search for: virtutel

...ead over those CPUs very well... I'm not sure if they had something special happening that caused their symptoms, but, from your dual core machine you should be able to see whether or not the load is already being spread across the 2 cores OK with your workload... d 2009/3/27 Mike <list at virtutel.ca> Thanks that`s great info, and I've already subscribed to the HA mailing list. I understand call handling takes little CPU, but half my calls are transcoded from ulaw to g729 and vice versa. That seems to take my single CPU, dual-core 2.5Ghz machine up to ~35% CPU utilization. I imagi...

Here is what I had to change on the phone1.cfg file: I had this value in my 1.6.7 file, put in there following suggestions from the Wiki (http://www.voip-info.org/wiki/index.php?page=Polycom+Soundpoint+IP+501) : reg.1.server.1.expires="30" Now, this worked flawlessly with 1.6.7. But with 2.x, this seizes up the phone with a huge CPU load (approaching 100% at times) and makes it

Hi, I`m getting this (from one of my registered phone that has been installed at some location I can`t access at the moment) in the Asterisk CLI. Notice the last 3 digits of the IP address in the error message: Apr 12 23:06:16 WARNING[25593]: chan_sip.c:7036 check_via: '67.39.117.611' is not a valid host Of course it's not a valid host! But, when using "sip show

Somebody was helpful enough to give me the very latest release of Polycom's firmware (2.1.0). Unfortunately, I still get that issue. So I'm stuck asking again: Anybody ever got that? Mike _____ From: Mike [mailto:list@virtutel.ca] Sent: Wednesday, April 11, 2007 13:37 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Polycom 501 issue with latest firmware : sluggish keys Hi, I've upgraded a few Polycom 501 to SIP 2.0.3b (can't get 2.1.0 because of Polycom's firmware policy, b...

...ead over those CPUs very well... I'm not sure if they had something special happening that caused their symptoms, but, from your dual core machine you should be able to see whether or not the load is already being spread across the 2 cores OK with your workload... d 2009/3/27 Mike <list at virtutel.ca> > Thanks that`s great info, and I've already subscribed to the HA mailing > list. > > I understand call handling takes little CPU, but half my calls are > transcoded from ulaw to g729 and vice versa. That seems to take my single > CPU, dual-core 2.5Ghz machine up to ~...

...single phone for the customer, or is there a way? Mike you can redirect the ports of the router as well. Or you can configure your SIP phone to use a STUN server. Please read in voip-info.org about SIP NAT, there are good suggestions. regards On 1/20/06, Michakl Gaudette <michael.gaudette@virtutel.ca> wrote: > Hello, > > I'm a bit new to SIP, and I've set up a SIP line with Asterisk and my > wholesale provider. That worked, fine. I ahd to open up the ports on my > router, forward them to the correct box, again fine. > > Now, if I get one of my customers to c...

asterisk -rx 'core show channels' | grep DAHDI | sort -n Channels with a value of 1-23 are on your primary DS1, channels with a value of 25-47 are on your second DS1. Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 ----- "Mike" <list at virtutel.ca> wrote: > > Hi, I have just recently been using DAHDI, and I wanted to know how to monitor capacity. Let's say I have two DS1 (23 channels) coming in, one for Florida (let's say) and one for New York. How can I get a reading of how many channels of each T1 port is be...

Hi, I am looking for a good ATA recommendation, ideally something: 1) with one FXS and one LAN port (so it's as inexpensive as possible) 2) That can be provisioning using FTP (configuration and firmware upon reboot, ideally remote reboot from a sip notify) 3) Supports T.38 Nice to have would be: a) PoE powered and AC powered (my choice) b) Small size-wise I have been

Hi, I'm looking for a good network device that does bandwidth management. It can be integrated in a router or stand-alone, but must be SIP-friendly. I`ve tried the DIR-655 (latest firmware is SIP-hostile, and the latest hardware revisions can't downgrade to the version that worked well) and the DI-724GU (SIP-friendly, but bandwidth management is automated and not configurable

Hi, I have a question of using 2 SIP providers. Let's say I have provider A and provider B, and I would like my calls to go to A, and then B if A wasn`t available Something like this would work: exten => 1234,1,Dial(SIP/providerA) exten => 1234,2,Dial(providerB) exten => 1234,3,Hangup But what if I want to put in a delay? If I put 30 seconds on each of them, I'll wait a

Hi, I`m occassionally seeing CDR(accountcode)'s value empty at a place in my diaplan where it was filled with some value a few lines before, with nothing else having changed it. It`s giving me headaches (as I rely on it for MySQL queries). Anything I can do? Mick -------------- next part -------------- An HTML attachment was scrubbed... URL:

asterisk -rx 'core show channels' | grep DAHDI | sort -n Channels with a value of 1-23 are on your primary DS1, channels with a value of 25-47 are on your second DS1. Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 ----- "Mike" <list at virtutel.ca> wrote: > > Hi, I have just recently been using DAHDI, and I wanted to know how to monitor capacity. Let's say I have two DS1 (23 channels) coming in, one for Florida (let's say) and one for New York. How can I get a reading of how many channels of each T1 port is...

Hi, I'm looking to implement Fax reception on a SIP line. I`ve been looking at the Wiki and some other web pages and it`s far from clear what I need to do, or if it`s even possible. 1) Is it possible, or does it only work on Zap channels? (as I`ve read somewhere) 2) Is there a good reference on the web to do so? Thanks, Michael

Hello, I'm a bit new to SIP, and I've set up a SIP line with Asterisk and my wholesale provider. That worked, fine. I ahd to open up the ports on my router, forward them to the correct box, again fine. Now, if I get one of my customers to connect his SIP phone to my Asterisk box, and HE'S behind a NAT firewall, does he have to go through the same process, or is it just the Asterisk

That was exactly it! Thanks you VERY much! Mike ---- For the sip setting in sip.conf that setsup your voip provider add: canreinvite=no On 2/6/06, Michakl Gaudette <michael.gaudette@virtutel.ca> wrote: > > Hi, > > I've had a bit of a problem with one way audio, and it happens exactly when > I believe it shouldn't (and works perfectly when I would guess I could have > issues. > > Setup: > GrandStream GXP2000-------Linksys > Router-----------Inter...

Please disregard, I'll blame it on friday :-) It works fine now, just a typo. _____ From: Mike [mailto:list@virtutel.ca] Sent: March 24, 2006 1:36 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Extension a? Hi, I want my users to be able to get into VoiceMailMain when they press * while listening to their own greeting. It`s standard operating procedure with most voicemails...

Hi, I have a question on VoIP adapters. As far as I understand, those adapters are usually used to connect DSL/Cable access to a normal phone (Internet to Adapter, then to PSTN phones). I want to know if you can use those adapters to do the opposite: connect a few lines (1-4 let`s say) to the adapters, then deliver via SIP to an Asterisk box. (I know I could use a TDM400 and Asterisk, but I

Hi, I'm trying to make an apparently simple thing work, but I don't see how it is possible with Asterisk. This is my extensions.conf: exten => 1234,1,Dial(SIP/123456/555-555-5555|20|D(7777)) ;After call connects, send DTMF 7777 exten => 1234,2,VoiceMail(1234@context); What I obviously want is that if nobody answer the call, go to voicemail. Basic stuff. Problem is Asterisk

Hi, Where should I go to get the Polycom`s latest official (non-beta) version? I am registered on the Polycom customer website but that doesn't seem accessible. Regards, Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061102/284b67a7/attachment.htm

I`ve been noticing alot of those messages in the CLI lately: Apr 5 11:18:02 NOTICE[25593]: chan_sip.c:6444 check_auth: stale nonce received from '<sip:reg-1@pbx.domain.com> I haven't changed my configuration in ages. What could be the cause of this suddent appearance? Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: