similar to: asterisk +heartbeat

A little off topic but.... I need to put a 24 port Gig PoE switch into a small office - no computer room / rack etc. All CAT5 terminates near the owners desk (smart huh?). I want to put a PoE switch in place, with 24 ports and Gig speed. Everyone I've researched so far is LOUD... Anyone know of a quiet one? Thanks -------------- next part -------------- An HTML attachment

Hello Considering how cheap PCI modems are compared to even the cheapest PCI hardware from Digium, OpenVox, Sangoma, etc.... I was wondering why Zaptel can't be used with those to connect an Asterisk server to a POTS line for low-level use? It just seems overkill for SOHO users who only get a few calls a day. Is it because those modems are usually Windows-specific, and it would be too

> >> having two NICs on the same subnet > > I'm trying to wrap my brain around that in the larger network > picture. Two > NICs in the same subnet (presumably on the same computer) would have > access > to the same other devices. This could potentially increase bandwidth > (maybe?) and offer redundancy (if NICS, wiring or switches were the > biggest >

Subsequent to some previous E-Mails, I've been trying to dig into the ISDN - BRI situation a bit more. I have determined that I have a HFC card with Winbond chip, but I'm not sure what combination of drivers is best or usable. zaphfc is out because it only supports the cologne chip. misdn is a possibility. I haven't determined if it supports the card natively, or needs a card

Hello! I'm still wondering, how to park a call with an ISDN line. The setup is the asterisk server only, controlled via the CLI. I can originate a call and I can tell asterisk to start the JACK application. But I can't then park the call. I tried it with sending DTMFs with misdn send digit, no luck. I had a look at the CLI, but didn't stumble upon a command to park the call.

> I have a network that sporadically give 'network busy' errors. One morning > the workstations will all connect up fine and stay up throughout the days > work, yet the next day there is not one machine that can log on to the > network. It may run fine for two weeks, then bam, three days of no logins. > Yet the internet works fine. Could I copy my smb.conf file and rpm -e

Hello people, I am having difficulties figuring out the POE power draw in watts from a Polycom IP601. I want to know how many IP601's can be powered from the Cisco Catalyst 3560G-48PS. The IP601 wallwart has: Input 120VAC 60hz 19W, Output 24VDC 500mA. I assume the output is appropriate value to figure out how many phones can be powered. The Cisco 3560 datasheet says "the 48-port PoE

Hi, I'm in the process of setting up Asterisk in a SOHO environment using ISDN for trunking. More specifically a BRI 2B+D circuit where one SPID is used for the business and the other is used for personal. The circuit already exists, but is presently being interfaced to POTS phones via a TA. This configuration is not very common in the US, but we are fortunate that our LEC offers it price

Hi all, I've done this a few times with other PBX's but swyx has stumped me! I'm having some trouble getting Asterisk connected to a Swyx system using a sangoma A104dx... currently the setup is: BT <-> Swyx The above setup works fine... what i'm trying to achieve is BT & SIP Trunks <-> Asterisk <-> Swyx I have connected to our BT (2 x ISDN30 UK) with

Hello, I'm trying to set the 3rd span of a new digium quad card as a E&M T1 for Faxes to a Hylafax server. The 1st and 2nd spans are working as PRIs. When I start asterisk, the logs show a signaling error and chan_zap.c dies. I also get an error that it can't read the gains but they are the standard shown below. 2.6 kernel, Debian Stable, * 1.2 svn from feb 2007 my procedure: make

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

Hello, MOH always is choppy when someone calls from a cell phone to my pots or nufone 866. It sounds fine when it originates from a land line. I use zaptel hardware, and plenty of resources. I have tried to use different songs. None have the id3 tags, I tried the custom settings with -q -r 8000 -f 8192 -b 2048 --mono -s. Tried permanent resampling to 8khz, 16bit, filterd with lame -q1. I

Hello, I have been getting outbound nufone calls dropped after about 70 seconds. CLI shows "Attempting native bridge of IAX2". I have put "notransfer=yes" in iax.conf in the main section and all identifier sections. I tried a Tt in the dialstring, but it still tries the bridge. a cvs update didn't help. internal *server1 tdm fxs pci card <--> iax2 trunk -

Hi all Been messing about with the single port cards for a number of years, but never got good results, I was thinking of giving them another go over Christmas and was wondering if anyone would share there recent experience, as to which driver works best MISDN BRISTUFF etc with the latest version of asterisk that supports Zaptel , I'll probably have a TDM400 card in the same box Thanks

I'm looking for a solution to determine if a PSTN call to a zaptel channel was originated from a VoIP provider or not in real time. I'd like to use the callerid(num) to reverse match to the provider. Does anyone have a clue how I could do this? TIA Bart -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello there fellow R-users, I am using the RODBC functionality to query a database. I am trying to read in a columns of strings which have a character field lengths greater than 255. The data.frame that I recieve back from the RODBC query only contains the first 255 characters (the rest having been truncated). Any help on how to solve this problem would be greatly appreciated. Reagrds Wayne

In Asterisk 1.6, there is an option to use the 'new g.711 algorithm'. "Use the NEW ulaw/alaw codec's (slower, but cleaner)" By slower does this mean more 'expensive', or does it instead mean that there will be more algorithmic latency? Both? Can anyone speak to the relative increases? With regard to accuracy, can anyone speak to what kind of situation might

Hi, I am looking for a good ATA recommendation, ideally something: 1) with one FXS and one LAN port (so it's as inexpensive as possible) 2) That can be provisioning using FTP (configuration and firmware upon reboot, ideally remote reboot from a sip notify) 3) Supports T.38 Nice to have would be: a) PoE powered and AC powered (my choice) b) Small size-wise I have been

New to Aserisk 1.6 and find the 'installation tutorials' seem low to non existent. You go to the main Asterisk page (digium.org) and really just old install instructions for 1.2 are in the examples. Download links only give you asterisk itself and not dahdi or libpri which also are needed to run asterisk? It's very confusing to anyone who is new. Someone take notice! we need a link

Hello. I am looking for details of the maximum allowed/usable/effective wire/cable length of the connection from a FXS port of Digium analog cards to the analog telephone handset. To clarify my intention, I need to have an analog telephone connection to my asterisk box that is 3000 meters (3km) away at least. If you have any details of ATA boxes or other similar devices that I could use to