similar to: X100P to hardware PBX

Ok, the Zhone sucks and the Adtran 750/850 seems to be a little too expensive. Can anyone recommend a decent channelbank that won't break the bank? TIA, -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 Pulaski Chamber 2002 Small Business Of The Year

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

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,

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

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

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

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

Hi Folks, Prior to upgrading my Zaptel sources everything was working fine. I have a X100P connected to my analogue line. The handset port of the X100P is connected to my desk phone's line 2 input. When the analogue line rings I see the CID on my line 2 but not from Asterisk on line 1 via the Cicso ATA. This used to work fine until I upgraded the sources. I get this when watching the

Hi, I am the "IT guy" at a small startup based in UK. At the moment we have 3 analogue (PSTN) lines and we will be adding another 2 or 3 soon. Later on we should be changing to ISDN30. One of the partners mentioned getting an analogue PBX now, and when we move to ISDN, then get a digital PBX. I though of Asterisk. I have seen the website in the past and I know that it can do the job

If you are looking for real high density VOIP termination I would look at > something like a Lucent APX 8000, configure correctly it can pass 2500+ > g.729 calls to the PSTN course we paid lots of $ for ours. > > Chris > Chris, My experience has been that the APX and TNT products require a single SIP proxy, how are you load balancing 2500 calls? If all of the traffic is

Hi, I suspected that I the analogue phone should have got a pass through signal when the power was off to the server, unfortunately it doesn't. I kept asking digium support about that but they didn't give me an answer. The problem is how do I identify whether the X100P is incompatibel with the network or faulty without possibly wasting another USD100??? Aaron On Sat, 2004-05-15, Eric

Hi, I am trying to receive a fax with the spandsp library. The sending fax says "success" but there is no tiff file generated. I use "exten => 7000,1,rxfax(/tmp/testfax.tif)" in my extensions.conf. The connection is via SIP/G.711 as I have read on the list that this can sometimes work (I know Fax over IP is troublesome without T.38). I think the transmission should not

After scouring the list archive and not finding the answer I decided to post to the list. I'm sure the answer is glaringly obvious but please bear with me. Using Asterisk, I'm tasked with setting up a SOHO with 5 analogue (FXS?) lines and a number of soft-phones for internal extensions. I'm confused by the telephony hardware needed for this exercise - 1) I need the equivalent of two

After reading through several recent threads, I started to wonder why the Cisco document (and other VoIP documents) appears to present this issue as VoIP gateway specific. Don't (plain old) PBX' face the same issue if they use analogue interfaces? If there are analogue PBX' at all, how do they solve the problem? Yuan Liu

Hi All I've done some reading on the wiki and read some of the mailing list archives, but can't see anything on this. I guess this means I'm either searching on the wrong thing, or have totally the wrong idea... Can anyone suggest if the following is possible? Currently, our office has a 24 analogue extension PBX, and 2 ISDN lines providing it with external connectivity. We have

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

Are they any issues still with hyperthreading processors, I've read and been told by a few people to make sure its disabled in bios if I want to use * on a hyperthreading machine. Kind Regards, Chris Bond

Hi folks! I'm using an X100P (connected to my phone line) and an S100U, and when I calls out from the phone connected to the S100U it is a very bad sound quality, it "pops" and "jitters" a lot. But internal calls from for example a SIP client to the phone on the S100U sounds good. Calls from an SIP client to the outside world using the X100P also works good! My second

I have * configured with 2 X100P cards (fxs_ks). The lines from the telco are 'analogue both way ddi trunks'. This means that every inbound call contains digits that represent an extension on the PBX. I can make outbound calls from * with no problem however I cannot receive inbound calls on these trunks. Some investigation has shown that when the line is idle there is 50 volts present and