similar to: cisco ata-186 behind NAT

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

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 wonder if someone out there could loan me a peek at their sip.conf? I have conflicting advice, for instance, about whether or not to use "nat=1" and also whether or not the ATA should be registering with the instance of asterisk it is going to be using to dial out. Thanks in advance. B.

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

When I try to connect to * using a Cisco ATA 188 configured with a MGPC firmware (v3.1.1), I just keep getting this message every 30 seconds or so : May 10 10:08:21 NOTICE[7913]: chan_mgcp.c:1509 find_subchannel: Gateway '192.168.1.27' (and thus its endpoint '*') does not exist Using tcpdump, I have checked that the ATA188 (with IP 192.168.1.27 and port 2427) sends UDP packets to

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

Hi all,

Hi, I'm having trouble getting caller*id to appear on my phone connected to an ATA186, and being called from Asterisk. Does anyone out there successfully see callerid on their ata186-connected phone? The "From:" header in the INVITE to the ATA seems to have the "right stuff" - eg From: "Study phone" <sip:6002@195.217.255.45:5062>;tag=as412db061 But

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

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

My Cisco 7905 can't register with Asterisk (1.0.7-BRIstuffed-0.2.0-RC7k on Debian stable). It could, however, register with another installation of Asterisk and the settings on the phone (apart from the SIP proxy address) haven't changed since then. On the new Asterisk box my sip.conf contains this: [jeremy] type=friend regexten=801 allow=g729 host=dynamic secret=PASSWORD nat=yes

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 been searching the archives but can't find anything substantive on this. most of the music on hold documentation discusses integrating with zap hardware, but i am trying to send it across a sip channel. I have the following in extensions.conf: exten => 2100,1,Answer exten => 2100,2,MusicOnHold(default) and have uncommented the "default" line in musiconhold.conf:

I'm trying to register with net2phone. I've already changed chan_sip.c, User-Agent: string to say "User-Agent: Cisco ATA 186 v2.16 ata18x (030401a)". But still I'm getting the error msg. Here is the debug msg: IP Address is xxx.xxx.xxx.xxx 11 headers, 0 lines Reliably Transmitting: REGISTER sip:66.33.146.12 SIP/2.0 Via: SIP/2.0/UDP

in my setup i have several users with DID lines coming in from various sip/iax providers. within our old phone system, a user could call their own DID line, then hit the * key when they hear their voicemail greeting and be prompted for their password. is there any way this could be replicated within asterisk? i'm having trouble figuring it out since it steps through things sequentially,

Have anyone tried to interface BT's Broadband Voice with asterisk? Kannaiyan

Hello- As part of the asterisk build/installation instructions it mentions that the zaptel drivers should be built and configured first. My question is whether they are required at all, in the case of a system with no hardware cards at all (as is the situation in my case). With them loaded I continually get the following message on my console (server not asterisk): Zapata Telephony Interface

Hi, I know this is slightly off topic but I figured the knowlege here is probably the best on the subject.. I want to setup remote offices with 4 to 6 SIP phones (SNOM 200) using ADSL and the internet to connect to the Asterisk box.. These phone will be behind an ADSL router using NAT... I don't want to setup another Asterisk system in each office so IAX is not an option.. I could use

Hello- I'm obviously doing something wrong here in trying to get an inbound DID to work with voicepulse. I have an outbound context set-up for those calls in iax.conf, and the appropriate register in- statement. within extensions.conf I am doing something like this: exten => 212xxxxxxx,1,Dial(SIP/admin,t) (where admin is the phone i am looking to forward to from sip.conf). i'm

I'm working on setting up a macro that will allow users to call their own DID number, and when they hear their voicemail greeting hit the * key and be prompted for their password to check vmail. For some reason though the background command isn't working as I'd expect it to: [macro-vmessage] exten => s,1,Answer exten =>