search for: akiva

I have been attempting to download, compile and configure spandsp to function with * without much luck. I am guessing that some assumptions were made regarding the users knowledge level of Linux. Sadly, I must not live up to those assumptions. My problem begins when after compiling spandsp I look for the app_rxfax.c, app_txfax.c, app_dtmftotext.c and makefile.patch files to place in the

Hi, We've recently deployed 6 Uniden UIP200 phones (running firmware 4.54). We've been having some serious problems: 1) All the phones randomly reboot themselves. Typically when trying to answer or initiate a call. 2) All the phones will disconnect from a calls with the PSTN after 2-3 minutes. 3) The phones are unable to interact with a remote IVR (digit presses are not received at

...d my endeavors pan out, I would be more than happy to implement odbc connectivity. I need to become more familiar with post and mysql first however. Up to this point, I've been strictly a MSSQL DBA due to my job functions. Thanks again for your feedback. Brian D'Arcy Operations Engineer Akiva Corporation E-Mail: bdarcy@akiva.com Web: http://www.akiva.com Phone: 760-710-3209 -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Adam Goryachev Sent: Tuesday, June 01, 2004 5:50 PM To: asterisk-users@lists.digiu...

I have a UIP200 on the way for eval. Does anyone have tips or tricks to get it working right away with * ? I hate having to go through the pain someone else braver than I went through already. :) Tim --

I ordered 10 7960's with SIP today (YAY!), I should have them on Monday! So, to be better prepared come Monday morning, I was wondering if anyone knew of any * compatible screen configs for things such as browsing VM, etc, yadda, yadda. I checked out the wiki about ADSI but from what I see, that's not really applicable in a SIP setup? I'm guessing it's going to be a more XML

The new voicemail features are great, cheers to those who worked on it! Un-related to this though (maybe?) When using the voicemail features (not happening in meetme or other apps that I can tell), I'm getting a lot of error messages in the CLI regarding IAX: Apr 28 14:05:45 WARNING[9226]: chan_iax2.c:5487 socket_read: Received mini frame before first full voice frame Sometimes this causes

I'm running Asterisk CVS-04/28/04-13:22:35 (fairly current) Today when I setup queues for the first time (with one member in my default queue), I got some really strange behaviour, aside from my hysterical laughing after hearing the default MOH =) I only have one SIP hardphone I'm testing with right now, so I tested using DIAX, Firefly(IAX) and XLite(SIP). My hardphone is an analog

I've got a strange issue where I get no ringing on outbound PRI calls using the TE410P span 1. The call actually goes out and works, you just hear no ringing. The quality and features on the call are pristine, no cracks pops or any weirdness like that. If I specify the 'r' option on the Dial() string, I get a half of a ring, sometimes a full ring as soon as the dial completes.

Hello all, One of my remote employees is using a 7960 we sent him, on a public IP address at his home office. I've run pings and traceroutes both from the server to his phone, and from the cable modem to our server, there's never a high ping time, or a dropped packet, however about every 30 minutes to an hour into his calls (not all of them, it's random) he can no longer hear the

Hello Everyone, My company is about to deploy * as replacement for our existing commercial Altigen PBX. Meanwhile, I've been trying to find the best cost effective SIP VoIP phone which we can use in office for 20-30 employees, as well as a few remote staff. Normally I wouldn't post about a VoIP phone, however, this phone was released less than a week so I thought I'd give some

Hello everyone, I'm using a TE410P, no irq sharing, and all extraneous devices disabled, such as USB, Parallel etc. I'm getting a few IRQ misses according to ZTTOOL. We're running a standard PRI_CPE interface and seem to be getting dropped calls, and errors on the D-CHANNEL occasionally. The circuit itself is very solid, it was in use on our old PBX just a few weeks ago, never

Hello all, I'm having a nightmare of a time trying to get stable results with SIP clients on Asterisk. I can't seem to find a configuration that works! In our office, we run a Sonicwall Pro 200, which is a sip aware, stateful firewall. Originally, I had configured Asterisk to run on the NAT side so that those within the office could connect easily, and those outside the office

Does anyone have any recent news on the Virbiage FT201 IAX Hardphone? I'd *really really* like to deploy these phones instead of SIP hardphones, and I can't help but wonder if I'm going to shoot myself in the foot (or another sensitive area) by deploying a ton of SIP phones just to find the IAX Hardphones were released a week later... Thanks, Brian D'Arcy

Hello Everyone, My company is about to deploy * as replacement for our existing commercial Altigen PBX. Meanwhile, I've been trying to find the best cost effective SIP VoIP phone which we can use in office for 20-30 employees, as well as a few remote staff. Normally I wouldn't post about a VoIP phone, however, this phone was released less than a week so I thought I'd give some

Hello, In setting up * for my company's office and remote employees, I have a question about how to log one username into * as either a SIP account, or IAX account. For example, we will be using SIP phones in the office locally to the * server, however some employees travel, and want to use IAX (as it's much friendlier with firewall/proxy setups than SIP) clients on their notebooks.

Shawn, That's a complete load of manure. I have an office full of 7960's, they work great with asterisk with the SIP images loaded. I'm about to pick up a lot of 7912's (simple one line phones, same as the 7905 but it has a built in switch). These phones have also been confirmed to work with Asterisk. I would recommend not going directly to cisco, and just find a reseller who

Hello all, I'm going to tackle learning C this week, and start writing my first * add-on/contribution; assuming it's actually worthy of contributing once it's done.. I think I've chosen a hefty project for my first go round here... I'd like to get some feedback from everyone on a FindMe/FollowMe spec I've put together. Before you read on, let me say, I don't want