similar to: Channelbanks

Hi! Is there anyone that are using a E1-channelbank and have any tips about some type? Im looking at the TE410P and use one port for a PRI (Euro-ISDN, I think we're using some slightly modified version here in Sweden, but I'll check that tomorrow) and connect one port to a channelbank for 30 analogue telephones. It would also be great to get callerid on the analogue phones, so it would

First time around, I just unloaded/reloaded the modules. The box locked up tight. On reboot, I get this: general protection fault: 0000 CPU: 0 EIP: 0010:[<c01defb3>] Not tainted EFLAGS: 00010097 eax: f61d4260 ebx: f61d4260 ecx: ffffffff edx: f61d425f esi: f61d4264 edi: f61d4260 ebp: f4de7f14 esp: f4de7ef4 ds: 0018 es: 0018 ss: 0018 Process sh (pid: 387,

Can anyone point me to some sample Cisco QoS configurations suitable for IAX2? I've looked through Cisco's site, and get overwhelmed with the level of documentation (too much of a good thing). My PSTN gateway and PBX (both *) are connected via 2xT1 (per-packet load balancing) between a Cisco 7206 and a 3640. When the total bandwidth pushes much past 50%, I start getting some crazy

Quick question: My PHP script is now able to connect to the manager port and successfully authenticate using MD5. I would strongly prefer not to do plaintext authentication at all. Would anyone object to plaintext authentication being left out? -- JustThe.net Internet & Multimedia Svcs. [The Fusion of Content & Connectivity] 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve

First question: Do you need to use a special phone to use SIP? My setup is X100P and TDM10B (I have 2 computers with the same setup), will I be able to use SIP? I already have the two computers calling each other using IAX, but I would like to see if I can get them to work using SIP. My current setup: On computer 192.168.10.50 **sip.conf*** [general] port=5060 bindaddr=0.0.0.0

The demo 1235 extension that Asterisk ships with works fine and the messages are sent to the address I set in voicemail.conf. I guess that means that my configuration is working perfectly so far. But when I set up another extension with a voicemailbox, no mail is sent when a message is left, although I can dial voicemail and listen to the message just fine which I guess rules out voicemailbox

When setting include => daytime|9:00-21:00|mo-fri|*|* How does this determine what is different between 9 AM and 9 PM And after hours ??? I want different hours on Saturday and Sunday And a different welcome message after hours Any help appreciated Regards Mick

Ok, this is probably an incomplete thought, but over the last few weeks of reading this list, I think I'm ready to start designing my system, and would like to solicit input. Currently, I have 2 POPs, one each in 2 different states. Each has a Lucent TNT w/~400 trunks. Each TNT has the capability to provision a PRI that can be piped into an * system. So, my interface to the PSTN is

http://bugs.digium.com/bug_view_page.php?bug_id=0000847 I'm placing a $250 bounty on getting Zaptel drivers working under BSD. While the 'hood' is open on this, we'd like to have drivers completed for FreeBSD, NetBSD, and OpenBSD on the x86 platform (other platforms optional). This needs to be done in a timely fashion. I'd like it completed by 6/30/2004, but once the project

Hi all ! Have anyone a resource / link for documentation to configure Asterisk to act as a SIP 2 PSTN gateway (ISDN PRI) ? Thx. Regads, Andreas. -- "If you want to pray. Go to the sea." ---------------------------------------------------------------- Andreas Czerniak <cognac@amcs.net> PGPkey http://pgp5.ai.mit.edu:11371/pks/lookup?op=get&search=0xEDB224EC

Can anyone share any advice / documentation on how to configure a Tellabs 2572 T1 echo canceller? I connected one between a T100P and an Adtran TA750 FXO/FXS channelbank, but when echo cancellation is active I get a LOT of snap-crackle-pop (and other problems) on the line. The 2572 has a bunch of configuration options and they're impossible to guess without documentation, which I cannot find

All, For the last 10 months, I've been using strategy=ringall. This has worked fine and did what I wanted, but at this point, I'm needing to implement a 'penalty' or delay for some members of the call queue. 1: remote users (remote flunkies) 2: level-1 support (flunkies) 3: level-2 support (glorified flunkies) 4: level-3 support (super flunkies) When a call comes

Hey all, I've got quotas enabled and turned on, it works fine for my first test user: $ quota -u user1 Disk quotas for user user1 (uid 2001): Filesystem usage quota limit grace files quota limit grace /var/mailboxes 2772 450000 500000 30 0 0 Then, I used the edquota to generate quotas for the next 1000 users: $ edquota -p user1 2002-3000

Wondered if anybody might have some ideas about what could be causing this.... I have a T100p hooked up to an Adit 600 with 12 channels of voice off of a T-1 coming in. I have a t100p connected to a zhone z-plex with 24 fxs going to my stations. Some of the station are 2 line phones. These have 2 zap channels that are dialed when the extension is matched. Also, some extension require ringing 2

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I've been seeing some strange issues recently as it relates to our Asterisk phone system. Let me give a little background on things: We have a T1 provided by Knology. We recently had an outage of Internet and phones. The following problems (except for the log messages) seemed to have cropped up after the outage. I was prompted by a friend of mine

I'm sure asterisk would make a great stand alone voice mail server. Basically I want to get rid of our voice mail system and replace it with *, but the problem is we use a cisco cluster with skinny clients. So I was thinking the way to contact a * server, would be through our 3640. But so far any attempt has failed. I am wondering if anyone has done something similar. Just want to verify the

Hi All. I'm working on an * configuration. We require 8 inbound POTS lines, and CT1 or PRI seems like it will be quite expensive at that level. I've read that a T1 Channelbank plus the T100P would be a (the?) way to go for this situation. What is the recommended channelbank for use in this scenario? From searching the archives I see a lot of suggestions to get "a

I have asterisk successfully dialing out using a X100P over a normal analogue PSTN line. But when I try to dial out over an analogue line that goes via a hardware PBX the call asterisk does not dial. Is there a configuration change I should make ? I am thinking of something like not wating for dial tone.

The new app_groupcount looks great for most applications but it a is a step back for call queueing... since app_queue calls physical interfaces and not extensions, app_groupcont can't be used to limit the calls passed to a dynamically added agent. I presently use the broken sip incominglimit feature (even though it's less than ideal as it also limits outgoing calls preventing

Trying to find my feet here. If I wanted to connect Asterisk to a PRI and throw in a T1 Adtran channel bank into the mix for fax machines would the following work? Connect PRI line from telco to Port 1 on the Digium Wildcard TE205P. Connect Adtran TA-624-T1 to Post 2 on the Digium Wildcard TE205P also, would I need a crossover to the channelbank or is it a patch lead like the connection to the