Re: [asterisk-users] Interesting attack tonight fail2ban them

allowguest=yes? Which version of Asterisk are you running? Best regards, Jeroen Eeuwes -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs

Re: [asterisk-users] Interesting attack tonight fail2ban them

to make sure they get the usual errors like Registration from failed - no matching peer found? In other words, how did they get this far in the first place? Best regards, Jeroen Eeuwes -- _ -- Bandwidth and Colocation Provided by http

Re: [asterisk-users] Calls from PSTN on SPA3102

in entradas_pstn is already dialling to SIP/103 so that should be OK. Best regards, Jeroen Eeuwes -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs

Re: [asterisk-users] Sipp and asterisk 10

(deny=0.0.0.0/0.0.0.0) you have to permit afterwards. If you permit first and then deny everyone will be denied. Best regards, Jeroen Eeuwes -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk

Re: [asterisk-users] Beep file with Record

Hi Arjan, I also try to place the voicefile in the directory /var/lib/asterisk/sounds/ and /var/lib/asterisk/sounds/applications/ of but without any success. Just for double-checking, but what directory is listed as the astdatadir in asterisk.conf? Best regards, Jeroen Eeuwes

Re: [asterisk-users] Beep file with Record

/asterisk. If your asterisk.conf says this: [directories](!) ; remove the (!) to enable this you should remove the (!) to enable the alternate directories in asterisk.conf so it should only say this: [directories] Best regards, Jeroen Eeuwes

Re: [asterisk-users] Monitoring second leg being dialed?

don't think it works like that on a PSTN line, you just have to listen to the sounds yourself. Best regards, Jeroen Eeuwes -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live

Re: [asterisk-users] How does wrandom strategy works with Queue?

regards, Jeroen Eeuwes -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users

Re: [asterisk-users] How does wrandom strategy works with Queue?

in the docs. See https://wiki.asterisk.org/wiki/display/AST/Routing+Incoming+Calls+to+Queues Best regards, Jeroen Eeuwes -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live

Re: [asterisk-users] Variable stripping/removing part of string

this: exten = 0424449631,n,Set(TESTING=${CUT(CALLERID(name),\(,1)}) exten = 0424449631,n,NoOp(${TESTING:0:-1}) Best regards, Jeroen Eeuwes -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk

Re: [asterisk-users] context problem

($[${CALL-TO} : .*52525252.*]?TRUNKin,52525252,1) exten = s,n,GotoIf($[${CALL-TO} : .*59595959.*]?TRUNKin,59595959,1) exten = s,n,etcetera Best regards, Jeroen Eeuwes -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-users] cannot answer incoming calls

Hi John, Interestingly RINGING and REGISTER messages are working OK. The NAT router is out of our control. Are we looking at a SIP ALG getting in the way? It probably is the NAT router. Have you tried canreinvite=no in sip.conf for these phones? Best regards, Jeroen Eeuwes

Re: [asterisk-users] Asterisk hangs up call after 20s

to register on my server (that was usually the hard part!). In sip.conf I have added for all the remote users the setting canreinvite=no. The downside to that setting is that Asterisk is always in the audio path. For my situation that does not really matter. Best regards, Jeroen Eeuwes

Re: [asterisk-users] Simplifying dial-plan

and you've got 10 digits or if you only receive 7 digits you want to add both an 1 and areacode 555. Because you've added the extra digits yourself it will match to the _1NXXNXX extension and start there at 1. Best regards, Jeroen Eeuwes

Re: [asterisk-users] Simplifying dial-plan

want that like this: exten = _NXX,1,Set(CALLERID(all)=No one cares 0) exten = _NXX,n,Dial(SIP/${ext...@abcdefgh) exten = _NXX,n,Goto(h,1) Best regards, Jeroen Eeuwes -- _ -- Bandwidth and Colocation Provided by http

Re: [asterisk-users] Specifying DID for outbound calls

= _1NXXNXX,n,Goto(h,1) But in my case I had two different domains. E.g. Dial(SIP/${ext...@provider-id1) and Dial(SIP/${ext...@provider-id2) instead of setting the CallerID. Not that the Cut doesn't work correctly if you use a minus-sign in the username. Best regards, Jeroen Eeuwes

Re: [asterisk-users] Configuring Softphone

is blocking stuff it is bound not to work. Something else you could try is to configure a softphone on a PC on the same LAN as the Asterisk box. That way you are by-passing any router issues. Best regards, Jeroen Eeuwes

Re: [asterisk-users] Configuring Softphone

box. Perhaps you are over-complicating the issue? If you have a working dialplan for other phones then why are you trying to set it up differently? Have you tried just using the same settings as a working phone? Best regards, Jeroen Eeuwes

Re: [asterisk-users] Configuring Softphone

you have to either put the real IP address or a resolvable hostname on the defaultip line OR remove the defaultip line. The latter is probably the easiest. Best regards, Jeroen Eeuwes -- _ -- Bandwidth and Colocation Provided