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> -----Original Message----- > From: Andrew Kohlsmith [mailto:akohlsmith-asterisk@benshaw.com] > Is the channel physically being hung up before the * tone is heard? Good question. If it is, Asterisk doesn't detect it -- the PBX doesn't support Kewlstart-style disconnect notification. The sequence I hear on the extension, when I plug in an analog phone,...

Has anyone has issues with echo using a Wildcard with a PRI from a major Canadian Telco? (Bell, Telus, AllStream, Sprint, Group Telecom). We are using a T1 from GT that is giving use annoying echos whenever a SIP/IAX2 client calls a local analog line. Calling cells phones is no issue since its digital. Regardless, there should be no issue with echo on a PRI at all. NOC at GT is telling us

Hi: I use shared internet bandwidth and the calls are very clear from around midnight till about 4 pm when it goes bad after that. Is there a way to boost the internet bandwidth for Asterisk at the peak time? Thanks Discover Yahoo! Have fun online with music videos, cool games, IM and more. Check it out! http://discover.yahoo.com/online.html

> -----Original Message----- > From: Andrew Kohlsmith [mailto:akohlsmith-asterisk@benshaw.com] > Sent: Tuesday, 25 November, 2003 08:56 > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] Picking a channel (FXO port or SIP) for outbound calls > > > > Yep, we use it for international calling. Works great: > > exten => _...

...elf can associate ring tones with specific callers from its contact directory too. Hopefully, someone will chime in with more precise (and helpful) detail before I return to the office, but I hope my reassurance is helpful anyway... Date: Thu, 9 Feb 2006 10:49:08 -0500 From: Andrew Kohlsmith <akohlsmith-asterisk@benshaw.com> Subject: [Asterisk-Users] Polycom IP501 with Asterisk - distinctive ring? To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <200602091049.08777.akohlsmith-asterisk@benshaw.com> Content-Type: text/plain;...

At 1/16/04 7:25 AM, Andrew Kohlsmith <akohlsmith-asterisk@benshaw.com> wrote: >That's pure bullshit -- I use software RAID *specifically* because I value >my data. I don't want to buy two hardaware RAID controllers to have one >sit on the shelf just in case the first dies... and if the second dies >you're SOL beca...

...ion. I spoke to the telco and they said everything looks good on their end, and since this a production T1 it doesn't need turning up. Now that I know support comes with these cards, I might give Digium a shout. :) Terence Date: Mon, 1 May 2006 07:20:51 -0400 From: Andrew Kohlsmith <akohlsmith-asterisk@benshaw.com> Subject: Re: [Asterisk-Users] PRI Issue: D-Channel woes To: asterisk-users@lists.digium.com Message-ID: <200605010720.51416.akohlsmith-asterisk@benshaw.com> Content-Type: text/plain; charset="iso-8859-1" On Monday 01 May 2006 01:42, Terence Burnard wrote:...

Hello, I am just asking this because I am note sure if the problem is on my side or not, I saw some comments on SIP realtime today so I was wondering, has anybody has SIP realtime working with a softfone ? If yes, please confirm, that would give me a light. My previous message to the list is below. Thanks. Frederic ----- Original Message ----- From: Frederic Jean To:

On Friday, June 10, 2005 3:16 AM, Andrew Kohlsmith [SMTP:akohlsmith-asterisk@benshaw.com] wrote: > On Friday 10 June 2005 04:08, Terry H. Gilsenan wrote: > > Received: from source ([81.56.129.44]) by exprod5mx8.postini.com > > ([64.18.4.10]) with SMTP; Fri, 10 Jun 2005 00:29:16 PDT > > > > Your MTA claimed it was called "SOURCE&quot...

RE Message: 5 Date: Sat, 7 May 2005 23:18:46 -0400 From: Andrew Kohlsmith akohlsmith-asterisk@benshaw.com Subject: Re: [Asterisk-Users] At home Asterisk via Broadvoice? To: asterisk-users@lists.digium.com On May 7, 2005 11:04 pm, John Stegenga wrote: > Broadvoice will give me 2 lines, with 2 phone numbers each - distinctive > ring - for a reasonable fee... Please do a googl...

Hi all I been looking into the whole code strugture of chan_iax and i see there is a option to specify the refresh rate of registrations: But there is no code to actually load this from the config file thus i changed the setting in chan_so.h, and recompiled. But still my refresh rate is 60 sec. I need to get this down to 15 sec (nat /pat firewall issue) any ideas? thanks Liaan

...X/SIP Client (Dan) 9. Re: USB handsets/headsets?? (Dan) 10. Re: Anyone using * in a live production environment? (Andrew Kohlsmith) 11. Re: Anti-Ex Girl Friend logic (was Re: [Asterisk-Users] Setcontext based on CID...) (Chris Hirsch) --__--__-- Message: 1 From: Andrew Kohlsmith <akohlsmith-asterisk@benshaw.com> Organization: Benshaw Canada To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] a bit frightened, guys Date: Thu, 6 Nov 2003 10:34:49 -0500 Reply-To: asterisk-users@lists.digium.com > But isn't it likely that many people call 911 simultaneously in cas...

Sorry to have had to post this, But I need urgent help with configuring one adit 600 I picked up from e-bay. Issues. I cannot access the console port, I am using HyperTerminal with settings VT100, 9600, 8-N-1 I also do not have any user-manual so I am kind of stuck. Any help in getting me started would be really appreciated. Any default settings like Ethernet port address, that can help me

*spews coffee over keyboard* - FUNNIEST - THREAD - EVER - Also one of the most insightful. Teddy, your gmail invite is on the way.

I have an issue with call parking and hope there is some undocumented feature for this. ;-) We are replacing our legacy PBX with asterisk, but to save money over time (handsets and network), I am trying to maintain the use of our legacy PBX. Asterisk extensions can not use the call parking features (not usable over trunk cards) of the old PBX, so I have to get the old PBX to use asterisk's.

Some guy on eBay is trying to sell the Grandstream Budgetone Phone 101 as the 102D. And to make matters worse he starts the bid at $90.00 Beware. http://search.ebay.com/search/search.dll?query=sip+phone&ht=1&sosortproperty=1&from=R10&BasicSearch= -- Costas Menico Meezon Software Corp 201-224-8111 costas@meezon.com --

Hello, I resolved my echo issue using grandstream/estara etc etc sip phones and wcfxo interfaces from digium. I swapped out my via kt400 based msi kt4vl motherboard for an asus p4pe? i845? based motherboard and the echo has completly gone away along with aggressive suppressor option in the makefile. I hope this helps others. Brian J. Schrock Anistone Technologies, LLC 6926 Avery Rd. Dublin, OH

Assuming the price of an ADSI screen phone (say, Aastra 390) was the same as an IP screen phone (say, Cisco 7960) and someone was setting up an * server for their 20 employees (each of whom would have either an ADSI or IP phone on their desk), would there be advantages to using the ADSI phones over the IP phones, or vice-versa? For discussion, let's assume that the hardware needed to

Seeking recommendations on handsets for use with Asterisk. I've been looking at the Aastra 480i CT because of its cordless handset and also the new Linksys SPA-942. Anyone using either one of these with comments on them? Any other thoughts on good reasonably priced handsets? This is for just a couple of people who work from home offices and will be connecting to an Asterisk server hosted

...for their call to go to. The best solution for my company is to be like the old systems and allow people to define the exact extension that they park their calls on, and if that park extension is busy it would give a busy signal. MATT--- -----Original Message----- From: Andrew Kohlsmith [mailto:akohlsmith-asterisk@benshaw.com] Sent: Monday, October 13, 2003 6:59 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Call Parking and Paid Digium software modifications > Is there an underlying reason you want to do this? Because if a call is > already parked on 701 and you transf...