similar to: ignore a DID?

Hello, Is there a published apps API? Or do I need to just start reading source code? -- Jesse Guardiani, Systems Administrator WingNET Internet Services, P.O. Box 2605 // Cleveland, TN 37320-2605 423-559-LINK (v) 423-559-5145 (f) http://www.wingnet.net

Hello, I've installed speex 1.1.7 and asterisk 1.0.6 from Gentoo's Portage and I'm experiencing asterisk crashes whenever I try to make a connection from my X-Lite client under wine to asterisk using the speex codec. I know speex is being attempted because SPX lights up on the X-Lite display. Also, I know that speex is causing the crash because X-Lite works fine if I use GSM, ILBC,

Hello, Does anyone know of an audio player for Linux that will play asterisk's GSM files? I know I can convert them using sox, but I'm hoping to play them natively so I can test out some voicemail settings. Thanks! -- Jesse Guardiani, Systems Administrator WingNET Internet Services, P.O. Box 2605 // Cleveland, TN 37320-2605 423-559-LINK (v) 423-559-5145 (f) http://www.wingnet.net

Howdy list, Sorry if this is a frequently discussed topic, or an off-topic question, but I couldn't find much info about my question by performing quick searches in the archives, and my question is pretty tightly related to security... Background: =========== I have a number of FreeBSD machines. Most are 4.x, but a few are 5.x (mainly the testing/devel machines). I also have a single Red

Hello, Can anyone tell me what the "normal" number of interrupts per second is for an X100P card? I've used FreeBSD 5.3 and a linux 2.6.11 kernel on the exact same hardware (only the disk changed) and `systat -vmstat 1` on FreeBSD and `procinfo -dS -n1` under Linux. For both, I'm seeing roughly 1000 interrupts per second on my X100p card. It was a bit worse under FreeBSD, and I

With softphone I mean linphone csipsimple or whatever. How should a dialplan lokks like? On 31.01.19 11:26, Antony Stone wrote: > On Thursday 31 January 2019 at 10:59:01, basti wrote: > >> Hello I use this dial paln: >> >> [o2-in] >> exten => o2,1,Answer >> exten => o2,n,Playback(hello-world) >> exten => o2,n,Ringing >> exten =>

Hi, here's my result for # doveconf -n|grep ssl ssl = yes ssl = required ssl_cert = /etc/exim/ssl/wildcard_cantella_com.crt.chained ssl_cipher_list = ALL:!LOW:!SSLv2:SSLv3:!EXP:!aNULL ssl_key = /etc/exim/ssl/wildcard_cantella_com.key verbose_ssl = yes On 04/20/2017 11:27 AM, B. Reino wrote: > On Thu, 20 Apr 2017, Alvaro Lacerda wrote: > >> These are my dovecot logs. It

Hi, This is my environment: SMTP: Exim 4.89 with Mailscanner 5.0.3 IMAP: Dovecot 2.2.10 At the moment I'm just trying to test out my Dovecot to check if it's listening on port 993. *netstat -tuln:* Shows that my machine is listening on ports 143 and 993. *telnet localhost 993: *This is my issue, I get the following message: # telnet localhost 993 Trying ::1... Connected to

Hi Webert -- Thanks for the quick reply and observation. When I try "telnet localhost 143" I get the same result. It does connect but then the connection is dropped. # telnet localhost 143 Trying ::1... Connected to localhost. Escape character is '^]'. Connection closed by foreign host. I never get the "* OK [CAPA....]" Thanks, On 04/20/2017 10:41 AM, Webert de

Hello, I'm running Asterisk 1.0.6 with zaptel 1.0.6 on Gentoo Linux with a 2.6.11-gentoo-r2 SMP kernel (but no SMP hardware) and mpg123 0.59s-r9. When I leave a voicemail message via my X100P, the message is way too quiet. I can barely hear it. I googled this a bit, and I saw similar complaints with older versions, but no resolutions. Also, many complained that it was only too quiet via

Hi B. Reino, These are my dovecot logs. It looks like this error pops up every time I run "telnet localhost 143" Apr 20 11:11:39 imap-login: Fatal: Couldn't parse private ssl_key: error:0906D06C:PEM routines:PEM_read_bio:no start line: Expecting: ANY PRIVATE KEY Apr 20 11:11:39 master: Error: service(imap-login): command startup failed, throttling for 60 secs On 04/20/2017

I'm presently working on an office move and evaluation of telecommunications services needed at the new location. I'm presently wrastling with an issue related to portability and geography between landline carriers. Presently certain people within the organization are hopelessly in love with our 909-822-xxxx number(provided by pacbell/att). As that number is presently provisioned it

Hello I use this dial paln: [o2-in] exten => o2,1,Answer exten => o2,n,Playback(hello-world) exten => o2,n,Ringing exten => o2,n,Dial(SIP/10&SIP/20&Local/s at no-op,25,rt) exten => o2,n,Playtones(425/1000,0/4000) exten => o2,n,Wait(30) exten => o2,n,Hangup() All is fine. Hello world is Playback and I hear a ring tone. If I remove the Playback hello-world. No ring

Hello, I noticed this while attempting to run a DOS software suite. The CALL command in wcmd doesn't seem to honor the set PATH. If I put this in my environment (via regedit): c:\windows;c:\windows\system;e:\;e:\test;f:\;f:\downloads\daphne\tools Then I put the following in f:\downloads\daphne\tools\bob.bat: path c:\windows;c:\windows\system;e:\;e:\test;f:\;f:\downloads\daphne\tools

Hello, I've been attempting to get this working all day. :) I have an RHEL3 machine connected to a Windows 2000 PDC as a domain member. I'm running samba 3.0.21c, compiled from an SRPM: # rpm -q samba samba-3.0.21c-1 When I try to list my users, I get this: # wbinfo -u Error looking up domain users Exit 1 Similarly, `getent passwd` fails to show any SMB users. But `getent passwd

Hello, Is anyone successfully using spandsp 0.0.2pre25 with either asterisk 1.0.x or 1.2.4? I've built a Gentoo ebuild for this version of spandsp and app_rtxfax, and it builds, but I'm not having any luck getting it working. 99% of my test faxes fail. Reverting to 0.0.2pre20 yields a much higher success rate. I've bumped the console debugging level in logger.conf to include debug

Hello, Does gnophone 0.2.4 work with asterisk 1.0.6? I'm running Gentoo Linux with a 2.6.11 kernel, and I'm getting the following when I try to connect to asterisk from gnophone: Mar 23 12:09:33 WARNING[25059]: chan_iax2.c:546 iax_error_output: Information element length exceeds message size Mar 23 12:09:33 WARNING[25059]: chan_iax2.c:5332 socket_read: Undecodable framereceived from

Is it possible to make a call from a SIP/IAX softphone, say Zoiper, on one computer to an Asterisk system without having an extension/account? If so what are the terms I need to search for to figure out how to do it? So far "anonymous SIP" has got me the closest I think but no brass ring. TIA, Rod --

Hello Everyone, Pardon me if im sounding like a total idiot, but im new to this and have to ask. Numerous people have been telling me that I will be able to somehow do long distance calling for free when I roll out Asterisk.. and yet none of them can explain to me how exactly that will be. So.. I'll ask the community at large.. is this total BS or is there actually a way to reduce my long

Hi I believe this is due to the way US phone systems work, however I'm going to ask anyway. In the UK there are several providers who provide national rate PSTN => Voip gateways which are free to receive calls on, (for the recipient), the caller pays the cost of calling. E.g 0844 0870 etc. I am looking for a US provier who offers the same sort of system. I don't call the US but I