search for: aspendora

I am looking to replace all of the old "Bell" (POTS) phones in my home and office with IP phones. As you can imagine I don't have a huge budget to work with but I want phones that will provide acceptable voice quality and durability. There are basically three categories as I see it 1. satellite phones (low cost, low function) 2. primary domestic phone (good quality, POE capable,

I'm getting Number but when I look at the CDR database. I do see the name -----Original Message----- From: Lacy Moore - Aspendora [mailto:aspendora@gmail.com] Sent: Mon May 01 17:10:26 2006 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Spam? Re: [Asterisk-Users] CallerID Name problem Do you get caller ID number? If so, WAITing is not going to help, since you already get the info. If you get caller ID...

...-industry standard and reversed send and receive in the jack that they plugged the CAT5 into. Sure it works, sure it is easier, sure it is not the correct way of doing things. Thanks, Steve ________________________________ From: asterisk-users-bounces@lists.digium.com on behalf of Lacy Moore - Aspendora Sent: Sat 4/22/2006 2:55 PM To: Paul Mahler; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: Pinouts for T1/E1 crossover cable WAS "RE: [Asterisk-Users]whatcable to connect a legacy PBX to a TE410P ?" at&t (formerly SBC, formerly Southwestern Bell, formerly AT&a...

Dear All, I am trying to find a way to stop people who use phones after business hours (a policy the company wants to implement), we have cisco 7940 and 7910 phones and sadly they don't have a phone lock password system (on these ciscos it locks config menu changes but not the calls but the cisco 7920 has this feauture). So I was wondering is there a way to make this happen in asterisk??

Hi all, I have just successfully configured a Cisco 30VIP to work with my Asterisk server. I have seven of these phones new and would like to deploy them. I am wondering if anyone has this phone deployed with Asterisk and can suggest configuration of the various buttons, etc. (Bare with me as I am new to Asterisk.) Thanks, Chris

Hello, In voicemail.conf, it's possible to edit the voicemail message, but when I define a pager email address, I get the message from "Asterisk PBX", and the content is fixed by the system. Is there a way to manipulate this message, as well? Thanks, David -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have a new asterisk installation (1.4.2) that is working fine with SIP. Now I'm trying to add 2 cisco ip phones (7960) running SCCP (latest chan_sccp). I have the phones booted, and the tftp directory all setup, etc. But the phones do not quite work right. When I lift the handset I only get a dial-tone 1 out of 5 or so times I try, though hitting the speaker button works. I can dial

I am looking to setup paging using the auto answer feature on the Grandstream GXP2000. I am thinking I will follow the method as described here: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page I will setup the 4th account on the phone to auto answer. Does anyone else have a method that works better? I also looked at the allpage AGI written on Voip-Info. But it seems

I agree. I haven't had a problem using CAT-5, even for long runs, however it's not a real T-Carrier cable and I didn't know how old the PBX is. Paul >I have not in my experience seen any problems with using a Good Quality >Cat5 vs. Cat 3 (telco standard) cable for X-connects. YMMV, but you >should be fine. As far as the shielding goes, I use UTP cables and >Connectors

I set up the Asterisk for my company which is a business center, I will assign a specific telephone number to my client that uses my serivces. All of their incoming calls will be first picked up by the receiptionist, can I disply the company name instead of the called number on my receptionist's telephone display, so that she can answer the call with the right identity at once... Regards, ML

...st want to confirm that someone is using this and it is working for them. If that's the case, then I know it is somewhere in my setup. If no one else has been able to get it to work, then it may not work correctly to begin with. I'm using the latest stable version 1.2.7.1. -- Lacy Moore Aspendora, Inc. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060504/a4f24da8/attachment.htm

While I'm frantically scouring this list, does anyone have any information about getting BLA (busy line appearance) working on Cisco 7960? The last I heard was that this was unsupported in Cisco's SIP firmware

Can someone provide an example of how to use HINT priority with Cisco 7960/SIP phones? I don't fully understand what exactly the hint does, but I believe it mimics a legacy PBX's bridge-appearance function. Is this correct?

I'm looking for the best recording I can get of Allison saying "Steve" or "Ian". I found gsm recordings of both out there but was looking for something higher quality. Can anyone point me in the direction of a WAV or ULAW recording of those names? Thanks in advance Steve

I'm wondering if anyone else has experienced this. Up until a few days ago, when accessing the CLI from my terminal program (Private Shell), the output was in color. I haven't upgraded, rebuilt, or to my knowledge, changed anything in Asterisk that would change this. My terminal settings were the same as well. I have two computers that I access the CLI regularly on, and neither show

Hi: Does Trixbox support ACPI and IO-APIC out of the box? My Trixbox server isn't seeing the mainboard's APIC. -Stephen-

Hi All, I have installed Asterisk 1.4.2 and have loaded ztdummy as I have no Digium cards. The problem I have is that MOH will not play. It starts and then stops. asterisk*CLI> zap show status Description Alarms IRQ bpviol CRC4 ZTDUMMY/1 1 UNCONFIGUR 0 0 0 I'm not sure if the above is correct.

I just noticed the Aastra 57i do something that I haven't seen before. I called from one phone (phone 1) to the 57i. I answered it. Then, I pressed Transfer and dialed the extension for the third phone (in this case a Cisco 7960 in Sip). I did not answer the Cisco, but noticed the caller ID was showing the Aastra (as expected). I hung up the Aastra to complete the transfer and noticed the

I think I'm missing a change between 1.2 and 1.4. When using 1.4 (so far 1.4.9, 1.4.13, and 1.4.14), music on hold is not working for transfers or parked calls. It does work when putting the call on hold. If I revert back to 1.2.23 using the same config and same music on hold files, it works. I've looked at the sample config files for 1.4 and nothing seems to jump out at me as to what

...s Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <479F73A4-7938-4EFD-99B8-EEED7E4ED70C@nosignal.org> > Content-Type: text/plain; charset=US-ASCII; delsp=yes; format=flowed > > > On 1 Feb 2007, at 14:14, Lacy Moore - Aspendora wrote: > > > On 2/1/07, Andy Davidson <andy@nosignal.org> wrote: > > > What I would expect to happen, is that Asterisk would transcode > > > between the ulaw/alaw party, and me, wanting to listen via > g729. Is > > > this what *should* happen ?...