similar to: Asterisk iaxy adapter annoying beep during conversation

I just bought an IAXy and have been using it with * and NuFone. Around once every 24-48 hours of operation (not off hook, just powered up), when I pick up the phone I hear loud static instead of dialtone. If there's an incoming call, the phone will ring and * CLI will show that it's trying iaxy, but when picking up--no answer to * and loud static. The incoming call goes to VM. Cycling

In features.conf, I have uncommented the transfer features under feature map, but I still cannot transfer using a POTS phone on an IAXy adapter. I think I am missing something here.... Any help is appreciated. Here is features.conf: ; ; Sample Parking configuration ; [general] parkext => 700 ; What extension to dial to park parkpos => 701-720 ;

Hi, I have three Iaxy devices (s101i) parts. Two of them seem to work fine. The third plays a loud repeating click sound when an analog phone is plugged in. I can provision all of them, and make calls to all of them. The clicking one will blink when a call is incoming, but no audio from the call can be heard on the handset, and the caller only hears silence. The same handset works on the

I have only been able to confirm this issue with my IAXy as it is the only ATA I have. I am running 1.2.5 stable. Example: User @ exten 100 (iaxy) receives call from PSTN (call 1). While on the call, another user from PSTN (call 2) calls 100 (iaxy) sending a call waiting beep to the iaxy. User 100 answers the call from PSTN call 2 putting PSTN call 1 on hold with music. While call 1 is on

Does anybody know what DTMF coding does S101i adapter using? I've been testing one for over a week and here are my observations: - DTMF signaling is working perfectly with Asterisk, much better than Sipura 3K Though, I think the Asterisk "iaxy" firmware is buggy, the unit is using auto-update feature; so I have Asterisk 1.2.13 and iaxy firmware version is: 23 When enable in

We've had a few people complain that the "beep" before leaving a voicemail is not loud enough and too short. Does anybody have a recorded beep that they can share, that is a little louder and a little longer? We've had this box in production for 2+ years, so I hate to mess with the gain on the PRI or anything like that because everything else works fine. I know nothing

Is there anyway to turn off the call waiting beep in the grandstream and/or cisco ata186? I have a dial statement in my extensions.conf that rings 5 phones at the same time by combining them with the & in the dial statement. i.e.) exten => blah,blah,Dial(SIP/GS1&SIP/GS2&SIP/GS3&SIP/ata186a&SIP/ata186b,25,t) If one of those lines is being used, then the user gets a really

Digium has discontinued their ATA iaxy adapter; don't blame them, too expensive so they can not compete. The adapter is upgraded automaticaly when it is connected to new asterisk version; since this adapter is discontinued will it still work with asterisk 1.6 and beyond or will it be\ just a "door stopper"? -- Joseph

Hi All, I have two old S100 units (the blue ones, not the newer black ones). I am trying to reset these to factory default using the following instructions, but it is not working. Does anyone have any other suggestions to reset this model of the adapter? Tried this: 1. Remove all of the cables, except for the power cable. 2. The factory reset button is next to the RJ45 jack. Press and hold the

Hi, hoping that experienced hands will quickly show me the right way: after a fruitless web search i am turning to this list with my rather elementary question: is there any other way to communicate with the IAXy besides using special utility software that needs to be compiled under UNIX? Here is the story: about two months ago, after some not very satisfactory attempts at using SIP (my phone

Does anyone have a more clear beep tone for the voicemail? The default one seems to cut off and give the feeling that there is a problem with the vm :) If no one has this available, I may try to create a new one. Thanks. -- respectfully, Joseph - ------=============

Hi, I've provisioned an IAXy adapter on a network segment local to my asterisk server. Provisioning is fine, as is the registration and use of said device. Since the local address is private address space, I setup the public IP address of my Asterisk server as the alternate. When taking it to another location (NAT again behind a WRT54G), it doesn't seem to work. Here's my current

From what I have found the IAXy doesn't handle echo very well. About half of the analog phones I try on the adapter create an echo on the far end. The person I am talking to can hear themselves. I am using Asterisk 1.4 and have tried it with 1.2 as well with the same results. Is there is anything I can do in Asterisk to help solve the echo problem? Thanks, Sean Dennis

Anyone using this 110-220v 9v 1500ma supply? If so, which DC plug adapter does the IAXy need? A friend brought the wall wart over here for me but she said you get one plug free and she didn't know which one it took. Does anyone here know? http://www.radioshack.com/category.asp?catalog%5Fname=CTLG&category%5Fname=CTLG%5F009%5F001%5F001%5F003&Page=1

Does anybody know why Digium S100I IAXy was discontinued? -- #Joseph

Hi everyone, I'm using the IAXy boxes and i'm having some trouble when I use it with the ADPCM codec. When I use the ADPCM codec only one person (out of the two of the conversation) is able to hear the other, but when I switch to ULAW codec everybody can hear the other. The ULAW codec is too heavy for my bandwidth (64Kbits/s) and its sounds choppy, the ADPCM codec sounds good but only

When I call my internal extension and hang up the phone keep ringing, I get: SIP/11-00000048 connected line has changed. Saving it until answer for IAX2/iaxy Is there a solution for it? SIP does not detect that IAX has hang up the line. -- Joseph

I'm having some problems with a bunch of older IAXy devices. I have three S100I devices working. I can call each device from each other and everything works perfectly. When an inbound call comes in (from PSTN) and is routed to the device, it works perfectly as well. The problem is that when I place an outbound call (to PSTN), the remote party can hear me well, but I cannot hear the

John Novack wrote: >And the only IAX2 box made is the Digium one, with it's current shortcomings ? >From reading through the archives, it seems there is currently no way to reset to factory default, no >written MAC address on an individual box, and some other instabilities requiring frequent resets. Yeah, there's the rub. Dunno if it's worth it, I'm willing to give it

Hi everyone, I'm using the IAXy boxes and i'm having some trouble when I use it with the ADPCM codec. When I use the ADPCM codec only one person (out of the two of the conversation) is able to hear the other, but when I switch to ULAW codec everybody can hear the other. The ULAW codec is too heavy for my bandwidth (64Kbits/s) and its sounds choppy, the ADPCM codec sounds good but only