similar to: Free Fax for asterisk

Am 19.08.2015 um 19:07 schrieb Steve Edwards: > Please don't top post. > > On Wed, 19 Aug 2015, James Cass wrote: > >> Steve, would you be willing to share that "quick bash script"? > > There's no magic in the script, but here it is, embarrassing myself: > > cp sample-call-file /tmp/ > chmod +x /tmp/sample-call-file >

Markus Weiler <markus_weiler at mailworks.org> schrieb: Hi > from voipinfo... > > If an Asterisk command specifies a sound file in a*subdirectory*, > Asterisk looks in that subdirectory for the language subdirectory. For > example, theSayDigits > <http://www.voip-info.org/wiki/view/Asterisk+cmd+SayDigits>command may > play the sound file

Just a test. -- Matthew Fredrickson Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

Hi again I'd like to configured my Asterisk to use german sounds for the "Say"-commands... I installed the sounds-files and I tried them with "Playback(de/demo-echodone)" and it works. Now I tried to add an extension to say the current time: exten => 24,1,Verbose(2,Time asked by ${CALLERID(num)}) Exten => 24,n,Set(CHANNEL(language)=de) Exten =>

Hi Group, we're just wondering, in German we call the different types of phone-numbers (Geographic,mobile,national,VoIP...) Rufnummerngassen (phone number alleys ;-) ) Is there an english word for this? -- ----------------------------- Markus Weiler markus_weiler at mailworks.org -----------------------------

Hi List, we're calling a sip phone from our Asterisk Server, and try to add logic depending on the dialstatus Stripped down example; exten = 494XXXXXXXXX,n,Dial(SIP/4120089,15,w) exten = 494XXXXXXXXX,n,Goto(98-${DIALSTATUS},1) exten = 494XXXXXXXXX,n,Hangup() ..... exten = 98-BUSY,1,NoOp(Busy) exten = 98-BUSY,n,ExecIf($["${Voicemail}" =

Hello all, I am trying to set up asterisk and hylafax to send and receibe fax. The machine is connected to the PSTN is an Eicon Diva BRI (1 BRI port). My problem is that , when I send a Fax from the PSTN to this machine, the asterisk or diva or hylafax, does not detect this call as a fax and asterisk answer that call like a voice call. How sould I configure the software modem (iaxmodem) to use

Do bristuffed zaptel (zaphfc) supporting rxgain/txgain in zapata.conf? I'm changing rxgain in zapata.conf, and reloading zaptel, but sound level on ISDN(HFC) is always the same (loud).

Hi, I have an asterisk installation with 2 E1 cards Software version is Asterisk 1.2.6 Libpri 1.2.2 Zaptel 1.2.5 I'm having problem with fax transmission, let me explain better my setup: My fist TE110P E1 card is connected to the telco line the second TE110P E1 one to an Nexspan PBX so the server is basically sitting between the line, and the pbx. every call coming from the line is

When I send email from my local asterisk machine, my IP address get's RBL'd. Asterisk is my only reason for running sendmail, so to keep it simple, I tried to make my ISP's mail server a 'smart host' (relaying to a trusted mail server) but my ISP doesn't allow ANY kind of relaying these days. I imagine there are many like me who are not sendmail experts who want to

Hi I have a requirement to bridge Digital ISDN call through an asterisk box but no matter what I setup in the dial plan the second leg of the zap bridge is always set to Transfer Capability of SPEECH, I wondered if any one has come across this and managed to fix it? Thanks in advance for your help Robb

Hi Timo, thank you very much for your quick reply. I tried that but it is not possible to SELECT such a mailbox. I have in dovecot.conf: --- mail_location=maildir:/data/messages%h namespace private { separator = / prefix = location = maildir:/data/messages%h inbox = yes list = yes } namespace public { separator = / prefix = greetings/ location = maildir:/data/greetings%h

Okay, I've been lurking on this list for several weeks and have finally gotten rsync to work well when transferring files from one box to another. It is really great for passing lots of files from one machine to another, i.e. ripped MP3 to my home Linux machine, as one example. Now I am ready to take the next big step. I have a large 1.5 terabyte server from Raidzone that I have NFS mounted

Hi Jeremy,   I got some info on that topic from the illumos devs:   > It's a sporadic issue, you're lucky enough to not encounter it on 4.9.5. > > I confirmed in 4.10.2, it happens: > > winbindd.log:  tdb(/tmw-nas-3p/samba/var/lock/gencache.tdb): tdb_open_ex: tdb_mutex_init failed for /tmw-nas-3p/samba/var/lock/gencache.tdb: Device busy > > So either apply OS fix, or

Hi All, I was hoping someone would know the answer to this question I had... I know the rsync daemon has the "Max connections" module, which works great. However, I was wondering if there was a good way for me to: 1: Query rsync to get the number of live connections it thinks it has? 2: Or better yet, how does rsync itself check for the number of connections? I'm trying to

Hi ya'll- I've got this odd openssh problem with Solaris 10 I was hoping someone could shed some light on. Not sure if it is a bug... Basically I'm trying to use pubkeys as an auth method, but am having issues. I can log in using passwords no problem, but as soon as it notices a matching public key it closes the connection. I ran the sshd server (on Solaris 10 box) in debug

I added LOG statements to mutex.c printing the strerror from the return values.   From   ret = pthread_mutexattr_init(&ma); in line 570   I got errorno 16 --> device busy.   But due the manual of pthread_mutex_attr_init() there should only be ENOMEM and EINVAL...   Doing gdb smbd -> run Doe not reveal anything, but I never used gdb for debugging anything. I just gave it a shot...        

Hi all, inside my * is a E400P. The machine is a PII 400Mhz with 256MB Ram. OS is Debian woody. * is the newest cvs co. I have written a little callgen script which make outgoing calls through my *: #! /bin/sh set -e n=$1 # Nummer anz=$2 # Anzhal der Versuche anz2=$3 # Kan?le sle=$4 # Timeout bis zum n?chsten Versuch if [ -z $4 ]; then sle=0 fi s=1

I have my peers registered to SER.asterisk seems to be sending mwi for the peers seen in the sip show peers CLI command. i have my ser server registered with asterisk as a type=friend and all clients register to ser.how do i get mwi to work for these clients registered to SER. Thank you, -AA

hi, i'm having problems getting asterisk spliced between an E1 PRI (german Telco Arcor) and an Ericsson Business Phone 250 digital PBX. The Asterisk Server has a TE405P with it's port 1 connected to the E1 PRI provided by our telecommunications provider Arcor and port 2 connected to the E1 PRI of our Ericsson BP250. the setup before: Arcor TelCo PRI(E1)