Re: [asterisk-users] Asterisk 1.4.26.2 died after 80 days uptime

I may be late to this thread, but my own restarted every 3-5 days until I upgraded to 1.4.29 (I skipped 1.4.28). It`s been running for 8 days now, which isn't long enough for me to declare whatever-it-is fixed, but enough to say it's at least better with 1.4.29 stability w

[asterisk-users] Removing line 2 from CISCO 7940g

still trying to log into Asterisk with that account. I have tried removing the config line and blanking out the options but it still persists. Does anyoen know how to get rid of the thing? Mike. signature.asc Description: Digital signature ___ -- Bandwidt

[asterisk-users] Problem configuring TDM400

rator, dahdi_cfg # # Global data loadzone= uk defaultzone = uk fxoks=1 fxsks=3,4 I have tried only bringing up certain channels but that still fails. Does anyone have any idea what could be wrong? Mike. signature.asc Description: Digital sign

[asterisk-users] Message Waiting Indicator on DAHDI line

nyone have any suggestions please? Mike. signature.asc Description: Digital signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.ne

Re: [asterisk-users] Message Waiting Indicator on DAHDI line

that MWI is carried by CLID. Would it still look for MWI if usecallerid=no? Mike. signature.asc Description: Digital signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Regist

Re: [asterisk-users] Message Waiting Indicator on DAHDI line

message in that mailbox? Why should commenting it out help? I assume that what happens is that the FXO line rings, so Asterisk rings the FXS phone as per the extensions.conf, this creates a MWI event which goes to the voicemail system, which then passes a MWI event to the SIP phon

Re: [asterisk-users] Message Waiting Indicator on DAHDI line

On Wed, Aug 05, 2009 at 02:13:01PM +0800, D Tucny wrote: > The problem is that your mailbox line was below channel=1, as such, it > applied to the next channel, channel=3 not channel=1... > d Nice one. Thanks for spotting that. Mike. signature.asc Description: Digital

[asterisk-users] Stale auth messages

calls, lots of registrations of course, but nothing worth 2Mbits/s) Regards, Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net

Re: [asterisk-users] Stale auth messages

Sorry, that is running 1.4.26.1. Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Thursday, August 13, 2009 23:22 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users]

[asterisk-users] Polycom push application for microbrowser

Fire Drill at 2pm Please exit and congregate at your appropriate location outside Using curl to send it to the phone ("192.168.1.54/push") on the LAN as a first test. (all urlencoded, yes) Did anyone ever succeed in doing this here? I'd apprecia

Re: [asterisk-users] Polycom push application for microbrowser

Hi, yes I did, I did have errors at first but that hurdle has been cleared. Thanks for the try :-) Mike > -Original Message- > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > boun...@lists.digium.com] On Behalf Of Dave Fullerton > Sent: Thursday,

Re: [asterisk-users] Polycom push application for microbrowser

I've tried turning logging way up for the relevant portions of the sip application, but no telnet. Not sure how I would go about this to get more info that what I already have. The phone is giving me a response, it's just that the response is "push message cannot be d

[asterisk-users] Voicemail - remove option to save in different folders

I am looking to configure the asterisk voicemail system to stop asking for the folder ("work", "personal", etc) in which to save messages when I do save them. Is there any configuration to do this? Mike ___ -- Band

[asterisk-users] Aastra, Asterisk 1.4 and Voicemail

e no doubt many people have done this. Anyone care to point me in the right direction? Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webina

Re: [asterisk-users] Aastra, Asterisk 1.4 and Voicemail

Hi Bob, Thanks for replying. I've thought of doing that, but softkeys are limited and for a phone with many call appearances (4-5) that would be using many of the softkeys. Mike > -Original Message- > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-us

[asterisk-users] DID number

Hi All, Anyone one info of where I can get a 'free' DID number ? I have setup my asterisk box (home) and want to learn more but I need a #. thanks in advance, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital

Re: [asterisk-users] Wanted: free DID number and provider feedback

Ok, I see there's alot out there of voip providers. Curious what to watch out for ? charges and fee's, etc ? If anyone has feedback as to a GOOD voip provider experience (one that gave FREE DID) Please share. Again, I am doing this to learn about asterisk, I'm currently testing it at home. than

Re: [asterisk-users] Wanted: free DID number and provider feedback

My bad, I'm in Los angeles california usa On Thu, Mar 18, 2010 at 1:06 AM, SIP wrote: > What country are you in? Makes somewhat of a difference. > > N. > > > On 3/17/2010 8:49 PM, Mike wrote: >> Ok, I see there's alot out there of voip providers. >> >&g

[asterisk-users] Question about 1.6: multiple IP on a single Asterisk box / multi ISP routing

same system (And more to the point, allowing easy outgoing routing based on which NIC was used). Am I correct? Bonus question if I am indeed correct: how stable is 1.6 right now, compared to the latest 1.4 (1.4.31)?

[asterisk-users] OT: Windows TAPI command-line driver

Hi, This is a bit off-topic, but still related to telephony. Is there a barebones TAPI driver that exists that would allow me to call up a command line with, as parameter, the number to dial. For exemple, Outlook integrates with TAPI, so that TAPI driver would allow me to call my own app w

Re: [asterisk-users] OT: Windows TAPI command-line driver

Thanks, will take a look. Althought none of those things seem to allow me to call up my own handler for calls, does it? Or am I misreading? Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Wednesday, May

[asterisk-users] How to have Asterisk respond from the IP address used for registration

t one? Regards, Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-

Re: [asterisk-users] How to have Asterisk respond from the IP addressused for registration

I should have mentionned this is already done. I can see that is a SIP response when trying 192.168.1.3, but the phones fails to register. I suspect a NAT/firewall issue because packets are leaving for 192.168.1.3, but coming back from 192.168.1.2. Mike From: asterisk-users-boun

Re: [asterisk-users] How to have Asterisk respond from the IP address used for registration

Re: [asterisk-users] How to have Asterisk respond from the IP > address used for registration > > > > On Thu, 27 May 2010, Mike wrote: > > > Hi, > > > > > > > > I have a test server with 2 NICs, each with it own IP address. Let`s say > > 192.168.1.

Re: [asterisk-users] How to have Asterisk respond from the IP address used for registration

Hi Andrew, Thanks, I'll look this up. The term "packet mangling" wasn't used in my many google searches. Mike > > On 28/05/2010, Mike wrote: > > That was a simplified example. I actually have two links from different > > ISPs, totally different network

Re: [asterisk-users] How to use one single IP as origination

See bindaddr here: http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf That should do exactly what you want. Regards, Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of CDR Sent: Sunday, May 30, 2010 10:06 To

Re: [asterisk-users] How to use one single IP as origination

s, at least on 1.4. I read somewhere (can`t find the page) that 1.6 works differently. Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Monday, May 31, 2010 9:55 To: 'Asterisk Users Mailing List - Non-Commercial

Re: [asterisk-users] How to use one single IP as origination

policy (if it came in on NIC 1, send it back the same way even if it`s a less direct route). Somebody told me to lookup Packet Mangling, which I have yet to do. Will probably write a wiki page about this if that works, because I don`t seem to be the only one with this need. Regards, Mike

[asterisk-users] Reloading queue members (realtime DB)

ot;show queues" from cli. They aren't reflected when a new call comes in, or when I reload the dialplan. What do I need to do for the changes to be shown in the CLI, short of restarting Asterisk? Regards, Mike -- __

Re: [asterisk-users] Reloading queue members (realtime DB)

I did, "show queues" doesn't show the membernames, but the interface (which is normal if the membername is NULL in the table, but it isn't). Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Aksel Celasun

Re: [asterisk-users] Voicemail : mail attachment to multiple mail-addresses

Actually IIRC comma or semi-column worked. Try both, one of them will do. Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Monday, May 31, 2010 16:08 To: Asterisk Users Mailing List - Non-Commercial

Re: [asterisk-users] How to have Asterisk respond from the IP address used for registration

be gone or ignored. I am no routing expert, but coming out of Asterisk the responses back aren't marked are they? I mean, strictly speaking those aren't the same packets as the ones that came in. Mike > -Original Message- > From: asterisk-users-boun...@lists.digium.

Re: [asterisk-users] Reloading queue members (realtime DB)

I ended up restarting Asterisk (that worked, proving that my config isn't the issue) but I was hoping something less "intrusive" would work. Regards, Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Aks

Re: [asterisk-users] OT: Windows TAPI command-line driver

Klaus, Thank you, that clarifies it a lot for me. I will look those things up and figure out something. Meanwhile, if someone wants who has experience with TAPI services wants to offer me his (paid) services I would be glad to consider. Mike > -Original Message- > From:

[asterisk-users] Slightly OT: trying to mangle packets from Asterisk for a multiple ISP setup (reward)

uting table. What the heck am I missing? I believe I have done my homework, but there is no more door left to bang my head on. Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New t

Re: [asterisk-users] Slightly OT: trying to mangle packets from Asterisk for a multiple ISP setup (reward)

ectly according to my logging) with iptables. Mike > -Original Message- > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > boun...@lists.digium.com] On Behalf Of Joe Freeman > Sent: Tuesday, June 01, 2010 10:56 > To: asterisk-users@lists.digium.com >

Re: [asterisk-users] Slightly OT: trying to mangle packets from Asterisk for a multiple ISP setup (reward)

through the routing rules. I will definitely post something here when I am finished, you never know, I might not be the only guy with this question. Mike > -Original Message- > From: Gordon Henderson [mailto:gordon+aster...@drogon.net] > Sent: Tuesday, June 01, 2010 12:01 >

Re: [asterisk-users] Slightly OT: trying to mangle packets from Asterisk for a multiple ISP setup (reward)

e any reason who a fwmark matching rule acts differently when it comes to rewriting the src address ? Google can`t help, I`m certainly not the only one who's ever tried that. Mike > -Original Message- > From: asterisk-users-boun...@lists.digium.com [mailto:asteri

Re: [asterisk-users] Slightly OT: trying to mangle packets from Asterisk for a multiple ISP setup (reward)

I guess I can now boil this question down to it`s essence: How do I change the source IP address (from x.x.x.x to y.y.y.y) of an outgoing packet generated locally on an host (i.e. by Asterisk) using iptables, if I'm not doing any NATing (preventing my from using SNAT --to-source)?

Re: [asterisk-users] Limit total length of calls to a specifig SIP peer

your call. Then again, if you have one hour left, you should divid this by six (in case everyone calls out at the same time). So if you have 2 hours left, only allow one call to use 120minutes/6 Mike. > -Original Message- > From: asterisk-users-boun...@lists.digium.com [mailto:as

Re: [asterisk-users] Priority between calls in different queues

Hi, Isnt there a parameter called "weight" for each queue that defines exactly that? I never tried it, but it appeared to do exactly what you want (according to the invaluable but often out-dated wiki). Regards, Mike > -Original Message- > From: asterisk-users-boun...@

Re: [asterisk-users] Call ended after 31 seconds

You`re using Xlite/eyeBeam by any chance? Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña Sent: Friday, June 11, 2010 16:12 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Call ended after 31

[asterisk-users] Multiple parking lots - 1.6

parking lots on a single Asterisk install? Thank you, Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http

[asterisk-users] Slightly OT: Cisco SPA525G and network errors

for power users. What should I be looking for? The Wifi router is a few feet away, and the Wifi activity is limited to that phone and a (low usage) laptop, so Wifi overutilization is probably not an issue. Regards

Re: [asterisk-users] OT: Physical SIP phone with inbuilt VPN support

I have a Cisco SPA525G that seems to support it, but I've never needed that. I would assume most Cisco SPA phones would support that too. Mike > -Original Message- > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > boun...@lists.digium.com] On Behal

[asterisk-users] PRI span problem - no D channel

Primary channel 72 as D-channel anyway! Any clue, anyone? Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:

Re: [asterisk-users] PRI span problem - no D channel

I thought the same, but the cable is in and the light is green. Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of liuxin Sent: Tuesday, June 22, 2010 22:05 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject

Re: [asterisk-users] PRI span problem - no D channel

Same as the other spans coming out of the same MUX! signalling = pri_cpe I got 2 spans working fine from the same MUX. This one isn't working for some reason. The all have similar configs, except where I need to change channel number and primary channel number. Mike

Re: [asterisk-users] PRI span problem - no D channel

about the DS3 timelsot. Since it worked fine prior to a reboot (after which, I assume, the dahdi services jumped from some old version to the newest one I had) I hesitate to blame my provider, but I've opened a ticket with them just in case. Mike > -Original Message- > Fro

Re: [asterisk-users] PRI span problem - no D channel

I checked, it made sense. But it isnt it :-) The fourth span is commented out, but the third one isnt (and shouldn't be since it's active). Thanks for the tip though, could have easily been this. Mike > -Original Message- > From: asterisk-users-boun...@

[asterisk-users] Meetme delay - normal?

cases. Regards, Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk

Re: [asterisk-users] Call drops on group paging asterisk - 1.4.22.1

The phone brand and model might matter here, I have had no such problems with Polycom phones. Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of das sandesh Sent: Friday, June 25, 2010 12:58 To: Asterisk Users Mailing List

[asterisk-users] Asterisk 1.6 and multiple parking

asking the same question. Is there some sort of sample conf file of how I would get this functionnal on the latest 1.6.x? Regards, Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-d

Re: [asterisk-users] Asterisk 1.6 and multiple parking

What am I missing you think? I only set the CHANNEL(parkinglot) value when parking the call. Do I need to set that value when picking up a call? (after all, I have no accessz to extension 800 it is created by features.conf) Regards, Mike From: asterisk-users-boun...@lists.di

Re: [asterisk-users] Asterisk 1.6 and multiple parking

setup the extensions myself. For futur reference. Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Wednesday, June 30, 2010 11:22 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject:

Re: [asterisk-users] Asterisk 1.6 and multiple parking

Here is my only question left about parkinglots in 1.6. How does the parkinghints=yes parameter work? I've tried using "core show hints" , but there are never any hints. Even when a call is actually parked in the correct parking lot. Any tips? Mike From: aste

Re: [asterisk-users] Asterisk 1.6 and multiple parking

I know, I've done this with 1.4 manually with hint extensions. But in 1.6 there is a parameter called parkinghints=yes that is supposed to set them up automatically. It certainly doesn't seem to be doing anything for me. Thanks, Mike From: asterisk-

[asterisk-users] 1.6.2: Using hints on multiple parking lots

for parkinglot1 and parkingloit2 so that my phone , which is subscribing to 800, only see parkinglot1 and NOT parkinglot2? I tried the obvious answer exten => 800,hint,park:8...@parkinglot1 but that didnt seem to do anything. Regards,

Re: [asterisk-users] Y-cords - What are they ?

I believe people use this for headsets, to have a superviror listen in on a call with the agent (for training purposes). You can therefore plug in two headsets on the same phone. Mike > -Original Message- > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-

Re: [asterisk-users] Management interface

I use a custom script that I run using SNMP, and graph that using cacti. Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce McAlister Sent: Monday, July 26, 2010 13:57 To: asterisk-users@lists.digium.com Subject: Re

Re: [asterisk-users] Aastra phones occasionally show "No Service" - Is there any network setting I can tamper to facilitate a quick DHCP renewal on the Aastra phones?

Hi Bruce, Did you ever get a working solution and confirm the underlying issue ? I am having the same issue on a set of phones, my next step is replacing the router, but I was wondering if you found something else. Regards, Mike From: asterisk-users-boun...@lists.digium.com

Re: [asterisk-users] What can make G.729a codec hostid change?

to think about this. Obviously it may not be a solution for everyone. Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tiago Geada Sent: Tuesday, September 07, 2010 14:44 To: Asterisk Users Mailing List - Non-Commercial Di

Re: [asterisk-users] SIP softphones answer but do not connect...

nect if no audio received for x seconds". We turned this option off, and things went back to normal. Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a liv

[asterisk-users] Forking a call

d the second an independant inbound call. Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/

[asterisk-users] Asterisk 1.6.2.13 - have asterisk reply from same IP address

with Asterisk). Is this possible? I know it wasn't with 1.4, but I was told 1.6 had something like this (something to do with not breaking SIP over TCP) Mike -- _ -- Bandwidth and Colocation Provided by http

[asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes

nyone have an idea of where to start looking? Can Asterisk report somehow what is causing this? Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live in

Re: [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes

Core show threads doesn't show how busy the threads are, unfortunately. But I can't see a difference between normal and spiked CPU looking at that output. Mike > -Original Message- > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > boun...@lis

Re: [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes

I found that bug before I wrote, and I was hoping you were right, but recreating those two missing "files" didn't help. I wasn't running 1.6.1 anyways, but I figured I'd try. There must be a way (Linux or Asterisk-centric) to see if a particular thread/mo

Re: [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes

DI Version: 2.3.0.1 Echo Canceller: MG2. On that server I have a 4-port PCIE PRI card and a TCE400B card. A few hundred SIP peers, but nothing 1.4 couldn't handle easily (same server). Mike > -Original Message- > From: asterisk-users-boun...@lists.digium.com [mailto:a

Re: [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes

Thanks Dave. I'll do a complete system restart when I can, and see if it helps. (an Asterisk restart didn't). > -Original Message- > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > boun...@lists.digium.com] On Behalf Of Dave Platt > Sent: Monday, September 27, 2010

Re: [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes - FIXED...?

e the storm and Dahdi will behave under load. Thanks for the help Shaun, I appreciate you helping point me in the right direction. Hopefully this will help someone. Mike > -Original Message- > From: Mike [mailto:l...@net-wall.com] > Sent: Tuesday, September 28, 2010 2:55 > T

Re: [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes - NOT FIXED

(sorry for the series of email, I realize I seem to be having a discussion with myself). I was wrong, the "problem" isn't fixed. Is having IRQ spikes every 10 minutes (under no load at all) the norm with Dahdi hardware? Mike > -Original Message- > From: Mike [mai

Re: [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes - NOT FIXED

If, as a non-HW guy, I understand correctly, it means that each card is sharing interrupts with a USB port. Nothing is plugged in those USB ports. If I remember correctly, the HP BIOS didn't let me do what I wanted exactly. Would disabling the USB ports help? Mike > -Original

Re: [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes - NOT FIXED

How do I go about finding which device is causing those interrupts? (to make sure it`s actually Digium hardware and not something else?) Mike > -Original Message- > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > boun...@lists.digium.com] On Behalf Of Mi

Re: [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes - NOT FIXED

ren't more interrupts, but the IRQ% time for the CPU shoots up. What can make it do that? Mike > -Original Message- > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > boun...@lists.digium.com] On Behalf Of Shaun Ruffell > Sent: Tuesday, September 28

Re: [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes - NOT FIXED

just grasping at straws? Mike > -Original Message- > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > boun...@lists.digium.com] On Behalf Of Mike > Sent: Tuesday, September 28, 2010 13:54 > To: 'Asterisk Users Mailing List - Non-Commercial Discu

Re: [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes - NOT FIXED

for a fire, maybe I'm just being paranoid. That`s what trending and SNMP alarms are for I suppose ;-) Mike > -Original Message- > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > boun...@lists.digium.com] On Behalf Of Shaun Ruffell > Sent: Tuesda

Re: [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes - NOT FIXED

Thanks. Will try all that. Night time work again ;-) Mike > -Original Message- > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > boun...@lists.digium.com] On Behalf Of Shaun Ruffell > Sent: Tuesday, September 28, 2010 15:37 > To: asterisk-users@l

Re: [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes - NOT FIXED

Thanks. The problem is I only have two PCIe slots, and they are both taken. I`ll definitely try disabling USB, I have no need for it right now. Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Luis Antonio Prata Barbosa Sent

Re: [asterisk-users] 2 way intercom recommendationforrestaurantkitchens

Polycom 501s were designed before the PoE standards were set in stone. So the PoE is actually not part of the phone, but part of the special PoE cable that is optional. So you absolutely need that special RJ45-like cable. Mike -Original Message- From: asterisk-users-boun

Re: [asterisk-users] DMTF Mode

orting the issue to them helps. But of course, verify on your side first. Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo Sent: Wednesday, October 13, 2010 10:12 AM To: Asterisk Users Mailing List - Non-

Re: [asterisk-users] DMTF Mode

Can you send us the SIP config of the sip provider (in sip.conf), removing appropriate passwords and static IPs of course. Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo Sent: Wednesday, October 13, 2010 10:22 AM

Re: [asterisk-users] advice re: Page() application

as advertised. Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cassius Smith Sent: Wednesday, October 13, 2010 7:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] advice re: Page

Re: [asterisk-users] advice re: Page() application

My SIP registration are name sort of like this : phonea-exten1, phone1-exten2, etc. Makes it easy to loop, I can send you a snippet tomorrow. But you have to know in advance all the SIP peer names. Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun

[asterisk-users] How to check PRI status from dialplan

Hi, Is there any way to see the status of PRI from the dialplan? I`d like to know whether it was Up or not before I attempt to dial on it. Second best would be to know if was down when I tried dialing (but I need to differenciate between Down and Up-but-provider-was-congested). Mike

Re: [asterisk-users] Soundpoint IP 430 -- discontinued.

eries. The problem with the Polycom 3xx series is that Polycom decided to be cheap on the phone UI, and it`s dramatically less-friendly than the other phones for anything else than simple calling/answering. Mike -- _ -- Bandwidth

[asterisk-users] Asterisk parking question

context anywhere, and I don`t think I should….? Anyone knows what I can do to ensure the call isn’t dropped? Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us

Re: [asterisk-users] Asterisk parking question

#x27;t this work? And more importantly: 2) what's this park-dial context you speak of ? Is this a hardcoded context calls go back to? Can I set one per parkilots (remember: I use multiple parking lots) Thanks for taking the time to answer my question. Mike -Original Message-

Re: [asterisk-users] Asterisk parking question

arkinglotA. The "unnamed" context seems like a bug, maybe I'll open something in the bug system if nobody can explain it... Or is there a secret place to define this in features.conf? Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users

Re: [asterisk-users] TCP port, VPN and resolving the cutting voice problem

n't result in more congestion created by the original congestion. That being said, isn't UDP sometimes looked at as being lower priority than TCP by many routers out there and dropped first when congestion does occur? That makes it a good reason to use TCP in some

Re: [asterisk-users] TCP port, VPN and resolving the cutting voice problem

e. VoIP may have changed some people`s mind, of course. Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Jones Sent: Tuesday, November 30, 2010 1:33 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] TC

Re: [asterisk-users] Asterisk with MySQL Cluster

The only thing I found workable, is to use a hostname (i.e. asterisk_sql) and update /etc/hosts according to which SQL server is up or down. It's a bit of a hassle, and it would be easier if Asterisk supported fallback SQL servers, but once done it works well. Mike From: ast

[asterisk-users] Asterisk error - 1.6.2 SVN - voicemail files "corrupted"

s? I don't mind manually fixing it when it happens, but what's wrong exactly? 2) If this isn't the right list for this (considering it's a SVN question), what is? I'm using SVN because of the blind transfer issu

Re: [asterisk-users] Asterisk error - 1.6.2 SVN - voicemail files "corrupted"

Thanks Jonathan, I did that, it worked. I thought it had something to do with 1.6.2 SVN, since I`ve been using Asterisk for 5 years now and the first time it happened was the day I used SVN. Regards, Mike -Original Message- From: asterisk-users-boun...@lists.digium.com

Re: [asterisk-users] Asterisk error - 1.6.2 SVN - voicemail files "corrupted"

Hi Tilghman, This particular customer was one of my less sophisticated customer, and I know for sure he isn`t using anything else than Voicemailmain. Not even the basic voicemail to email function. But I will keep an eye opened for any future problem. Mike -Original Message- From

[asterisk-users] Snom (vs Polycom) - provisioning

or me? Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing li

Re: [asterisk-users] Snom (vs Polycom) - provisioning

Thanks, that`s actually a useful document but it doesn't specify whether I can do FTP provisioning. I could assume I can`t, but I hoping it`s not the answer. Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of

Re: [asterisk-users] Snom (vs Polycom) - provisioning

Thanks Nicholas, One thing to keep in mind: TFTP and FTP are very different things, security-wise (none vs some) But I will definitely try to fudge my way with ftp:// Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of

Re: [asterisk-users] Snom (vs Polycom) - provisioning

Thanks. Too bad, FTP makes it much easier for the multi-tenant systems (IMO) Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik Sent: Tuesday, December 07, 2010 11:20 AM To: Asterisk Users

Re: [asterisk-users] Snom (vs Polycom) - provisioning

Hi Danny, Maybe I should spend more time thinking about this. FTP (on Polycom) allowed me to easily "segment" my customers, give them different versions of the firmware, etc. Thanks everyone for your help. Mike -Original Message- From: asterisk-users-boun...@lists.

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