similar to: Polycon IP 300 SIP vs Grandstream BT-101 Deployment

Ah... Looks like I have the 500's...Sorry. Ty Purcell -----Original Message----- From: Derek Listmail Acct [mailto:listmail@oversimplified.com] Sent: Thursday, September 09, 2004 12:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycon IP 300 SIP vs Grandstream BT-101Deployment > I plugged it in, configured it, and it works great. I

Hi, all I am in Australia and I have to setup Asterisk in few offices. There will be IP phones in each office and I must be able to call between offices. I need actual handsets. I need "standard" handsets to be used by people. Those must support features like CID, call forward, etc. --- your normal office feature set. Also I need some sort of more complex handset to be used by

Basically I've found that the 2.0.3 version of Samba is not supporting shadowed password files on SCO Unix 3.2v4.2. It appears to be doing it on SCO Unix OSR5 however. This shows up both in smbd and swat. We found this out by playing with the newest version on a SCO Unix OSR5 box. I recompiled the software on that box for a SCO 3.2v4.2 box but was unable to get the resulting binary to work

Hi all, I am a beginner R user and need some help with a simple loop function. Currently, I have seven datasets (TOWER1,TOWER2...TOWER7) that are all in the same format (same # of col and headers). I am trying to add a new column (factor) to each dataset that simply identifies the dataset. Ultimately, I would like to merge all 7 datasets and need that column to identify what rows came from what

Is anyone using this combination successfully? I have a dell 500sc running rh9 and asterisk 1.0rc1. It is configured with an x100p. I have a Sipura SPA-2000, laptop with Xlite and a Grandstream Budgetone BT-101. I have signed up with Voipjet (they use iax2). I also have an FWD number via iax2. I can sucessfully call back and forth to all devices via zap, sip, and fwd. I can successfully

Recently our Unix Administrator installed Samba v.2.07. It was configured to attach to our NT domain. When this occurred we started to experience problems with user authentication's to the domain and it appeared that the "Samba" server had become the master browser. Have you seen this problem occur at other sites? If so, what is the work around? We have been using Samba v2.06

Hi all, I'm having a really frustrating time with a bunch of BT-101 phones. They've been trouble-free and working very well for the past several months. A couple of days ago, some of the phones (but not all of them, yet) have started acting very strangely. All phones are running firmware 1.0.6.7, and are identically configured (except for the user/authenticate/password things) on both the

Hi- I'm at my wit's end trying to get a Grandstream BT-101 POS to register on my * server. Asterisk version is 1.2.1. GS Firmware is rev 1.0.6.7. Basically, I've setup the phone following the instructions at voip-info.org, and it registers for about 10 seconds, then after receiving the SIP NOTIFY from the * server, goes into "flashing display" mode, which indicates some

Hi, This is a weird request, but does anyone have a Snom 360 monitoring extensions for BLF on several Asterisk servers accross a network? Alternatively, can anyone give me a pointer as to how to setup a Snom 360 to monitor an extension not on it's own server? My scenario is that I have a main site which will have its own server (for storage of call recording data etc because the remote

I am having a problem with SJphone registration, having read the list and wathced it for a while for similar problems. I just can't seem to figure out the problem. I tryed to follow a tutorial from http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+sjphone, but in SJphone (SIP tab), I can't find the following setting. Use local outbound proxy - checked. Proxy IP Address:

Hi, i am new to ajax and new to script.aculo.us. I created a blog, http://www.zorpal.com/blog/, using some of the script.aculo.us features and commands. My sites works great in firefox, which everyone should be using, but if you open it in IE it kills it. It only kills it after a script.aculo.us motion has been called and finishs it''s execution. i really like the layout and the visual

Hi, I was wondering if anyone else has a similar setup and can suggest settings for the jitterbuffer: I have a client with an ADSL connection at site A & B with site A being dedicated to voice and having no Asterisk server, site B combining voice and data with traffic shaping and housing an Asterisk server. There seems to be packet loss / jitter on this connection and I wanted to know

I had this happen to me on FC4 using 3.0.2-2 Xen. I updated the various external dependencies and rebuilt xen and tools and was able to sucessfully install FC4 in a hvm. I will give winxp a go later in the afternoon. C ________________________________ From: xen-devel-bounces@lists.xensource.com [mailto:xen-devel-bounces@lists.xensource.com] On Behalf Of Rene Purcell Sent: Friday, April

I can not get smbd to run. ====================================================================== [07:57:29] ~$ ps -A | grep mb 11330 ? 00:00:01 nmbd ====================================================================== I first noticed my problem when I tried restarting samba ====================================================================== [07:57:31] ~$ sudo /etc/init.d/samba

Howdy, I just did an "apt-get dist-upgrade" on my Debian unstable box, and noticed that the Asterisk version appears to be 1.0-1 in the unstable tree. I KNOW that 1.0 hasn't been released yet, so I am wondering who is responsible for the Debian packages? This will be VERY VERY confusing for people and it should be corrected ASAP. -rw-rw-r-- 1 1176 1176 1189696 Jun 7

Just come across a problem - we have sent out heaps of PAP-2 ATA's and just discovered that when joined in a conference they are choppy on the up leg (so the other users in the conference will hear them with a choppy sound) but the down leg is perfectly fine (so the end user can hear the conference participants perfectly). I have tested the same setup with different brands of ATA's

Hello, I was successfully using the BT-100 phone with CVS 11/10. Now that I've upgraded to 11/27, I can't place an outbound call. However the phone is registered and works well with inbound calls. Any suggestions will be appreciated. Thank you. Regards, Christopher

dear all, can anyone assist on how to enable callerID to appear on grandstream bt-100 sip phone. [1003] type=friend context=default username=1003 secret=**** ;fromuser=1003 ;callerid=John Doe <1234> host=dynamic nat=no canreinvite=yes dtmfmode=rfc2833 incominglimit=1 ;mailbox=1234@default ; mailbox 1234 in voicemail context "default" disallow=all allow=ulaw allow=alaw

I just noticed something when I 'sip show peers' from the CLI, I get the following: 6113/6113 x.x.x.x D N 255.255.255.255 62927 OK (66 ms) 6112/6112 x.x.x.x D N 255.255.255.255 50079 OK (160 ms) 6111/6111 x.x.x.x D N 255.255.255.255 60810 OK (141 ms) 6109/6109 x.x.x.x D N 255.255.255.255 51331 OK (151 ms) All of those

Good day all We have one Bt-100 that logs on to the server,works for a few min and then just starts loosing registration Jan 13 13:10:05 NOTICE[-1101505616]: chan_sip.c:7503 handle_request: Registration from '<sip:144@192.168.0.250>' failed for '192.168.0.145' Jan 13 13:10:05 NOTICE[-1101505616]: chan_sip.c:7503 handle_request: Registration from