similar to: Cisco ATA-186 Firmware upgrade

I am trying to set up the Message Waiting Indicator (stutter tone/light) so that my cisco ata-186 will let my phones know there is a message waiting. However this does not seem to be very well documented. I found this on wiki mailboxnumber@context ... where does that go? Do I put it in my SIP.conf definition for my cisco ata, or where. In my SIP cisco definition i already have a

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

I use both ports on my cisco ata-186. I run them using ulaw. Today I made numerous calls using my analog phone on port 2. I picked it up about an hour after the last call I made and the line was dead. There is no power at all over the phoneline to the phone, and the red light doesnt light up. The configuration is verified as unchanged. Has anyone seen this problem before. I was

Hello, I've got a Cisco ATA 186 from Vonage plugged into my Asterisk box with a X100P card. This works great for the most part, but I'm having a disconnect supervision problem. I suspect the Cisco device doesn't provide any sort of analog disconnect supervision when it gets a SIP BYE message indicating the far-end has hung up. This causes Asterisk to leave the channel up

How do you prepend. I want to be able to dial 7 digits instead of of 11 for local calls. Can someone post there extensions.conf part that is relavent?

Hi all: Very disappointed, finally I left the attended call transfer with ATA 186 using SIP. With image 2.16-1, ATA sens '486 - Busy Here' when trying to transfer the call.. I consulted with Cisco guys and accepts that some problems with this service exist. Soon as I can I will try using MGCP. My doubt now is if somebody proved the DynExtenDB application. I read some commentaries but

I see that there's now a 2.16.1 upgrade path for Cisco ATA-186 devices, dated (variously) July 11 or July 14 2003. Here are some interesting bugs that claim to be fixed. Most notable is CSCeb17953, at least from my perspective, as I've hit this bug before. CSCea42480 The Cisco ATA ignores the Require:100rel header and processes call. CSCea69889 The Cisco ATA builds a 302 Moved

Hello - I'm using ATA-186 devices, with RFC2833 DTMF encoding. I am having problems with routines that input long strings of numbers, in that I am getting more than a small number of double digit entries. As an example, I have a section that asks for the user to enter a call forwarding number, and then puts that number into a database. Almost always, there are double digits when the

Hello list, i am using: asterisk CVS-10/13/03-11:54:33 chan_capi-0.3.0 ATA-186 V2.16.1.ms over MGCP Situation: ISDN calls ATA ISDN speaks with ATA ATA-Phone presses Flash and speaks to another one (SIP/snom200) ATA-Phone hangs up ISDN talks to SIP/snom200 snom200 hangs up The incoming extension of ATA keeps busy for a time (20 sec?), even its not off-hook anymore! Any ideas? -- Swapping

Has anyone come across the Cisco ata 186 not passing *70? When I press *70, the ata just goes back to a dial tone. The strange thing is, its only *70 and not the reset of the 70's. *71, *72, etc all go through fine. I've tried removing the the dialplan all together from the ata to try and let it pass, but no go??? If I setup an IP phone to its extension, *70 goes through fine, so i know

i have been trying to get a newly liberated (from vonage) cisco ata-186 (sip ios v3.1) working properly with asterisk. my client is behind a linksys wrt-54g, which up to this point hasn't proven to be a problem (i have several sipura spa-2000's and polycom phones working just fine behind them). (i'm running cvs-head from yesterday). after looking at the various suggestions,

I have a ATA 186 with SIP firmware 3.1 when I changed the configurations to use the g.726 codec I received many erros and the calls doesn't work. I changed the fields: - LBRCodec: 6 <- the code for g.726 - TXCodec: 6 - RxCodec: 6 The errors: Jul 9 13:15:37 NOTICE[1192491824]: rtp.c:500 ast_rtp_read: Unable to calculate samples for format G726 Jul 9 13:15:37 NOTICE[1192491824]:

I want to setup an Asterisk network using the Cisco ATA 186. What is the best place to order those devices? I'm not finding them anywhere. Al __________________________________ Do you Yahoo!? Protect your identity with Yahoo! Mail AddressGuard http://antispam.yahoo.com/whatsnewfree

I use cisco ATA 186 ( Version: v2.16 ) with sip protocol and asterisk as pbx. I need feature called as 'three way calling' or 'transfer with consultation'. Registering,calling and 'blind transfer' work fine. Is this feature provided by sip clients or by asterisk itself ? What I have to configure in ATA and what keys I have to press on my phones ? Three way calling is

Calling from a Cisco FXO port to an ATA-186 (SIP 3.1 image) via * (either CVS-HEAD-06/28/04-11:43:41 or CVS-HEAD-07/12/04-15:49:58) I didn't hear any ringing sound & get the following on the console: -- Called 5503 -- SIP/5503-f6b5 is ringing WARNING[-1323201616]: channel.c:1375 ast_indicate: Unable to handle indication 3 for 'SIP/10.10.2.250-9903' -- SIP/5503-f6b5 answered

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 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: http://www.sst.com/downloads/datasheet/S71077.pdf The above document gives exact specifications on the 4mb flash EEPROM that stores all program and configuration data on the ATA-186 (aka Komodo.) If you

Anyone have call waiting working on the ATA-186 connected to Asterisk? Other VoIP phones seem to work, but I can not get the ATAs to allow call waiting. Christopher M Iarocci Network Admin JD Posillico 631-249-1872 X244

Hi people, I have a Cisco ATA 186 which understands only the SCCP protocoll, therefore I am a pure beginner and I hope that anybody of you could help me. How will I configure the ATA which has 2 analog ports? For any support I would kindly thank you Tamer

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