similar to: DTMF outbound problem with ata 186

Thanks, Mark! Here's a summary of what one needs to do in order to get NAT working with Asterisk. Please note that I have a Cisco ATA-186, and your experience may be slightly different based on the equipment you're using. You'll need to have a CVS updated version of Asterisk as 2003-03-06 ~2:00 PM EST. NOTE: This currently works for outbound calling only, not inbound. In other

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

Hi Is posible to make a call from site A to Site C, and my question is, the rtp data is from A to C or is from A to B to C Site A Site B Site C ata186<-------->FW<--------->Asterisk<--------->FW<----------->ata186 Thanks -------------- next part -------------- An HTML attachment was scrubbed...

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

I have a couple of Linksys PAP2T-NA & Grandstream HT-502 extensions that are receiving random DTMF tones on their side, but that are not heard by the outside party. I have been using Asterisk 1.6.6 through 1.6.10 and have always had this issue. I am only using SIP on the Asterisk server and all extensions and trunks are set to rfc2833; outside of this issue DTMF operation works fine.

Hi everybody and sorry by the insistence. Nobody has working Tinc Server over a Raspberry in an environment in production? Best regards and sorry again, Ramses De: Ramses II [mailto:ramses.sevilla at gmail.com] Enviado el: martes, 08 de octubre de 2013 17:59 Para: tinc at tinc-vpn.org Asunto: Tinc Server and Raspberry PI (Rev. B). Dear gentlemen, I need configure a VPN

Hello, im having trouble with the following: [Asterisk]<------>[ISP]<------>[ADSL Modem]<------>[Linksys Router]<------>[Grandstream ATA]<------>[Analog Phone] On server: - Asterisk 1.6 - A2Billing 1.4 A2Billing have 2 Trunks: - TrExt: Voip Provider - TrInt: Internal Calls This structure works on first day (Asterisk+A2Billing installation/configuration). But on

Hi When i make a call using IAX2, the log of the remote asterisk say Nov 17 20:20:12 NOTICE[28686]: File rtp.c, Line 263 (process_rfc3389): RFC3389 s upport incomplete. Turn off on client if possible Nov 17 20:20:22 NOTICE[28686]: File rtp.c, Line 263 (process_rfc3389): RFC3389 s upport incomplete. Turn off on client if possible Nov 17 20:20:26 NOTICE[28686]: File rtp.c, Line 263

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) >

Hi - I need to configure Asterisk (Trixbox 2.2) and my ATA-186 with both ports. I need to be able to call one port from the other-- the idea is to have two phones in two different locations that _can_ call each other. So, in reading the Asterisk Wiki and other sites, the best documentation I found was this: http://www.loligo.com/asterisk/Cisco/ATA-186-guide.v20030628.txt

Hello, I have installed sacesfully Asterisk. I connected to that two ATA'a 186. I can call between them wihout a problem. Everything works fine. I' ve installed X-Lite on couple PC's and registered with Asterisk SIP proxy. Problem is now that I cannot call between ATA 186 and X-Lite. I can call from X-Lite to X-Lite and also ATA to ATA. If i try to call X-Lite to ATA or ATA to X-Lite,

I must be doing something incorrectly, or something is wrong with ATA-186 reINVITEs in SIP. Perhaps someone more enlightened than me can lend me a hand. I have been attempting to get two SIP phones to reINVITE to each other, and I've been unable to think of or uncover the correct method. The calls always go through the Asterisk server, no matter what I try. I've simplified things

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

Hello all! I have an FXO port on a cisco router that is directly connected to our PBX. Our ATA-186 (firmware version 3) registers with asterisk, which connects to our cisco router's fxo port to give me a dialtone on our PBX from the ATA. How do I pass the flash button to the PBX? It seems the ATA-186 wants to control the flash by putting my first call on hold and prompts me to dial another

My asterisk box doesn't recognize DTMF from my analog phone, plugged into my ATA(linksys pap2 version2). I can make/receive calls fine... it's just that, for example, I cannot login to my asterisk voicemail. Softphones (such as x-lite) are fine. I've turned up a few articles via google where some people have this trouble, but have not seen suggestions on how to fix. I presume

I have an Asterisk server it runs great with SIP phones, soft SIP phones (twinkle) and a soft SIP phone app on my Android phone but I am having problems getting two ATA boxes working. I have a Linksys PAP2T, it is unlocked and I have used them before with no problems. I was able to receive calls with from any local SIP phone or from my Link2VoIP connection via the Internet but it could not call

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

Does anyone have experience with ATA-186 Dialplans? If I was doing this in Asterisk I would use the following patterns: 911 _9NXXXXXX _91XXXXXXXXXX _NXXX _*XX I tried the following in my ATA-186, but it doesn't seem to work: 911S|9.r6S|91.r9S|^1.r2S|*..S Does any one have any ideas/suggestions? Thanks in advance, --Eric

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

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,