search for: 9xxxxxxx

Incoming calls from my Sipura 3000 don't seem to be correctly routing to Asterisk (or something?) Here is my Asterisk configuration for my incoming PSTN line: Code: [1000] type=friend host=dynamic context=incoming secret=6769 dtmfmode=rfc2833 disallow=all allow=ulaw insecure=very Inside of extensions.conf, I have this: Code: [incoming] exten => s,1,Answer( ) exten =>

Do you have DISA setup? We're seeing lots of attackers running scripts that send digits until they strike a DISA, misconfigured mailbox, etc. (Assuming it wasn't a stupid employee forwarding an inbound call to a 9xxxxxxx number etc). Have a look at SecAst (www.generationd.com) - it detects callers sending too many digits, monitors digit dialing speeds, etc. to help identify and block these types of attacks. The free version is better than nothing (but if you've already suffered one $25k attack then you probab...

I'm having a problem getting 3 way calling to work correctly using two outside lines and one extension. The two outside lines are connected to the X100P's and a standard model 2500 phone is connected to the TDM10. When I dial the first outside destination 9xxxxxxx, the call completes correctly. When I flash the hook switch and dial the second location 9yyyyyyy. The call doesn't complete, and most of the time (but not always) the dial tone is not broken by the digit following the '9'. My configuration files are: # # zaptel.conf #...

Dear Sir, I use asterisk 1.8.11 (192.168.100.202)to connect lync server .I use tls port 5068 to connect to this lync server . The tls is ok to establish and I make call from softphone 3200 (register to Asterisk) and dial 9XXXXXXX (9+85225082162) , this prefix will dial to trunk lync_trunk and pass to lync server(192.168.100.14) using tls . But the lync client in opposite side ringing and they recevie the call , but when they answer the call , the call drop and hang up immediately .In sip trace I see there is "call leg...

Hello everyone, I'm trying to send a received fax with mutt, when I try it from the Linux shel it works, but when trying with Asterisk's System command it doesn't. Successful Linux command: echo | mutt -s "New fax" earohuanca at gmail.com -a /tmp/faxes/201306191111.tif Unsuccessful Asterisk Command: same => n,System(mutt -s "New fax" elder.arohuanca at

...9;t dial out to the PSTN? I have tried everything, and I'm hoping someone else can shed some light on this. I'm open to ideas. I've already removed the barrier codes, and disable access code requirements on Tie and Non-Tie lines, with no effect. I made sure that the Asterisk is dialing 9XXXXXXX when sending the call over the DAHDI trunk to the Merlin. Whenever you call from the Asterisk to the Merlin you are redirected to the "Unassigned Extension" extension, and dropped to the Operator. I have a suspicion that this might have something to do with the NetwkService on the Slot 4...

Hello, I'm investigating a situation where there was a hundreds of minutes of calls from an internal SIP extension to an 855 number in Cambodia, resulting in a crazy ($25,000+) bill from the phone company. I'm investigating, but can anyone provide some feedback on what's happened here? I'm investigating how this happened as well as what types of arrangements can be made with the

Hello, I'm having issues connecting throu PRI with the following error "Requested transfer capability: 0x00 - SPEECH" Below are the logs: == Using SIP RTP CoS mark 5 -- Executing [97052660 at voipphones:1] Set("SIP/4856-00000003", "CALLERID(num)=xxxxxxxxx") in new stack -- Executing [97052660 at voipphones:2] Dial("SIP/4856-00000003",