similar to: nvfaxdetect in 10.0

Hello list, This may or may not be Asterisk related, but if I had hair I'd pull it out over this. I have a TDM400P card in a Dell POWEREDGE 1550 running Asterisk 1.4.30. Everything works great except that every time it rains, I get flooded with this CLI message - == Starting post polarity CID detection on channel 1 -- Starting simple switch on 'DAHDI/1-1'

Hello What is the difference between these 2 version of Asterisk in terms of functionality. For a small office am I going to run into problems if I use the easy version... Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050717/311c56ec/attachment.htm

Good Afternoon, I have an Asterisk box that is acting like it is passing "core show" before every command I type. For example, if I type sip, I will get "No such command 'sip' (type 'core show help sip' for other possible commands). Any ideas? -- -jayson

Hello, I'd like to run Asterisk on an embedded device, where space is scarce. It should be able to handle calls from a VoIP provider in SIP, calls from the PSTN through Dahdi, and voicemail. If someone's already done this, I'd like to know which directories/files are required for a basic install? Does this look right? ================= /bin/asterisk /etc/asterisk/ asterisk.conf

Hello my friends, I'm very worry about a problem i'm having...my asterisk got freez some times, every 5 or 6 days with NO trace in /var/log/asterisk/messages What i want to know is if safe_asterisk has something to be with this? This is what i have on my server: [root at mypbx ~]# ps -A | grep asterisk 9118 ? 00:01:30 asterisk [root at dreampbx ~]# ps aux | grep asterisk root

Hi all, I am handed a project to setup *. The requirement is that it can handle 8 T1s. Half of the calls coming into the system will be routed to SIP extensions (with transcoding). The machine we have in our disposal is a new dual Xeon 3.2gHz server with 2g of ram and an dual 1000mb nic. Voice will be coming in from the PSTN (through 2 quad digium cards) in g711ulaw, and most of the time will

Hello, I have Asterisk 1.6.2.9-2, the directive "sendrpid" does not work! I placed this in my peer: (sip.conf) sendrpid=yes trustrpid=yes or sendrpid=yes trustrpid=no (and restarted Asterisk) and the line "Remote-Party-ID" does not appear in my sip debug! Please help me, Mickael. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello - I am looking at connecting 7 - 10 locations together using Asterisk and possibly some VoIP gateway appliances. I need to insure best voice quality as these trunks will be used primarily for customer calls. I am considering implementing a full T1 frame relay circuit to each location which can be done for a reasonable cost. DSL and Cable are currently at each location and setup for

Sending works but on receive it keeps failing - reporting back 'training' failure. I am using Asterisk 1.6 with T38. What should I post to the list to assist diagnoses? Michael

Anybody here having trouble receiving email from the list on Gmail? I havn't received anything since Friday July 29.

hi folks. recently some of our customers complained about bad voice quality on the phone system. i looked at the logs and found a lot of these: [2012-11-03 08:26:38] NOTICE[11305] chan_dahdi.c: PRI got event: HDLC Abort (6) on D-channel of span 1 [2012-11-03 08:26:45] NOTICE[11305] chan_dahdi.c: PRI got event: HDLC Abort (6) on D-channel of span 1 [2012-11-03 08:26:54] NOTICE[11305]

Hi, Sorry for maybe not a very list related topic, but I have always been curious if there is information on how many Asterisk based PBXs are operating Worldwide? Thanks and hope the community will not reject my curiosity! :) Best Regards, Vallu Sevana Oy -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have compiled zaptel on Mandrake following everything I have always done on Fedora. It is 2.6 udev so... I had to modify the 01-devfs.rules Make linux26 Make Make install... Everything appears to compile correctly but it says module not found when doing "modprobe zaptel" Is this a rights issue? Jordan Novak -------------- next part -------------- An HTML attachment was

I am posting here since you guys are my last hope. I am trying to configure a Polycom Soundpoint IP 335 with MWI. Is there any way to eliminate the scrolling messages and Msgs softkey? I am trying to get it where it's just the light that indicates the new messages. I don't know if Asterisk has to send a different notification or what have you. Thanks, --Eric -------------- next

I'm trying to setup asterisk to accept call from Teliax, request the 10 digit number from user, then dial it thru the VoIPJet. If I'm not wrong I will be charged by both providers because both connection is active during conversation. So my question is can I set the things so that I pay only to VoIPJet? Specific configuration snippets will be greatly appeciated. Thank you.

Hi, Has anyone met any success, installing localized (ie non-english) menus within SIP firmware enabled Cisco 7941 ? Those phones seem to be trying to download localized menus from Cisco Call Manager but as they are managed by an Asterisk server, I'm looking for a workaround. Any advice ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

We have written some monitoring and stat collection scripts that use asterisk -rx "command" The script runs once a min and logs data and posts any critical notifications. Everything is working well with this method but we get the -- Remote UNIX connection / disconnect message once a min and we would like to suppress it. Is it possible without reducing the verbose logging level.

Hello, I run a small network consulting company in the Chicago area and I have a client who is interested in doing an asterisk based VOIP installation. My company does not have the necessary experience to carry out the project alone so I am looking for an asterisk guru to lead the project. I'm interested in someone from the Chicago or northwest Indiana area who is very experienced with

Please help me test my setup by dialing 800.564.0215 and listen to the queue for a bit. I have a quad port T1 with NFAS setup. I can dial-out but I cannot dial any 800 numbers (Global Crossing says I need LDS service and that will be a couple weeks) so I cant test it myself. I need at least 24 callers to feel comfortable enough that it is working properly. Thanks, Steve Totaro

Has anyone got this annoying sidecar to illuminate when users are on the phone? In my function key settings I have: Context: Active Type: Extension Number: <sip:4000@serve.address.com;user=phone> (4000 is the extension I want to see/dial on the key). I can press the key and it will dial the extension, it just won't illuminate when the user is on the phone or on DND Since I have