similar to: Pager Voicemail Message

Greetings; I've got a situation where I really need to be able to select from a number of various (body) textx for email/pager messages when voivemails are left, but can't seem to figure out how to change them on the fly. Now, I know there are pre-defined defaults, which can be overridden by entries in the [general] section of voicemail.conf, but that does not get me there. I need to be

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,

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??

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'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

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

The "telco guys" probably did something non-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

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

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

Hi, I have set emailsubject in voicemail.conf as follows: emailsubject=New Voicemail from ${VM_CALLERID} in Mailbox ${VM_MAILBOX} This works fine, but the pager e-mails come through with "New VM". I would like the pager e-mail to be the same subject as above as I get this as an SMS to my mobile phone. I thought this would be config'able but I can't find any mention of it on

The following sequence of commands (for example): >my.data<-rnorm(10000, 5, 1) >my.data fills my pager buffer beyond its capacity. When I type "my.data", the pager displays approximately the bottom 7000 cells of the file, but "loses" the first 3000. Is there a some setting that increases the size of the pager buffer, or is this a limit in the Windows version?

Hi; Before I tell a customer that this would require custom development I figured I would ask here. Does Asterisk support pager notification of new voicemails out of the box? Or do I need an AGI script to do that? Also, if I want to call a number from an automated program in Asterisk and get the DTMF tones entered by the user on the other side, is there an easy way to do this? Best

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

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

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

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.