similar to: Poor analog line quality, wireless "base station", FAX-ing

Getting "no service" display on aastra 480i. Sip debug shows an "unathorized" blub when the aastra tries to register. Some reading indicates that 1.4 firmware wants aastra.cfg and mac.cfg in /tftpboot/. There are none. Anyone have basic config files? Or can point me to a good link? All links I have tried, that purport to have config files, are either dead or error out.

Since the beginning (of my Asterisk life) I have an install that is, supposedly, set up for call waiting. Using a TDM400p, with FXO and FXS modules. On the Analog phones, I can hear the Incoming call (call waiting) tone, but the system does not respond to a "hook flash", to place the current call on hold and answer the incoming call. I have not attempted, nor research how/if this can

Any ideas on how asterisk could accept an email (such as an email to SMS or "number at mybox.org" sort of thing) and make a phone call to a specific number and make an announcement? I imagine the first part is the big question. joe a.

This is a follow up to an earlier post. Looking for a means to "individualize" incoming FAX, so as to distribute them to the intended recipient. While the PBX is based on Asterisk, it is not possible for me to "enter" the box to modify things, to any great degree. I thank those who mentioned IAXMODEM, earlier, but that seems a no go. Currently, there is a dedicated T1 into

I have an existing Hylafax system using a mainpine 4 port board, 4 POTS lines. Have a recently installed Asterisk system, with a dedicated T1 line. (Well, it's really a fonality system). What would I need to do, or where is the reading material, for what I need to do, to convert the Hylafax server to use the T1 line? Reliably. Preferably to use DID's as well. The current FAX works

My Polycom phones are displaying time, off by one hour. Seems they are on the old DST rules. How do I fix this? joe a.

I am having a problem detecting incoming FAX. TMD22p (tdm400p 2 fxo, 2fxs) As I understand it, I must have faxdetect = incoming to enable detection of the fax tone. Then, I must have a [fax] context to pickup the line and send it to whatever extension the FAX device is on. In my case, I ask it to answer immediately and do a distinctive ring (r3) to alert that is its a FAX call so no one picks

Having had various issues with local vendor (begins with "V"). am looking to move to all wireless. Anyone know if current vendor can refuse to port the current land line numbers to a wireless provider? >From what I've read, the Fed's seem to say "no", they cannot refuse, or impede this. joe a.

Excuse me if I recently posted on this, but I cannot find it, in my, or the list archives. Is it possible, when transferring a call, to set the user ID to that of the "outgoing number" instead of the incoming number? I believe the answer is (was) yes. New twist, does it matter what the destination "media" is? Meaning, the call would be coming in on a T1, going out on a T1,

Any great disadvantage to using polycom 300/500/600 vs the 301/501/601? joe a.

Is there more than one version of the Cisco 7960? I see some items advertised as 7960 or 7960G, but searching on 7960 only brings up 7960G info, or ambiguous stuff. joe a.

http://www.wired.com/print/politics/security/news/2007/08/wiretap

Experiencing one way audio using IAX2. I did see some other posts on this, and see there may be some internal issues with asterisk and one way audio. Can this be a widespread problem? So many seem to be using IAX, I find it puzzling. Some information points to this being a problem on asymmetrical connections. This is a decidedly asymmetrical connection, with 1.5 Mbs download and 256 kbs,

This is, hopefully, just a case of brain fade. With zapata.conf and zaptel.conf in place, asterisk loaded, no dial plan and all LEDS on the card lit, I get no dial tone, plugging an analog phone into ports 1 or 2, only a buzz and click. zaptel.conf - defaultzone=us loadzone=us fxoks=1,2 fxsks=3,4 zapata.conf [channels] signalling=fxo_ks language=us context=phones-1 group=0

I have a setup with a polycom 601 and an act p160s. All on local segment, no NAT. Can call the act p160s, from the polycom, rings, connects, and a conversation can take place. The reverse is not true, Dialing from the act to the polycom does not work. SIP debug shows, at the end, "Incoming call: got sip response 416 "unsupported URI Scheme" back from 192.168.0.xxx. Which is

Was messing with a polycom 501 and changed the IP from dhcp to static. Working with a user remotely. Now, the user says the phone does not show anything on the LCD and does not respond to any buttons. When rebooting, there is text shown as it proceeds. ?? Is there a way to reset this to a default? Does not respond to ping on the address we set. joe a.

Is Zaptel no longer available? I returned to a long shelved project (using TDM400P and a customized, canned version of *) and, getting to the configuration, find wctdm is not there. I recall the authors where very "enterprise" oriented and focused on T1 cards. So they left analog support out. Anyway, before I abandon all hope and dive into the "new stuff", I thought I would

Try this subject line if you will. On 11/14/06, joe a. <joea@j4computers.com> wrote: > > Did not know how to make up a subject line for this. > > I have a dial plan that allows a caller can try my cell phone. And that's > fine. If the call cannot be made, it sends caller back to voice menu. > > However, I'd like a way for the caller to elect to go back to the

Hello, I'm trying to use the package ttda, wich is involved in text analysis, for my own data about answers in a company survey. I've installed it, as well as ispell, but when trying to use an example: > zz <- file("stupid.txt", "w") # build a data file > cat("{comment - stupid data file} \n" , file = zz) > cat("<uci=1>

Hi guys, I'm looking for a affordable conference phone and a wifi phone that has a cradle. Polycom seems to make pretty nice conf phones but the price is a bit crazy for us. I saw the recommendation with ATA plus an analog Polycom phone but I do prefer a SIP phone. All because it's just too difficult to pull a phone cable into the current conference room. Is there any cheaper SIP