similar to: ANI2 AKA Info Digits not supported?

A lot of people are going for the "VOIP only" approach, but SBC says you have to have an active analog voice circuit before they will sell you DSL. Does anybody know which DSL providers will sell you DSL without making you pay for a voice circuit? Thanks in advance, ------------------------------------------------------------------------ Steve Edwards sedwards@sedwards.com

I have Sipura 841's talking to a CVS-HEAD of August 4. If I disconnect the power to the Sipura, Asterisk does not hang up the channel. My sip.conf for this phone looks like: ; [super1] ; Sipura 841 disallow = all allow = ulaw callerid = "super1"

This is weird. If I have 2 members call into meetme using zap PRI channels on the box, they can here each other's keypresses. If I have 2 members call into a separate box using the same PRI's and then forward (dial(iax/...)) them to the previous box into the same meetme, they only hear a minor "squelch" for each other's keypresses. How can I completely mute a

I'm wondering if tweaking [rx|tx]gain would improve my fax reception success rate. Running ztmonitor when receiving a fax shows 4 "octos" and an * on the RX side and nothing on the TX side. At the end of the page, there's a burst where RX goes to about 1/2 and TX goes to about 2/3 of the range displayed. Any opinions? Thanks in advance,

Is there any way to transfer a call from host to host and keep the call's variables intact? -- specifically, UNIQUE_ID and user created variables like CARD_NUMBER, EXPIRATION_DATE, and CVV2? Thanks in advance, ------------------------------------------------------------------------ Steve Edwards sedwards@sedwards.com Voice: +1-760-468-3867 PST Newline

I'm trying to pass a variable between servers using "setvar" in iax.conf. I have a box (ts2) with a t100p in it. It answers the call and dials another box (ast0) via IAX. I want to pass a variable along with the call from ts2 to ast0. I'm running CVS-HEAD-03/07/05 on ts2 and ast0. ts2's iax.conf: [general] disallow = all allow

I'm trying to spy on an agent (Agent/54321). I can "dial(Agent/54321)" successfully. If I "chanspy(Agent/54321)" or "chanspy(Agent)" all I get is a series of beeps. Any clue where I should start looking? Thanks in advance, ------------------------------------------------------------------------ Steve Edwards sedwards@sedwards.com Voice:

My boss is complaining that the delay between speaking and hearing in a meetme conference is noticeable and doesn't want to roll out our system until I can eliminate the delay. Personally, I don't think the delay is significant, but I don't sign his check. The system consist of 3 1u's, each with a single quad t1 card. Each card has 2 t1's running NFAS. The "t1

I have what my SBC/ATT representative calls a "Supertrunk," but he can't tell me the specifics I need to know to configure Asterisk to work with it. By fiddling about, I've come up with the "almost working" configuration below. This works except that it takes about 4 seconds from when the console says the line is answered until it plays my prompt. Watching the

On Sat, Aug 29, 2009 at 10:30 PM, <asterisk-users-request at lists.digium.com>wrote: > Send asterisk-users mailing list submissions to > asterisk-users at lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject or body 'help'

is typing his number, though there is a 15 seconds timeout, and even if = I type the number very fast it still may happen to me.<o:p></o:p></p><p = class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><p class=3DMsoNormal>It has = been my casual observation that the speed at which I enter digits on my = phone is unrelated to the speed at which my

I'm trying to determine the capacity of my host running Asterisk 11.8.1 on CentOS 6.5. The host is an Intel E3-1240v3 with 8GB RAM, an SSD, and gigabit Ethernet. The primary application will be bridging groups of users using meetme(). I'm using 2 boxes -- 1 to initiate calls using call files (box1), and 1 behaving a bit more like a production box -- bridging calls (box2). The call

We i have settup it like this it giveme caller id agent id and date-time on gsm file but i want them to be in folder on every day basis datewise. exten => _1NXXNXXXXXX,2,Set(CALLFILENAME=${ACCOUNTCODE}-${EXTEN}-${TIMESTAMP}) exten => _1NXXNXXXXXX,3,Monitor(gsm,/rec/asterisk/apr07/${CALLFILENAME},mb) Any Idea ? Faisal > ------------------------------ > > Message: 16 >

"asterisk-users-request at lists.digium.com" <asterisk-users-request at lists.digium.com> wrote: Send asterisk-users mailing list submissions to asterisk-users at lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to

On Mon, 27 Apr 2015, Bryant Zimmerman wrote: > exten => _9XXXXXXX,n,Set(dialnumber=${l_HomeAreaCode}${EXTEN-1}) Missing a colon? ${EXTEN:-1} -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000

I've googled for way too long, where are the 2 letter language values defined? I know: en = English es = Spanish fr = French but what about Croatian, Russian, Serbian, Vulcan, etc? Is there a list documented for Asterisk or is it "just use the 2 letter country code Internet TLD?" Thanks in advance, ------------------------------------------------------------------------ Steve

I was debugging a turnup with Global Crossing the other day and they presented me with a web page that displayed a 'ladder diagram' of a call including a ton of detail all neatly organized in tabs and links so you could drill down to any level of detail needed. The copyright notice says 'Copyright? 2008 Empirix.' Is there any free software available to analyze a pcap or

On Fri, 26 Jun 2015, Dale Noll wrote: > I added a filter to the /etc/rsyslog.conf file > > :syslogtag, contains, "asterisk" stop > > Syslog is still receiving the messages, but is discarding them. Nice to learn a new (to me) feature of rsyslog. What does 'logger show channels' show? -- Thanks in advance,

According to http://kb.digium.com/entry/12/ The Iaxy will respond to pings on port 9999. You can ping your broadcast IP on your network and listen with tcpdump on your network on port 9999 which will show the Iaxy responding and what IP address it is coming from. Ex. ping 192.168.1.255 tcpdump -i eth0 udp port 9999" Before I get my karma whacked again, does this work for

I'm doing a new install for an old customer. The customer is running a custom version of Asterisk based on version 1.2.7.1. It works for them -- aside from a memory leak requiring a restart once every couple of months... I think the "corresponding" version of Zaptel is 1.2.5, but I'd like to run a bit more modern like Zaptel 1.2.27. Am I just asking for trouble? Thanks in