similar to: 'ztcfg -s' causes system hang

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:

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

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:

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."

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

Hi Bradley, Yes, I experienced quite a lot of echo with my IAXy, until I switched analog handsets - in my case, it was severe acoustic coupling in a cheap handset. Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On Mar 7, 2006, at 11:38 AM, Bradley M. Kuhn wrote: > I just purchased an IAXy

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

Greetings, I have just updated our test server to 2.6.9-34.EL and get the following error messages when compiling zaptel: make[1]: Entering directory `/usr/src/kernels/2.6.9-34.EL-i686' CC [M] /usr/src/zaptel/zaptel-1.2.1/zaptel.o /usr/src/zaptel/zaptel-1.2.1/zaptel.c:383: error: syntax error before "zone_lock" /usr/src/zaptel/zaptel-1.2.1/zaptel.c:383: warning: type defaults

Greetings, I am trying to get either of the above features to work with *, but can't seem to get it quite right. If anyone has them working, I'd sure appreciate an extract from the relevant config files. Or, maybe I'm tilting at windmills, and * doesn't support them - in which case, the underlying business need is to provide the one incoming call on more than one

Hello all, I have Asterisk 1.2.1 and a TE110P connected to a Nortel Meridian Option 81C system. The PRI line is currently setup as DMS100. Here are the relevant lines from zaptel.conf and zapata.conf: zaptel.conf: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 loadzone = us defaultzone = us zapata.conf: [channels] language=en context=from-internal musiconhold=default switchtype=dms100

I just implemented a system using a TE411P hardware echo cancellation card. Per Digium, I setup zaptel.conf, and zapata.conf the same way as I always have. To my surprise calls out to the PSTN had a terrible echo. 1 - 2 second delay, and quite clear. The echo was so bad that I had to remove the hardware echo cancellation module from the card. We are only using the 1st span of this card right

Hi R-helpers, I would like to subset my dataframe, keeping only those rows which satisfy the following conditions: 1) the string "dnv" is found in at least one column; 2) the value in the column previous to the one "dnv" is found in is not "0" Here's what my data look like: ??? POND_ID 2009-05-07 2009-05-15 2009-05-21 2009-05-28 2009-06-04 4 ? ? ? 101 ? ? ?

Is anyone getting '500 Internal Server' errors back from their Polycom phones when Asterisk sends a SIP NOTIFY message to them? I called Polycom tech support, who where utterly useless. Of course Polycom won't officially support it anyway, as they only support Asterisk Business Edition. We're using 1.2.9, but it's been ocurring for quite some time. We have about 35 phones and

Dear OpenSSH Developers, Thanks for all the great work on this important tool. We've built version 3.1p1 on SAPTC platforms under Solaris 2.8 using gcc 2.95.2. Several quick notes and a question: 1) There are several discrepancies between the INSTALL file on the openssh web site ftp://ftp.ca.openbsd.org/pub/OpenBSD/OpenSSH/portable/INSTALL and the output from "./configure

Hi R-helpers, I'm trying to use this code > pvh_dnv<-pvh[sapply(pvh=="dnv"),] to make a new dataframe containing the rows from pvh that contain the value of "dnv" in ANY column. But, it's not working. I get this error Error in match.fun(FUN) : element 1 is empty; the part of the args list of 'is.function' being evaluated was: (FUN) which, to

Hi, I'm on a Dell VEP 1405 running Debian 9.11 and I'm running a few tests with various interfaces given in PCI passthrough to a qemu/KVM Virtual Machine also running Debian 9.11. I noticed that only one of the four I350 network controllers can be used in PCI passthrough. The available interfaces are: *# dpdk-devbind.py --status Network devices using kernel driver

I am wandering if anyone can explain this unusual scenario. I am running Asterisk on Fedora. After the box boots, I run insmod wcfxs at a command line. The phones keep the red message light on and are in a state like they are not recognized yet. If I then run the command ztcfg, I don't get anything after the command, meaning that everything should be OK in zapata and zaptel confs and the