search for: pajari

Hi all, I'm confused, in digium website, it says: TDM2400P: It supports a combination of up to 6 FXS and/or FXO modules for a total of 24 lines. 6 plus 6 is 12, how come it's 24? if I have 24 PSTN lines, i'll be needing 24 FXOs. Pls. elaborate. thanks. Lito -------------- next part -------------- An HTML attachment was scrubbed... URL:

...Nortel, NEC, Plantronics, Plycom, Samsung, Tatung, and Vitelix--announced the public beta program for Microsoft Office Communications Server 2007 and Microsoft Office Communicator 2007." http://news.com.com/8301-10784_3-9719931-7.html?part=rss&subj=news&tag=2547-1_3-0-20 -- George Pajari (dCAP), netVOICE communications 604 484 VOIP(8647) x102 www.netvoice.ca www.ip-centrex.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102)

...y: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 10000 T305 Timer: 30000 T308 Timer: 4000 T309 Timer: -1 T313 Timer: 4000 N200 Counter: 3 (a) What is causing this? (b) How can it be fixed? (c) Why does Asterisk not recover automatically to what appears to be an intermittent problem? -- George Pajari (dCAP), netVOICE communications 604 484 VOIP(8647) x102 www.netvoice.ca www.ip-centrex.ca www.ip-pbx.ca www.vpas.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102)

...ND be able to use # during outbound IVR calls? The obvious hack is to have a different dial plan for outside calls that does include the T option in the dial plan (i.e. dial 8+ for IVR calls, 9+ for normal calls that can be parked/transferred) but I'm hoping for something more elegant. George Pajari www.netvoice.ca www.IP-Centrex.ca

Hello! In my relentless quest for knowledge, I pose this question: who's got the biggest dialplans, and how big are these monsters? What's the biggest dialplan in use right now? If you feel you are a competitor, let me know how many contexts/extensions/priorities you are dealing with. Maybe the context with the most extensions, the extension with the most priorities would be

Michael Robertson, founder of both MP3.com and Lindows, has launched a new company to supply inexpensive SIP phones ($129 for two) and related services. See today's press release at http://www.sipphone.com/tiki-index.php?page=SIPphone%20Inc My question for the list is who will be the first to report on the compatibility and usability of the SIPphone with Asterisk? The functionality

...sion.com/columnists/2004/0607faceoffyes.html On a related note, they also have an article arguing the contrary position (see link within article). I'm too busy right now to write up a response showing the flaws in that column but others on the list might wish to contribute to the fray. George Pajari www.netvoice.ca www.IP-Centrex.ca

I'm trying to get a Carrier Access Corp. Channel Bank I working with a Digium T100P without success. What is stranger is that the status lights on the channel bank and T100P seem to change almost each time I power cycle the channel bank or reset the T100P. The channel bank has three status lights: T1, Framing, Status. T1 is green, Status is yellow, and Framing is usually red but sometime

>Date: Mon, 07 Feb 2005 02:22:07 -0800 >From: George Pajari <George.Pajari@netVOICE.ca> >Subject: [Asterisk-Users] Remote MWI via IAX? >To: Asterisk Users Mailing List - Non-Commercial Discussion. > <asterisk-users@lists.digium.com> >Message-ID: <4207414F.10807@netVOICE.ca> >Content-Type: text/plain; charset=ISO-8859-1; form...

We have a customer with a handful of Sayson/Aastra 480i phones behind a Juniper Networks Netscreen firewall registering with our hosted PBX service. The Netscreen monitors the REGISTER messages and only keeps the reverse mapping open for the duration of the registration period. It appears that every so often the Sayson does not send out another REGISTER message after the registration has expired

I wrote earlier: > Please tell me the obvious mistake I'm making here.... The problem was a lack of sleep. Sorry to have troubled the list. -- George Pajari, netVOICE communications 604 484 VOIP (484 8647 x102) Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) www.netvoice.ca www.ip-centrex.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca

...using outgoing directory until we get a winner, throw the willing called party into the same conference. (2) Park the incoming call, make the outgoing calls, transfer the willing called party to the parked call extension (not sure this will work but?)? What is the quality solution? -- George Pajari, netVOICE communications 604 484 VOIP (484 8647 x102) Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) www.netvoice.ca www.ip-centrex.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca

I am looking for an ATA that has had very reliable results when passing FAX over IP. I was thinking of testing the Cisco (not Linksys) ATA 186 I1, ATA 186 I2, ATA 188 I1. This is what I'm looking for: FAX -> PTSN -> through Asterisk -> ATA -> Fax Machine. I have QoS from PSTN entry to ATA on the network so I can assure precedence. What has everyone out there been using

I'm looking at switching VOIP providers, but want to ensure I move to a company with sufficient capacity. Can any Canadian VOIP users post/email me with feedback on their providers? I'll post the results for all to read...... -------------- next part -------------- An HTML attachment was scrubbed... URL:

...his being developed for Asterisk and yet it would appear to be a critical component needed to migrate customers used to fully integrated "Unified Messaging" systems to Asterisk. (a) Has anyone cracked this nut (or started on it)? (b) Anyone interested if we post a bounty? -- George Pajari, netVOICE communications 604 484 VOIP (484 8647 x102) Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) www.netvoice.ca www.ip-centrex.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca

...shows "Status: Provisioned, Down, Active". It appears that Asterisk is not recovering from the errors. Restarting Asterisk will not bring the PRI back up -- that requires the zaptel drivers to be unloaded and reloaded. Why is this happening and what can be done about it? -- George Pajari (dCAP), netVOICE communications 604 484 VOIP(8647) x102 www.netvoice.ca www.ip-centrex.ca www.ip-pbx.ca www.vpas.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102)

The question was posed: "incomming calls for 45 or so people that will call in 3 or 4 time each day during (approx) normal business hours" The comment was made (taken out of context): "The quick math says that 45 people with 4 calls is 180 calls a day. In a 8 hour day you have 480 minutes. From 480 minutes 1 port could handle the load if the call was under 2.5 minutes long and

We have a customer who is connected to our PSTN gateway using IAX and noticing that even when the traffic from their site is modest their outbound audio has short dropouts. Inbound audio is fine. (They have ADSL so it is expected that outbound audio would be the first to experience problems.) We have several questions to pose to the collective wisdom of this list. Q1: Are there any statistics

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > George Pajari > Sent: Thursday, June 09, 2005 10:19 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Voicemail and MS Exchange Synchronization > > > We have a customer considering migrating from a large Nortel > PBX with a > third-party voic...

Any opinions on the best ATA? For example, if someone was having a problem and I wanted to rule out any ATA glitches or firmware issues, what device could I give them that I could count on to always be a trouble free top performer that just plain works?