similar to: rx/txgain per device?

Hi all, I'm getting some noise gate like effects on our sip lines & I think I need to disable silence supression, I'm searching docs & not finding where this can be set, does * have a setting to turn this off? basically what's happening is when we stop talking, the other end hears total silence, but when we talk, they can hear the background noise in the office, this sounds odd

I'm trying to get a Zultys Zip 2 phone working with Asterisk. The phone seems to be failing registration (see sip debug output below). However, I can place calls TO the Zip2 from other SIP phones (Grandstream BT-101, Xten X-Lite, and eStara Softphone) and from Nortel PBX extensions coming in to Asterisk over a PRI T1. The problem is that I cannot dial any extensions from the Zip 2. Any

I am trying to get a new Asterisk installation running using a Zultys ZIP2 phone and a Uniden 200 phone. I have the system working reasonably well (although probably not optimal) except for a couple of items. First, I can't get the voice mail message light to work on the Zultys phone but it works just fine on the Uniden phone. Second, the time presented on the Uniden UIP200 phone is 1 hour

Hello All, I'm having trouble getting a Zultys ZIP2 to work with Asterisk, along with some other troubles in general. I keep getting a "Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from x.x.x.x). Even when Asterisk reports that the ZIP2 registered correctly, I can't make any calls out from the phone, or calls into the phone. Occaisionally I get a

I've google for hours trying to find a discussion of a similar problem as the one I'm having, so forgive me if this has come up before. If it has, please point me in the right direction! The problem occurs when a caller (A) is transferred by an intermediary party (B) to voicemail (Voicemail or VoicemailMain), either directly or by being taken to voicemail when the callee (C) doesn't

I would like to configure my Zip2 phones via tftp, however the tokens in the config file are (apparently) not all documented. Specifically, the username/password/callerid fields seem to be only configurable via the web interface. I find this hard to believe, but the documentation and examples that Zultys provides don't help. If you have an example of a tftp-loadable config file for the Zip2

Hi everyone, I just installed asterisk on my system with the purpose of rerouting calls on sip channels. I don't think i need any hardware for that. I am using LIPZ4(zultys) and sjphone as softphones. I tried setting up both of them and to call one from the other on the same machine, however could not. I 1-) I could connect sjphone in isolation to freeworld dialup howver i got no sounds

Thanks for taking a look greg and hank. This seems to be getting bettre everyday..help please My sjphone is running on the same box as asterisk...i believe then the red hat firewall should not be a problem. Whenever i dial from CLI i get ######### Executing Goto("OSS/dsp", "default|s|1") in new stack -- Goto (default,s,1) -- Executing Wait("OSS/dsp",

Hi all, Does anyone know of a method of sending a raw G711 stream to an address in Asterisk. For example, an application that takes a argument of a phone and a port. The reason? I have found a method to paging on Zultys ZIP2 and ZIP4x4 handsets. Basically it involves sending a stream of RTP data to port 3771 to multicast address 224.0.0.1. Would it need to involve me writing my

I have purchased a single G729 license - however, how many are actually needed? All my IP phones have G729a codecs built in (Cisco 7960 / Zultys ZIP2) - I would have assumed that if the phones can do it, and canreinvite=yes, then the phones shouldn't need to go through asterisk anyway? For calls that do go through asterisk, is a single license required for each side of the stream? (i.e. a

Do bristuffed zaptel (zaphfc) supporting rxgain/txgain in zapata.conf? I'm changing rxgain in zapata.conf, and reloading zaptel, but sound level on ISDN(HFC) is always the same (loud).

Hey all again, I'm wrestling with echo problems on our sip extensions. I've set these items in zapata.conf but tweaking these values doesn't seem to make much difference echocancel=yes echocancelwhenbridged=yes echotraining=2500 rxgain=8.0 txgain=1.0 are there other settings that can help me tame this beast? Been searching but not turning up anything that'll work here. Thanks

Hi again, I'm trying to setup our non pri T1 (they call it a Long Distance T1), our current pbx has the signaling set to E&M, I can set em in zapata.conf, but I'm trying to track down the proper entries for the zaptel.conf file. The digium docs only show a PRI example. Our current system has these settings: Signalling: E&M Framing mode: ESF Line Coding: B8SZ here's my

I just got a flyer from my buddy on these phones today, totally SIP based, includes the G.729 speech compression codec. http://dm.zipphones.com/dm/zip2/index.htm Any word on these? -- Mark Street, D.C. Red Hat Certified Engineer Cert# 807302251406074 -- Key fingerprint = 3949 39E4 6317 7C3C 023E 2B1F 6FB3 06E7 D109 56C0 GPG key http://www.streetchiro.com/pubkey.asc

Hi All, I have set following milliwatt and dial in from my mobile phone. and i adjust txgain value in chan_dahdi.conf but no effect on dahdi_monitor value is same. even i change its to 20 and more but monitor still showing same value. But i can hear tone variation on my mobile when i am changing txgain... is there any specific value i can set in txgain or leave it default 0.0 ( # = Audio Level

Hi, Do you have some experience with the "best" values for those parameters in youyr particular case? I mean the best raport between sound level in both direction and echo cancellation. For me, the best result I can get is with: rxgain=10 txgain=15 ... the sound level is good, but the echo is a little bit to strong for my taste. Something interesting is that if I put a txgain value

Is there an option txgain for SIP in Asterisk? My users all complain that their other parties think that they are way too silent even though they all have their mic volume all the way up and also enabled the 'mic boost' option. This happens with all the clients that we're using and also with different model headsets, so my last hope is txgain for SIP. Thanks Christoph

In trying to get the asterisk and faxing working I had to resolve to using iaxmodem and hylafax. I have incoming working, but outgoing the other fax rings but it would appear from web searches that the fax signals are too low to be "heard" I can read about rxgain and txgain for zapata. my fax setup goes direct from aterisk <-SIP-> SIP Provider <- Fax Machine -> It never

Hi, does anyone have test numbers in Germany that would allow me to tune my rxgain/txgain settings? I know there are numbers provided by other providers in UK e.g. but have yet failed to find a number in Germany (esp. by Deutsche Telekom). Kind regards, JP

Anyone gotten two of the 2572 echo canceller cards to work in a 253c mounting assembly? I can get one to work, but when I install two, one always fails. I've tried all my cards solo in the enclosure, on each side, and they all work properly when only 1 is installed, however, when I install two, one of them will come up, but the other always fails. Anyone know what might be causing this?