similar to: Asterisk, ATA186 and callerid

Hi I have a brand new ATA186 with the following firmware: Version: v2.15.ms ata186 (Build 020919a) I have been through the archives about how to configure it, but my colorful configuration web page does not have the same fields that people say I need to adjust. Even the examples on Cisco's web site don;t match. For example, I don't have the GtkOrProxy field, which is an important

I am trying to do a "factory reset" of an ATA186 using the widely-available instructions (basically dialing "FACTRESET#" on the keypad while at the menu prompt). I have done this a number of times before with success, but on this unit the lady spells out "P A S S W D" when I finish up the entry. Does anyone know what to do next? Hitting the star key (which is

I dont have a cisco acount yet can some bady hel me with the ata18x-v2-16-030401a-1.zip file. thanks in advance Rodney Acosta Coya. Dpto. Tecnologia de la Informacion. racosta@moanickel.com.cu Tel:(53)(24) 62 611 -----Mensaje original----- De: Paul Rodan [mailto:asterisk@glitch.cc] Enviado el: Martes, 23 de Noviembre de 2004 11:24 a.m. Para: 'Asterisk Users Mailing List - Non-Commercial

Hi guys, sorry to be iterating this on the list once more, but I'm not able to get this stuff to work as I'd expect. So far, I've always managed to keep it out of NAT environments :-> My home LAN is NATed by a simple Draytek router. In the home LAN is an ATA186 with SIP. On the internet (public) is an Asterisk server. I have nat=yes in the sip.conf and the connectmode is set

Hi all, I have a Cisco ATA186 connected to an Asterisk Server (SIP) The dialplan uses 1XX for local extensions and XXXXXXX for external numbers, where the first digit is always different than 1. In this moment, when I dial 123 for example, ATA waits till timeout before dialing that number. The same for the longer one. How can I do to make it dial imediately when 3 digits starting with 1 are

I've installed Asterisk and configured an ATA-186 as described at this link: http://www.djernes.org/~shawn/ata186.htm Unfortunately this guide abruptly ends before it explains how to deal with the sip.conf and extensions.conf files. So I left extensions.conf alone and my sip.conf looks like this: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0

I noticed a few issues with app_queue just wanted to know if its sip related or ata186 related: Ext 111 and Ext 112 are dynamically loged into the queue via AddQueueMember. Call hits queue with fewestcalls routing. Rings ext 111 if 111 doesn't answer. It rings ext 112. If for some reason ext 112 doesn't answer it rings back to 111. Again at this point ext 111 isn't answered it

Hi All, I'm looking at getting some Cisco VoIP hardware to play with in combination with a Asterisk server. I've heard that there is beta software available to do SIP on the 7905G. So, I'm thinking of either getting a 7905G or a ATA186. My dillema is, which one to buy? I can get both for about the same price, has anyone had any experience with using a 7905G with Asterisk? On

Hi, I'm away at a conference in Amsterdam. My home is in Cambridge in the UK. On a whim, I tossed an ATA186 and a phone into my bags before leaving home. I was able to plug my ATA186 into a LAN here at the conference and was connected to my home Asterisk in a few seconds. Total time from unzipping my bag to talking to home no more than 15 seconds. OK, so the kit could be more portable,

Hi, I saw your posting about the cisco ata186 I2 vs I1 and the simple vs complex impedance. I ordered a cisco ata186 i2 for use in Canada by mistake, didn't know that I needed the I1 version. Will the I2 version work in Canada with regular anlog phones, or will I need to change it. Thanks for your answer. Samy -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all: I'm testing a new installation of *, bringing up some ATA186. In * environment, all stuff works greats. The only thing that don't work is a Call Transfer, but the 3Party works ok. Some time ago I read that somebody had proven this functionality successfully. If somebody knows what I missing, please let me know. Thanks in advance, Gus -------------- next part -------------- An

Guess who's here to do an Asterisk demo this week without the power supply for his SPA-841. I have an ATA186 with me. Both phones use a 5v supply. Does anyone know whether the supplies are interchangeable? Thanks in advance; sorry for the noise. B.

I have a Vonage acct that uses the Cisco ATA186. Currently, I have the ATA186 plugged into a SPA3000 to act as the FXO port. I installed a X100P card with the idea of replacing the SPA3000. Now, when I plug in the ATA186 into the X100P card and make a call into the system (from cell phone) and hangup when the IVR is playing, Asterisk is not detecting a hangup and keeps looping the IVR. If

Hello: I have problems sending dtmf signal to an ATA186 my configuration is: ATA186 --> asterisk --> ATA186 --> FXS to FXO Converter --> PSTN The ATA186 are set to send dtmf RFC2833, but it seems that the ATA186 can't generate dtmf so I can dial to a PSTN number. Is there a setting that can fix my problem, inband dtmf does not work because I'm using G729 codec Thanks

I have done my homework on this, I hope. I have a customer with an ATA186 who uses Nufone as his IAX provider. His network operations center in the Bahamas was destroyed by the hurricanes, and I'm helping him rebuild. We have a nagging problem getting his ATAs (located in public IP space) to talk through his IAX provider (Nufone) to the outside world. As far as we know, things worked OK

Hi, I'm still hassling with the consultative/attended transfer stuff. Someone please help me identify this A lot has already been said about the ATA186. Some report it works fine, others say it doesn't. Lets get clarity on this. My scenario is reasonably simple (I think) Phone A: SIP/video1 Phone B: SIP/werkkamer Phone C: IAX2/provider Phone A calls phone B, they chat: *CLI> show

This deal has got me confused. My dial plan rings my ATA186 on all incoming calls. If I don't pick up, it goes to voicemail. Under either of those circumstances, the callerid screen on my phone stays blank, and the message waiting indicator does NOT come on. But anytime a call comes in for me while I'm already talking on the phone, BOTH of those things happen. . . So what do I

I have Panasonic TD1232 pbx, few cisco ata186 and linux box with asterisk. Can I dial from asterisk into ata, then indicate phone number playing tone (use DISA feature at panasonic) and connect to any analog phone connected to panasonic ? I think some of Playtones application within Dial application can help me. But I don't know how. -- Pavel Zheltouhov, Comlink ISP, Voronezh, Russia

All, I was wondering if anyone has any experience with the Cisco ATA186 (SIP image) and outbound faxing with Asterisk. Inbound faxs from PSTN into * and on to the ATA work fine but outbound faxs receive congestion from *. I've got packet dumps from both sides and everything appears normal but after about 3 seconds the * servers sends the AS5300 a CANCEL and sends the ATA a '503 Service

Hello list: I'm not sure what is going on, but I am using: asterisk -stable cvs (november cvs download) SIP channel Cisco ATA186 Zaptel 4-port PRI for PSTN Caller ID is enabled on the cisco ATA and seems to work fine. We do not get caller name at this time over the PRI ... Caller name works fine VoIP to VoIP and for PSTN calls, it displays the calling number like caller ID. But for some