similar to: IAXy (S101) echo?

You can even set it to zero. Mine works well when in zero. The line pick up immediately :> -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Chris Stenton Sent: Thursday, February 09, 2006 6:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SPA-3000 VOIP-PSTN

Hi there, I'm getting a bunch of these errors from Polycom phones in 1.2.1: ERROR[24301]: chan_sip.c:10790 handle_request_subscribe: Got SUBSCRIBE for extensions without hint. Please add hint to 4003 in context internal I've searched the Wiki and archives to no avail - what do these errors mean? Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web:

-----BEGIN PGP SIGNED MESSAGE----- CERT Advisory CA-2003-21 GNU Project FTP Server Compromise Original issue date: August 13, 2003 Last revised: -- Source: CERT/CC A complete revision history is at the end of this file. Overview The CERT/CC has received a report that the system housing the primary FTP servers for the GNU software project was compromised. I. Description

Greetings, We are trying to make our corporate directory (around 400 entries) available via TFTP to some Polycom IP501 phones. A small (~40 entries or so) file works, but the full file fails to load. Does anyone know what the upper limit on directory entries is? The size of the XML file itself is only 60K - you'd think that would all fit into the phone with no problems..... I would

I remember seeing something like this on the list a while ago, but I'm darned if I can find it. We have a number of Polycom IP501 phones, some of which have more than one registration on them. When a voicemail is left for a phone with only one registration, the MWI lights up and stays lit until the voicemail is listened to. However, on our phones with more than one registration, the MWI

I'm in the UK and have had terrible trouble getting the right contract. After looking on voip-info I set off looking for a vendor only to find that no-one sells the $9 contract. Cisco retracted that one :-( I asked for the equivalent and they said I needed a ?20 contract. I said fine but after 20 minutes or so got a call saying I couldn't have that one as it didn't give me access

Greetings, We are currently testing a Sipura SPA-3000 as a gateway from our Asterisk system to a PSTN line for 911 access. We have a number of locations and want to place an SPA-3000 in each, connected to a PSTN line that will provide the correct ANI/ALI information to 911 for each location. It all works great, except for a reasonably significant (4 seconds) delay between when the SPA-3000

Hi there, We're trying to get IMAP voicemail storage working on an MS Exchange server - I would be grateful if anyone who has successfully done this could post the magic soup here, as extensive Google searching has yielded nothing other than tantalizing references to it being done without any specifics. -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web:

On Sep 18, 2012, at 5:46 PM, Bradley M. Kuhn <bkuhn at sfconservancy.org> wrote: > I'm sorry for the intrusion of a project policy discussion onto the > developer list, but there may be many here who may have thoughts, input, > or question regarding Conservancy's offer for LLVM's membership, > discussed below. If you're not interested in that topic, please feel

Greetings, The Polycom SIP 1.5 Admin Guide says this: "3.1.8 Connected Party Identification Where possible, the identity of the remote party to which the user has connected is displayed and logged. The connected party identity is derived from the network signaling. In some cases the remote party will be different from the called party identity due to network call diversion."

Hi there, We are experiencing an issue on RHEL 4 (2.6.9-22.EL) with our TDM110P - whenever we enter 'ztcfg -s' to stop the span, the entire system crashes, requiring a reset. I have seen this (http://lists.digium.com/pipermail/asterisk-users/2005-June/ 112097.html) and thought it might be the answer, but we still get the crash on the first step. I have also seen this thread

Greetings, I am trying to gauge the level of interest in an Asterisk users' group in Vancouver, BC (or in BC in general). If you would be interested, please reply off-list. Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp

Greetings, This is my annual post-Astricon attempt to get an Enterprise Asterisk User Group off the ground. We are a municipal government using Asterisk to replace a legacy PBX. I'd be interested in starting a group of similar enterprise users (say, 100 seats or more) other than resellers, carriers and call-centers who are using Asterisk to support their non-telecom-related business

John Novack wrote: >And the only IAX2 box made is the Digium one, with it's current shortcomings ? >From reading through the archives, it seems there is currently no way to reset to factory default, no >written MAC address on an individual box, and some other instabilities requiring frequent resets. Yeah, there's the rub. Dunno if it's worth it, I'm willing to give it

I have a brand new IAXy I'm playing with. I do not get a dialtone on the phone, or any response at ll on the phone. No sound, no dialing, no ringing. The phone and wire are tested and known to be good. I think I have it setup correctly. When I give the iaxprov command I get this: #iaxyprov 192.168.1.90 iaxy.conf 02: c0 a8 01 5a 05: 11 d9 03: ff ff ff 00 04:

Hi, I've got an IAXy, and provided it with connection data - unfortunately didn't provide a fixed ip address, but left with dhcp. Everthing went fine, and I could work a while with it. After a restart of IAXy, it didn't negotiate a new IP address, nor used the old one. I even restarted the DHCP server and cleared its cache. No success. I went to another office with a different

Hey Everyone, I recently purchased a few IAXy units for some testing, and have come across a some problems. One of them can be provisioned but will not ring the connected telephone. It also does not provide a dial tone when you pick up the phone. When a call is made to the IAXy one of the green lights on the board goes from a solid on state, to a flashing state that coincides with the

The file used with a Digium IAXy device: "iax.conf" has the line: "port=5036" (I also use the "bindaddr=192.168.1.91" entry) but when Asterisk talks to the IAXy device it used port 4569 (from "tcpdump"). How are the port numbers assigned? What tells the IAXy which port to use. The IAXy provisioning file "iaxy.conf" file does not specify a port

The file used with a Digium IAXy device: "iax.conf" has the line: "port=5036" (I also use the "bindaddr=192.168.1.91" entry) but when Asterisk talks to the IAXy device it used port 4569 (from "tcpdump"). How are the port numbers assigned? What tells the IAXy which port to use. The IAXy provisioning file "iaxy.conf" file does not specify a port

Hi All, I have two old S100 units (the blue ones, not the newer black ones). I am trying to reset these to factory default using the following instructions, but it is not working. Does anyone have any other suggestions to reset this model of the adapter? Tried this: 1. Remove all of the cables, except for the power cable. 2. The factory reset button is next to the RJ45 jack. Press and hold the