that sometime my real server (nasir.server.com) becomes unreachable and this error is returned NOTICE[3898]: chan_sip.c:11489 sip_reg_timeout: -- Registration for ' abc at nasir.server.com' tim...

Hi list! Finally I got my Asterisk works with my two phones... It was a problem on my Firewall (for the phone of my wife) and on my Dialplan (for forwarding calls). Now all works as expected, at least in the simulation I did with AsteriskNOW. Hopefully it will work later, when Deutsche Telekom changes my ISDN to VoIP... Well, now I have some time to spend with "fooling"... My phone

...To: lines seem to both show it from hostname jtsd05, though there's also the line saying it's going to 192.168.200.99 (the phone). I've played with all sorts of settings in sip.conf, but the messages persist. Here's what I've got: [hft0] type=friend username=hft0 secret=mysecret context=outtrunk-office host=192.168.200.99 disallow=all allow=ulaw dtmfmode=rfc2833 progressinband=no ;Polycom phones have trouble with the progressinband=never callerid="HFT Booth 0 <(619) 364-4850>" allowsubscribe=yes And some of the Polycom phone config: reg reg.1.display...

Hi, I am new to this list and this is first time i m posting here. please help me out currently I am working on dialing a sip peer on an asterisk server from 2nd asterisk server. scenario is like this. on my system i am using this peer in sip.conf. [abc] type=peer username=abc secret=mysecret host=192.168.0.20 context=default dtmfmode=rfc2833 ;restrictcid=no canreinvite=yes insecure=invite,port disallow=all allow=ulaw allow=alaw allow=g729 allow=gsm nat=yes qualify=yes and using following register string register => abc:mysecret at 192.168.0.20 <abc%3Amysecret at 192.168.0.20&g...

...39;cdr' table in my MySQL-DB. On this table the user 'asteriskcdr' has select, insert, update privileges. GRANT SELECT , INSERT ,UPDATE ON `Asterisk`.`cdr` TO 'asteriskcdr'@'127.0.0.1'; cdr_mysql.conf : [global] hostname=127.0.0.1 dbname=Asterisk table=cdr password=mysecret user=asteriskcdr port=3306 sock=/tmp/mysql.sock userfield=1 I really don't know why Asterisk cannot connect to the table.. Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/201...

...hy. I don't understand why this wouldn't work. ----[Extra Information]---------------------------------------------------------------------------- My current setup: On computer 192.168.10.50 **sip.conf*** [general] port=5060 bindaddr=0.0.0.0 context=default allow=gsm register=>comp1:mysecret@192.168.10.51 [comp2] type=friend username=comp2 secret=bigsecret host=192.168.10.51 **extensions.conf*** [general] static=yes writeprotect=no [default] exten=>_221,1,Dial,SIP/comp2 -================= On computer 192.168.10.51 **sip.conf*** [general] port=5060 bindaddr=0.0.0.0 context=defau...

...ror: Broken pipe [Jul 26 16:29:14] ERROR[1579]: utils.c:1178 ast_careful_fwrite: fwrite() returned error: Broken pipe == Manager 'mark' logged off from 192.168.25.241 *This is my manager.conf:* [general] enabled = yes port = 5038 bindaddr = 0.0.0.0 webenabled = no [mark] secret = mysecret permit=0.0.0.0/255.255.255.0 read = system,call,log,verbose,command,agent,user,originate write = system,call,log,verbose,command,agent,user,originate *This is my code in PHP*: <?php function ast_claves(){ $socket = fsockopen('192.168.25.18','5038',$errno,$er...

...ice => /dev/ttyS2". The modem seems to be initialised. I am not sure what to do with "stripmsd=1"... extension.conf: I've added "exten => 5957,Dial,Zap/1" to "[local]". iax.conf: I've added the following block: [kubi] type=user context=local secret=mysecret I'd like to dial a phone number (5957, my other phone) from gnophone. gnophone's settings: Use asterisk server 192.168.2.254 port 5036 context local username kubi password mysecret When I try to dial a number with gnophone the server logs the following line: Jul 23 16:11:19 NOTICE[6554...

Hello, fresh install of Asterisk 13.11.2, client unable to register. For now I have IPtables disabled, also selinux is disabled [1006] type=friend username=1006 secret=mysecret context=sip-phone call-limit=1 callerid="iuser" <1006> disallow=all host=dynamic allow=all any ideas? Thanks, Motty -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/2016...