Displaying 20 results from an estimated 9000 matches similar to: "IP 500/600 1.1.0 Firmware"
2004 Jan 30
9
Adtran 750 DID question.
Hello All,
I've mostly solved my DID problem from a few days ago. Apparenly the lines weren't configured properly. Now heres the next question. 12 E&M wink lines from telco. I have them all plugging into an Adtran 750 with FXS cards. The Adtran ports are configured DPO. How do I signal this from Zaptel. I have them setup E&M in zaptel.conf and EM_W in zapata.conf. They
2004 Jan 08
3
Progress on the Polycom front...
Hello,
Good news on the Polycom front for those that are interested. It looks like
we may get a dedicated Engineer for Polycom/Asterisk!!! Happy Day!
Here's the message I got tonight:
Matt:
I heard back from our VP of Engineering- she is prepared to have an
individual dedicated to working on the Digium- Asterisk project.
Can we discuss again Friday or mid next week?
Scott Willard
2004 Apr 07
6
dreaded Caller*ID failed checksum
Caller*ID used to work as some point, but I can't seem to get it going
these days. The card is a x101p. I've tried going up and down the
rxgain scale. Can the txgain effect it at all? When I plug in a phone
into the line with a splitter it can decode caller id with no problems.
Reading through the mailing list archives hasn't given me any
move clues. Any ideas?
2003 Dec 26
2
Polycom Sip Registration
Hello,
Has anyone on the list been able to successfully setup a Polycom
Soundpoint 500 IP phone? I am getting failed registrations, and the
Polycom documentation is not very precise. Their web interface isn't
helping much either.
Thanks in advance,
Brent
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2003 Oct 29
2
Polycom SoundPoint IP 500
Hello all,
Has anyone used the SIP version of this phone with Asterisk?
I see Polycom has a H.323 and MGCP version also, does anyone know if
you flash the phone to swith protocols?
Thanks in advance for the info.
Ed
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2003 Dec 30
3
SIP phone as intercom
(new asterisk user - currently setting up Polycom IP600 phones)
Does anyone know if it's possible to make a sip phone instantly pick up
on speakerphone when a particular call comes in? Eg so that you can
quickly bother someone across the office without making them reach for
their phone?
2004 Sep 14
2
Mitel 5010 +5220
I know this is not strictly an asterisk issue but it is related I guess.
Just to let you know that after many calls to Mitel the consensus is that
they will be releasing a new version of the 5220 that is dual boot (minet
and SIP) next week or the week after. This firmware will only appear on
NEW phones manufactured after the release date (no one could confirm but
the 23rd of sept was
2005 Feb 01
4
astGUIclient users should not upgrade to Asterisk 1.0.5
Hello,
Just confirmed this on my end, because of the massive changes that have been
made to callerID handling in asterisk 1.0.5 many of the features of the
astGUIclient suite will not work on this new version. The latest stable
version recommended is Asterisk 1.0.3. We will work on trying to find ways
around the new callerID rules that the asterisk developers have put in place
and hope to have
2004 Apr 27
3
New ASTGUICLIENT released: 1.0.1
Hello,
We've released another update to our Asterisk GUI Client suite:
http://astguiclient.sf.net/
Screen shots: http://astguiclient.sourceforge.net/screenshots.html
The client suite runs on both Windows and UNIX and includes a dialer
(the suite is not an asterisk configuration tool)
In addition to the usual bug fixes, this is mostly an update for the
VICIDIAL dialer application.
2004 Aug 24
7
SMP Performance
We're looking at implementing Asterisk in our department in the near
future, we're looking at anywhere from 15-25 extensions. The machine we
were looking at running this on was a Quad Xeon 450mhz (2MB L2 Cache) w/
1GB of ram. I've heard bad things about running Asterisk on SMP
machines? Would we be running into any performance issues with this
machine?
Tim Jackson
Network Engineer
2004 Jan 30
3
Call quality questions
Our basic system is as follows:
P4 3.0 Ghz w/ HT, 1GB PC3200 RAM, 120 GB HDD, RH 9.0 OS, * from CVS
several weeks ago, working OK for routing, VM, and AA, calls in on
separate PSTN lines to Adtran TSU 600, into * server through T100P card.
The hardware is not taxed at all with little over 20% proc utilization
ever, low mem use, etc. All Phones are SNOM 200's with various firmware
revisions
2003 Nov 05
3
New Phone Review: Clipcomm 101
Hello,
I have received yet another new phone today, the ClipComm 101
(http://www.clipcomm.co.kr/eng/e_product/e_product_voip_ip_phone.html)
I bought it for $165 directly from the Korean Manufacturer(No US distributer
yet). Here are the features:
- Built-in NAT functionality, you can switch from Hub to Nat, great for home
DSL/Cable users
- This includes some limited port forwarding
2003 Jul 10
3
T1 config for robbed-bit E&M AMI
I have a couple of live T1s sitting around and they are not ISDN(like most
of the people that are using Asterisk seem to be using), they are regular
old 24 channel, robbed-bit, E&M wink start, D4AMI T1 circuits.
Can I get these T1s to work with a T100P Digium card and asterisk?
Searching through the lists and the documentation I haven't seen any
examples of how to configure this kind
2004 Mar 16
7
PRI Errors
I just had the same exact problem this morning. The only thing I've done in the last couple of days is update update zaptel. I rolled back my zaptel to 2/11/04 from 3/8/04. And kept my libpri from 3/8/04. I never had this error before updated. I had other issues, but not this one.
-sb
-----Original Message-----
From: asterisk-users-admin@lists.digium.com
2004 Aug 30
7
Polycom SoundPoint IP 300 Configuration
I just got a Polycom soundpoint and I set it up using the phone and web
based admin.
I cant seem to figure out the config files and they are confusing me
greatly and I dont have time for it :)
Some things are odd, like on every reboot it seems the volume I set is
reset? is there any way to fix that. And the ringer seems low. - Even
all the way up
Anyone willing to point out a good asterisk
2004 Jan 16
1
doublehash patch doesn't work in asterisk 0.7.1
Hello,
I was using the doublehash.patch that Iain Stevenson had created back in
August to change the transfer key from a single hash "#" to a double-hash
"#". It always patches properly, but when I went from CVS 2004-01-12 to
Asterisk 0.7.1 it doesn't seem to work anymore. I've attached the patch to
this email and I use the following command to patch it:
patch -p1
2005 Feb 26
3
listening to gsm files
Hello list,
I am having trouble listening to GSM files created by Asterisk using a
browser. I am noticing that some of my users succeed in listening to them
and some others don't. I guess it is something of a codec problem that
does not seem to be installed on all machines (though they are all WinXP).
Anybody knows what one should do to listen to GSM files?
I send files through the
2004 Jul 23
4
Doublehash transfers
Hello,
I recently tried an upgrade of CVS on my test server today and found that
the res/res_parking.c file is completely gone. This is where I had to go
into the code every time I do an upgrade and change the code to allow for
doublehash transfers instead of single hash transfers:
That means that you need to hit the pound key twice to initiate a
transfer instead of once. Because of our inbound
2003 Oct 17
2
Polycom IP 600 phone
Hello,
I have finally received the details from Polycom to get into the backend
configuration of their SoundPoint IP 600 SIP VOIP phone. The phone is quite
nice looking but the configs are very sparse, not even a place for a
secret(password) field in their SIP registration section.
If anyone else has one of these and needs the passwords to get into the back
end configurations, just send me an
2003 Nov 13
10
Graphical Interface
Hello. Was just curious to know if anyone is working on a graphical
interface to Asterisk using X windows, or something else similar.
Thanks!
David
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