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We're using Grandstream GXP-2000 with programmed buttons to the first 5 parking lot extensions. When a call is parked, whichever parking lot extension it's parked on lights up red. We've never used the "announce" part and I'm wondering if there's an option I can't seem to find to disable the announce so the transfer happens faster. Thanks for any help, Ken

...our area, get an ATA, a TDM card, and an Aastra/SNOM/Polycom/Granstream to play with. You can order the Aastra phones from your local electrical supply company (the place with a long counter and lots of electricians drinking coffee ordering their parts.). Andrew On 10/31/06, Ken Williams <ken@intermountainelectronics.com> wrote: > > > > I knew I should've waited til tomorrow to send the e-mail so I could > have a nights thought on the subject. > > That being said, scratch the FXO/FXS thing, what I really picture is > someway of passing proprietary information through the Asteris...

Hello, I configured both asterisk and grandstream 2000 accourding to howtos on the web.. And everything seems working fin. But if i reload asterisk grandstream stops working with BLF. I need to restart the phone to enable BLF again. Any clues??

Came upon a problem today that I thought I'd see if it's by design, if I'm missing an option somewhere, or if my fix is the way to fix it. We setup a remote location with a server, same as we've done with others, but for some reason when they would transfer an outside call anywhere it would pause for a few seconds and hang up the line. Well, after spending most of the day on

I am trying to set up the BLF on a GXP2000. Currently what I have is extensions.conf: [globals] polycom430=SIP/101 [internal] exten => 101,1,Macro(voicemail,${polycom430}) [macro-voicemail] exten => s,1,Dial(${ARG1},10,tT) exten => s,2,VoiceMail(u${MACRO_EXTEN}@default ) exten => s,102,VoiceMail(b${MACRO_EXTEN}@default) [ext-local-custom] exten => 101,hint,${polycom430}

Every few seconds I get the following message: == Parsing '/etc/asterisk/manager.conf': Found == Connect attempt from '127.0.0.1' unable to authenticate I'm trying to track down where it's coming from. I've used TCPDUMP & NGREP to monitor 127.0.0.1, no data's flowing. I've tried loading Asterisk with no modules, tried loading with a naked

After about one weeks time I've gone from no VoIP to a completely configured system for two of our offices to be able to page/communicate interoffice as well as handle existing PSTN communications (okay, waiting onf hardware for the PSTN side and I've likely jinxed myself now). I was sweating getting the two boxes talking to each other and I knocked that out in no time without even

I posted about this problem last week and thought it was a combination of SIP/ZAP causing issues in Asterisk. Since then I've realized it's only the SIP channel that's hanging. When this happens a call can still come in and hit the IVR, but no one can connect to the server from a SIP client. I tried reloading chan_sip.so today when this occurred, and I tried unloading chan_sip.so

...our area, get an ATA, a TDM card, and an Aastra/SNOM/Polycom/Granstream to play with. You can order the Aastra phones from your local electrical supply company (the place with a long counter and lots of electricians drinking coffee ordering their parts.). Andrew On 10/31/06, Ken Williams <ken@intermountainelectronics.com> wrote: > > > > I knew I should've waited til tomorrow to send the e-mail so I could > have a nights thought on the subject. > > That being said, scratch the FXO/FXS thing, what I really picture is > someway of passing proprietary information through the Asteris...

I've been doing a lot of reading over the last few weeks on Asterisk, and will be implementing a test system this week to play with. I've got two questions in regards to the ideal implementation for our company. First, has anyone written any drivers to interface with proprietary phones? Specifically we have a comdial system and if we could use our existing 35 phones instead of having to

About once a week or so my Asterisk box stops responding to all phones. I can pull up the console, do whatever I want at the CLI but the only way to get things working again is to restart Asterisk altogether. I finally cranked verbose & debugging way up (and watched my log files go from 1mb/day to 100mb/day), but below I believe contains my problem. The next line is 1.5 minutes later where I

Below is a copy of my log, zapata.conf & extensions.conf that relate to the ZAP lines. Basically when we dial out it takes on 10-12 seconds before the ZAP line actaully picks up. I'm hoping to find out what the cause is for this as it's causing user grief with extremely long connect times, and I believe it may be causing issues of cross lines (an outgoing call gets mixed with an

SIP channel hang ups are progressively getting worse and I'm really grasping at straws here trying to find out what the cause is. The problem start, once a week or so the SIP phones couldn't communicate with the server, though there was no error message on the server and everything appeared fine on the server. It's now doing it multiple times a day and I fear having to go back to our

I was planning on using a TDM400P with 3 FXO & 1 FXS, with the 1 FXS being used for a fax machine. It now appears that Digium doesn't support this, are there other manufacturers anyone can recommend that will support it? Has anyone used a TDM400P in this setup and had it work without much issue? Thanks for the help, Ken -------------- next part -------------- An HTML attachment was

I've installed Grandstream GPX-2000 phones and have successfully enabled one of my buttons to use Asterisk BLF for an extension. I can tell when this extension is available, is being rung, or is on the line. I'd like to do the same for my Zaptel channels, to be able to see when a line is onhook, ringing or offhook. I tried the following but alas, it doesn't seem to be working:

A day of banging my head against a wall and spamming this list is about done.... I've got everything working beautifully, and I'm ready to go full out and implement all across the board...save for one stupid little thing. We have 6 phone lines and I'd like the GXP-2000 to show the status of these 6 phone lines down, as well as allow a user to pick a specific line, the right hand

1. We just dial the extension directly and have speed dials setup for the first 6 parked positions. We don't use *8 at all. 2. Change the config on the phones under Account to "Send DTMF via RTP (RFC2833)" -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Noc Phibee Sent: Thursday, February 08,

Is there anyway of storing an incoming calls CallerID on a parked call and having it restored when someone picks up the parked call? I've tried storing the CID as a global variable and restoring it in my dialplan, and while NoOp shows it working, the phone ignores it and uses the parking lot extension for callerid instead. I believe this is because the phone is calling out instead of a call

I used to run Asterisk 1.4.4 but had to revert back to 1.2.13 to minimize a bug we were coming across. 1.4.5 looked promising, but the hints are broken and making it so I'll likely have to go back to 1.2.13 until I get the hints fixed. I'm using Grandstream phones & hints on the parked extensions. I should also clarify that when I upgraded versions, I renamed all Asterisk folders

Just a quick thanks for all being here. I started to type up a message and realized my problem, so instead I'm saying thanks for all the good information you all pass through my mailbox every day and giving me a place to realize my error before I even ask the question. -------------- next part -------------- An HTML attachment was scrubbed... URL: