similar to: ATA registration requests

I really should be doing something better on this beautiful weekend, but I'm trying to save myself some time for later projects by documenting some things that have been particularly troublesome in the past. That being said... I've written up a configuration guide for the Cisco ATA-186, which describes some of the features that are possible to set in the ATA and specifically what

I was looking for something to connect a couple of POTS handsets to my asterisk server and found this on ebay http://cgi.ebay.co.uk/ws/eBayISAPI.dll?ViewItem&category=162&item=5162868118 &rd=1 The documentation says that it does SIP - therefore will it work in an asterisk environment. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus

I was thinking about complicating my Voip setup by adding CME. I found this example here: http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Express+Integration and here: http://www.pasewaldt.com/cme/cme_index.htm Would anyone like to comment on their experiences using CME with Asterisk... I would like one of my Cisco phones to remain SIP connected directly to my Asterisk system. The

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

To whom it might concern, The Gesko Ikarus 1200S analog telephone has advanced callerid capabilities. When used with an ATA186, it show the username and the phonenumber of the caller. (or whatever you let * tell it) http://www.gesko.be/idgg004.htm Price is 77 euro something and available with Telec. (NL) Met vriendelijke groet, Pauline Middelink -- GPG Key fingerprint = 2D5B 87A7

I have an Asterisk server that I use at work. I have a phone that is at home that logs into the Asterisk server at work. My home phone is hooked up via DSL through a Linksys router. You can see the my sip.conf for the phone blow. The problem is each time the phone rings I can hear/be heard 50% of the time. Any suggestion on what to look for. I do have my reg time set for 180 seconds on the

I have an ATA186, a tech just told me to set UID0 and UID1 to the same username, and PASS0 and PASS1 to the same password. In my mind, this would seem to have the unit registering twice under the same account, which Asterisk wouldn't support. When a call comes in, it should go to the last line to register. So to me, this means the call could sometimes come in on Line 1 and sometimes on Line

I can buy a new ATA186 here, but it is sold with a 1-port user license UK, for euro 192, but does that license stop me from using both ports? I can't read the license agreement till I buy the thing, so I don't know what i'm buying... Michiel --

The details for the Australian cisco ATA186 are below: > -----Original Message----- > From: Tony Du [mailto:tony.du@action.com.au] > Sent: Tuesday, 8 July 2003 4:31 PM > To: 'Adam Goryachev' > Subject: RE: [Asterisk-Users] ATA 186 in Australia > > Hi Adam, > > I sold a Cisco ATA186 I1 2 port adaptor (Cisco code: > SW-SMH-UL-ATA-2P)to you on 16/10/02) >

Does anybody know of an ATA that works well with * and supports T.38 fax relay? The Cisco ATA186 does not support T.38 and I don't think the Sipura does, but I don't know for sure. I'm not talking about running fax over an ATA with g.711, I want actual support for T.38. Thanks. Peder

Hey all! I'm having problems trying to set up an ATA 186 with my Asterisk box. When I get the phone to place the call, I type the extension and I only get busy signal after 5 seconds. So I can't call my Asterisk box from my ATA and either call from my Asterisk to my ATA. Does anybody know what can be happing? Log is attached.. tks regards Oz -------------- next part -------------- >

Hi all, do you know any firmware release which fixes that issue for cisco ATA186? ATA 186 3.x.x Cisco ATA 186 v3.* CANCEL requests can be sent with a completely bogus URI, making it impossible to cancel a call. bug no workarounds This is from: http://www.communigate.com/SIP/SIPProblems.html. In other words, you make a call, other side doesn't pickup, then you hang up,

For some time now some have been working on a winIAX client, plus others.... Now basically as I have mentioned before (& no one has correct it) there are 4 types of clients.... 1) Softphones, eg msn, xten etc using all computer hardware. 2) Softphones plus hardware, eg: a usb type adapter (mind you there are some things not common available with these yet. 3) Hardphones, basically your

John Todd wrote.... > For those of you wanting to salvage your Cisco ATA-186 after > inadvertent locking, or after recovering your devices from a vendor > who has locked them, here is a rainy-day project for you: > somedoc.pdf..... Immediately strides to ATA, rips off cover... woohoo, EEPROM is socketed.... well maybe I'll just copy the contents of a working ATA into the

I've done something similar at home, but made my dialplan such that I can dial either 10 or 11 digits locally. I don't use a "throw away" digit at all. Any 7, 10, or 11 digit call will be appropriately mangled and sent out the PSTN / VoIP provider. ________________________________ From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On

hi i need many cheaps atas or some very cheap way to connect analogs phones to asterisk what do you recomend? i searches and only find solutions like 40 U$D (in the states, here in argentina is like 80 U$D) per phone any links or something? thanks! David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination.

Hi Everyone, Does anyone know of any ATAs that can do proxy failover without using SRV. I don't want to rely on dns if at all possible. Basically, I have Asterisk boxes in two different data centers and I need ATAs to be able to uses the server at DC2 if DC1 goes down. The servers are already in a HA setup at each datacenter. I am looking for added protection if one of the datacenters

Hello all ! Can anyone recommend me ATA device that REALLY has T.38 built in. So far I have heard of Telco Systems Access201, which seems to be impossible to bye in Europe (all resselers are droped Telco systems ATAs for some reason (tried in Germany and in UK so far)), and I have heard that SIPURA SPA-2100 should have T.38 built in into newer firmware, but I wasn't able to confirm that

Hi, I'm looking for a few pointers on using ATA to connect Asterisk to the PSTN. Basically, I'm running a Hosted PBX service, and in urban centers I can usually get SIP or PRIs. Since I sell my customers SIP hardphones, the data flow is like this: Customer's SIP Hardphone ---- My own Asterisk ----- Outside lines But when it comes to smaller villages (I deal with people in tiny

Hi. Does anyone know about a simple one-fxs ATA with pulse dialing support that can work with Asterisk? A SIP one would be ok. I've been told that the Digium S101i IAXy does support pulse dialing; although it's a iax2-only ata it could be enough. I need a bunch of them to convert some old fashioned rotary phones into VoIP ones (I'd like to disassemble the ATAs to remove the boards