similar to: zaptel telephone cards and asterisk in another pc

Anyone using this feature of asterisk's voicemail? I'd never heard of ADSI, and saw it as I was perusing the voicemail source this morning. Is it some kind of visual way of managing voicemail on your phone's display, or does it require a terminal of some kind? Thanks! j

Hi! I see lots of examples using , but core show application displays | So what is the correct character to use to separate parameters for application, functions and macros? thanks klaus

Asterisk segfaulted on me the other day; how do I tell it to generate a core file so -- if it happens again -- I can attempt to debug? I looked in the obvious places in "make menuconfig" and didn't see anything appropriate. Thanks, -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

So I'm using the READ() application within an IVR, and having a strange issue, and wondering if anyone else has had this problem. When calling from an outside line, and entering the digits during the read() part of my dialplan, it's accepting some of the digits twice, though it's only keyed in once. When testing the dialplan internally, it accepts only the digits that I key in.

Hello, I just bought a TE121 T1/E1 card, and now trying to install it on a 1.4.23.1 asterisk with dahdi 2.1.0.4 Actually first everything went on well and i managed to see my card on dahdi. Here's the output: #asterisk# dahdi_hardware pci:0000:04:08.0 wcte12xp+ d161:8000 Wildcard TE121 and this is the scan: -------------------------- asterisk# dahdi_scan [1] active=yes alarms=RED

Hello. I have a software developer creating a .Net / mono program to use as an AGI script. We are having problems getting it to stream files. From what we can tell, it is talking to asterisk correctly when called from the dial plan. Its stderr output goes to the asterisk console. But asterisk doesn't give any indication that it receives the STREAM FILE command. Asterisk simply quickly

Is there a convenient way to limit the number of call files (outgoing directory) that are processed concurrently? -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 3234 bytes Desc: S/MIME Cryptographic Signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20090226/a46e68fa/attachment.bin

Hello list! I've run into a strange problem today and I was hoping that someone here has seen this before and maybe can give me a hand: I'm using asterisk 1.6.0.22 in this config: (A)PATTON ISDN ->(B) ASTERISK -> (C)PATTON PRI -> PSTN -> (D)OTHER PBX Strange Problem: USER A calls makes a call to a PBX over the PSTN and ends into an IVR. When the user makes a selection and

I am thinking about a change to our company's phone "layout" and would like to get comments from people who have done something similar. Currently, we have 3 locations - each with their own Asterisk PBX. The corporate office has a PRI. Each remote location has a SIP provider for 5 channels of SIP going to their own PBX. Interoffice calls use the PSTN. Most inbound calls come to

Hi, all For some work i'am trying to connect to Asterisk's PBX using isdn cards. But I don't know anything about it. So i'll be pleased to get some information about it! Thanks, Best rgrds!!

I have a machine running 4.1 did hte update to 4.2 my zaptel settings (asterisk pbx) seemed to have gotten lost in the update. The file was /etc/udev/rules.d/50-udev.rules Jerry

Dear, I have two Asterisk PBXs with an E1 interface/RJ-45 port in both boxes. I need to connect both PBXs with E1/R2 and UTP cable. What are the requirements to deploy the UTP cable ??? Straight-through or crossover ??? What are the pinouts in both peers ??? Thanks a lot, Alejandro

Hi I came across couple of pointers on the Internet regarding solutions available for providing hosted PBX service. 1. Multiple PBXs: Using separate hardware to host each PBX. Pretty straightforward, but no hosting company wants to use it. 2. Multi-tenant PBX: Configuring multiple PBXs within the same instance of Asterisk. I.e. partitioning a single instance of Asterisk into multiple PBXs by way

I recently had a system crash and my inbox became inaccessible via IMAP. ls -ld /var/spool/mail drwxrwxr-x 2 root mail 4096 Sep 1 07:54 /var/spool/mail ls -l /var/spool/mail/ignacio -rw-rw---- 1 ignacio mail 7222021 Sep 1 07:54 /var/spool/mail/ignacio /usr/libexec/dovecot/imap * PREAUTH [CAPABILITY IMAP4rev1 SORT THREAD=REFERENCES MULTIAPPEND UNSELECT IDLE CHILDREN LISTEXT LIST-SUBSCRIBED

Hello list, I was looking for a way to implement non-blind call transfers with *, i.e. the usual behaviour of most PBXs when pressing the flash button: - A and B are talking - A pushes flash - A is free to compose a new number - B hears music on hold - A's call is answered by C - A hangs up - B and C are in conversation As much as I can understand, * only supports blind transfers, where if

Just curious if any of the Asterisk installers are doing anything special to protect themselves from a possible lawsuit caused by 911 failure during a Asterisk/computer crash? I realize that any traditional PBX or even a phone line can fail but, anything running on a computer is probably going to be less reliable than most PBXs. Anybody requiring customers to acknowledge and sign any kind of

I'm a network admin that maintains 3 commercial Asterisk servers for my employer. I am wanting to move away from the "pre-packaged" commercial PBXs to a more "pure" asterisk setup. The systems I have utilize a nice web GUI to make changes, but it really limits what I can do beyond what they have programmed into their GUI. Would I be better off starting with: a) Plain

Hello. I am going through the documentation and trying to find if asterisk can help me in my case. It is quite difficult to find answer because I do not know the exact question. I have two location. Each in different country. Both locations have Siemens HiPath - different type and software. I can not use card that would allow me to connect those PBXs using SIP. But I have some free ISDN and

Anyway everything seems to work well. The shares are correctly and users recognize them without problems. getent and wbinfo. The main problem is that apparently the Member DC does not get the UID properly containing the user in the ADDC. There is something that can change in the smb.conf the Member DC, I remember a few years ago with the same AD DC we get the UID correctly. 2016-06-14 12:58

I've installed fail2ban, it seems to be working as it identified my failed test logins, BUT, my question is: what can I do when I see same invalid name trying to login to dovecot, different IP each time, how can I say block each IP as used by this name ? or it that a bad idea ? I can see two persistent attempts as so: I don't have such user 'ignacio' or 'julian' # grep