similar to: Re: Setting [rx/tx]gain for spandsp/fax

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"

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:

I'm not receiving ANI2 (info digits) on my SBC PRI's. SBC said they're sending them. I called Digium support and was told it is not supported. Is anybody receiving ANI2 on a PRI? Thanks in advance, ------------------------------------------------------------------------ Steve Edwards sedwards@sedwards.com Voice: +1-760-468-3867 PST Newline pagesteve@sedwards.com

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

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

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

Hey Guys! We had echo issue before so we replaced old PRI card with Sangoma A102D HWEC. Now my question is i set rx/txgain level 0.0 default do i need to touch this value or default is best. I have read on google and people say it should around 14844 on ztmonitor for rx/tx level same. I just use milliwatt and test my default 0.0 rx/tx level and it come around 4600. Do you think i need to make

Guys. I was going to give spandsp-0.0.3pre1 a try under asterisk 1.2beta1.. Anybody done this? When I noticed that this particular release doesn't have the tx and rx fax apps on the tree as older ones. Anybody knows what happened?

Hello! I'm having an echo problem with a TDM card. The TDM card is being fed by a channel bank just 12 or so feet away. When you put an analog handset on the line, both the RX and TX volume seem to be just fine. However, when I use the TDM card, I have to have an rxgain of 13.5, and even then, the audio is relatively quiet. I'm also getting echo on these lines, so I have turned

I've done gain tuning as per the info I've found online. I've got my RXGain set so my volumes list as about 14,800 (using a milliwatt test number and ztmonitor -vv). However listening to the line now, this sounds too loud to me. The person speaking sounds fine, but I've now got a large amount of background hiss coming thru. In order to get the "recommended"

I'm trying to put together a package of asterisk-head, spandsp, and app_rx,tx fax. I can get everything to compile: spandsp-0.0.2pre20 asterisk-head (cvs co -r HEAD asterisk) the app_rx/tx from soft-switch.org in the 1.1 folder However, asterisk complains that there is unused symbols when running /usr/sbin/asterisk -vvvvvvvgc ARGH.. Does someone have a package with files that I could try?

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

>Does IAXmodem allows you to receive faxes with any extensions >(auto-detecting incoming faxes). You just let Asterisk do the fax detection for you, and when it hears CNG, send it to the fax extension, and your fax extension would just Dial() one of the IAXmodems (using IAX) >>DRi@b-w-computer.de wrote: >> sure in an small office you can use iaxmodem/hylafax to receive faxes

Hello, I have spandsp setup to accept incoming faxes and receiving tif files via Email. Using tiff2pdf, or tiff2ps -a2 or even tiffsplit, the last page of the fax is cut off and the quality of the text looks "squished". I "figure" it's a tiff parsing thing, as opposed to a problem with my spandsp installation (heh). Has anyone experienced the same thing, or can

Hi Everyone, I want to fine tune the Rx and Tx gain on an analogue Sangoma card by dialing into another server that is running on Sangoma PRI card (both services on Bell network). [mwatt1004khz] exten => s,1,Answer exten => s,n,PlayTones(1004/1000) exten => s,n,Wait(300) If I match the Rx/Tx numbers on both sides by monitoring "ztmonitor X -vv" am I right with my theory of

is /usr/local/sbin/mailfax flagged to 755? ________________________________ Von: asterisk-users-bounces@lists.digium.com im Auftrag von Rob Danz Gesendet: Mi 13.07.2005 17:17 An: asterisk-users@lists.digium.com Betreff: [Asterisk-Users] SpanDSP rxfax, no tiff. Hello, Let me start by saying I have checked the wiki and the archives and did find some relative information. I tried the

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