Re: [asterisk-users] GXP-2000 DST Change

[Todd H]

Thanks for the info, Ken.  I was about to research this tonight.
  Todd


On Mar 12, 2007, at 12:53 PM, Ken Williams wrote:

In case it hasn't been posted before, here's instructions to get  
the correct time to show up on your Grandstream GXP-2000's:


1. Login to phone
2. Go to Basic Settings tab
3. Change Daylight Savings Time to yes
4. Change Optional Rule to 3,2,7,2,0;11,1,7,2,0;60 (this means  
change clocks the second sunday of March and back again the first  
sunday of November - i.e., the new savings times).

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Re: [asterisk-users] gxp-2000 configure line appearances

[Matthias Fechner]

Hello Cavanna,,

* Cavanna, Richard <[EMAIL PROTECTED]> [27-07-06 15:59]:
> The real thing that would help is a complete list of the configurable
> comands on the latest firmware so I can create the config file.

try that config file, works perfectly for me.

Best regards,
Matthias

-- 

"Programming today is a race between software engineers striving to
build bigger and better idiot-proof programs, and the universe trying to
produce bigger and better idiots. So far, the universe is winning." --
Rich Cook



## Configuration template for GXP-2000 firmware version 1.0.2.13


##
##  Advanced/System-wide Options
##

# Admin password for web interface
P2 = admin

# Silence Suppression. 0 - no, 1 - yes
P50 = 1

# Voice Frames per TX (up to 10/20/32/64 frames for G711/G726/G723/other codecs 
respectively)
P37 = 2

# Layer 3 QoS (IP Diff-Serv or Precedence value for RTP)
P38 = 48

# Layer 2 QoS. 802.1Q/VLAN Tag (VLAN classification for RTP)
P51 = 0

# Layer 2 QoS. 802.1p priority value (0 - 7)
P87 = 0

# No Key Entry Timeout. Default - 4 seconds.
P85 = 4

# Use # as Dial Key (if set to Yes, "#" will function as the "(Re-)Dial" key). 
0 - no, 1 - yes
P72 = 1

# Local RTP port (1024-65535, default 5004)
P39 = 5004 

# Use Random Port. 0 - no, 1 - yes
P78 = 0

# Keep-alive interval (in seconds. default 20 seconds)
P84 = 20

# Use NAT IP.  This will enable our SIP client to use this IP in the SIP 
message. Example 64.3.153.50.
P101 =

# STUN server
P76 = 

#-
# Firmware Upgrade 
#-

# Firmware Upgrade. 0 - TFTP Upgrade,  1 - HTTP Upgrade.
P212 = 0

# Firmware Server Path
P192 = 192.168.0.251

# Config Server Path
P237 = 192.168.0.251

# Firmware File Prefix
P232 =

# Firmware File Postfix
P233 =

# Config File Prefix
P234 =

# Config File Postfix
P235 =

# Allow DHCP Option 66 to override server. 0 - No, 1 - Yes. Default is No.
# When set to Yes(1), it will override the configured provision path and method.
P145 = 0

# Automatic Upgrade. 0 - No, 1 - Yes (checking every defined days). Default is 
No.
P194 = 1

# Check for new firmware every () minutes, unit is in minute, default is 7 days.
P193 = 10080

# Use firmware pre/postfix to determine if f/w is required
# 0 = Always Check for New Firmware 
# 1 = Check New Firmware only when F/W pre/suffix changes 
P238 = 0

# DTMF Payload Type
P79 = 101

# Syslog Server (name of the server, max length is 64 charactors)
P207 = 192.168.0.251

# Syslog Level (Default setting is NONE)
# 0 - NONE, 1 - DEBUG, 2 - INFO, 3 - WARNING, 4 - ERROR
P208 = 0

# NTP Server
P30 = 192.168.0.251

# Allow DHCP Option 42 to override NTP server. 0 - No, 1 - Yes. Default is No.
# When set to Yes(1), it will override the configured NTP server.
P144 = 0

# Distinctive Ring Tone
# Use custom ring tone 1 if incoming caller ID is the following:
P105 =

# Use custom ring tone 2 if incoming caller ID is the following:
P106 =

# Use custom ring tone 3 if incoming caller ID is the following:
P107 =

# Disable Call Waiting. 0 - no, 1 - yes
P91 = 0

# Lock Keypad Update. 0 - no, 1 - yes
P88 = 0


# Primary Account (Account 1) Settings


# Account Active (In Use). 0 - no, 1 - yes
P271 = 1

# Account Name
P270 =

# SIP Server
P47 = sip.mycompany.com

# Outbound Proxy
P48 = proxy.mycompany.com

# SIP User ID
P35 = 8000

# Authenticate ID
P36 = 8000

# Authenticate password
P34 = 

# Display Name (John Doe)
P3 = 

# Use DNS SRV. 0 - No, 1 - Yes.
P103 = 0

# SIP User ID is phone number. 0 - no, 1 - yes
P63 = 0

# SIP Registration. 0 - no, 1 - yes
P31 = 1

# Unregister On Reboot. 0 - no, 1 - yes
P81 = 0

# Register Expiration (in minutes. default 1 hour, max 45 days)
P32 = 60

# Local SIP port (default 5060)
P40 = 5060

# SIP T1 Timeout. RFC 3261 T1 value (RTT estimate)
# 50 - 0.5 sec, 100 - 1 sec, 200 - 2 sec. Default 100.
P209 = 100

# SIP T2 Interval. RFC 3261 T2 value. The maximum retransmit interval for 
non-INVITE requests and INVITE responses.
# 200 - 2 sec, 400 - 4 sec, 800 - 8 sec. Default 400.
P250 = 400

# NAT Traversal. 0 - yes, 1 - no, 2 - No, but send keep-alive
P52 = 0

# SUBSCRIBE for MWI. (Whether or not send SUBSCRIBE for Message Waiting 
Indication) 0 - No, 1 - Yes.
P99 = 1

# Proxy-Require (A SIP extension to enable firewall penetration)
P197 =

# Voice Mail UserID (User ID/extension for 3rd party voice mail system)
P33 = 88

# Send DTMF. 0 - in audio, 1 - via RTP, 2 - via SIP INFO
P73 = 2

# Early Dial

RE: [Asterisk-Users] GXP-2000 and Shared Line Appearances

[shadowym]

Thanks Dustin,

Can you give an example of how this would be used when a call comes in?

How is this different from the existing call parking feature in Asterisk?
Looks very similar except the existing Asterisk call parking automatically
assigns an extension and then announces it to the person who parked the
call.

Features.conf file included with Freepbx 2.1.1 using Asterisk 1.2.9.1

;
; Sample Parking configuration
;

[general]
parkext => 70   ; What ext. to dial to park
parkpos => 71-79; What extensions to park calls on
context => parkedcalls  ; Which context parked calls are in
;parkingtime => 60  ; Number of seconds a call can be
parked for (default is 45 seconds)

[featuremap]
;blindxfer => ##; Blind Transfer
;disconnect => **   ; Disconnect Call
automon => *1   ; One Touch Record
;atxfer => *2   ; Attended Xfer

> -Original Message-
> From: Dustin Wildes [mailto:[EMAIL PROTECTED] 
> Sent: Monday, June 26, 2006 11:55 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] GXP-2000 and Shared Line Appearances
> 
> Daniel Salama wrote:
> 
> > Dustin,
> >
> > any updates on this?
> >
> > Thanks,
> > Daniel
> >
> Hey Daniel!
> Yes - just posted the link.
> I appologize for the delay.
> 
> Here's the link to the forum as well, if anyone is 
> interested. This should compile and run on Asterisk-1.2.4 and higher.
> http://www.vecsector.com/phonecall/valet/
> 
> Enjoy!
> 
> 
> Dustin Wildes
> VecSector, LLC
> 1.912.422.7082 x101
> 
> 
> 
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Re: [Asterisk-Users] GXP-2000 and Shared Line Appearances

[Daniel Salama]

Beautiful. Will test and give you comments.

Nice work.

- Daniel

On Jun 26, 2006, at 2:55 PM, Dustin Wildes wrote:


Daniel Salama wrote:


Dustin,

any updates on this?

Thanks,
Daniel


Hey Daniel!
Yes - just posted the link.
I appologize for the delay.

Here's the link to the forum as well, if anyone is interested. This  
should compile and run on Asterisk-1.2.4 and higher.

http://www.vecsector.com/phonecall/valet/

Enjoy!


Dustin Wildes
VecSector, LLC
1.912.422.7082 x101

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Re: [Asterisk-Users] GXP-2000 and Shared Line Appearances

[Dustin Wildes]

Daniel Salama wrote:


Dustin,

any updates on this?

Thanks,
Daniel


Hey Daniel!
Yes - just posted the link.
I appologize for the delay.

Here's the link to the forum as well, if anyone is interested. This 
should compile and run on Asterisk-1.2.4 and higher.

http://www.vecsector.com/phonecall/valet/

Enjoy!


Dustin Wildes
VecSector, LLC
1.912.422.7082 x101

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Re: [Asterisk-Users] GXP-2000 and Shared Line Appearances

[Daniel Salama]

Dustin,

any updates on this?

Thanks,
Daniel

On Jun 23, 2006, at 1:07 PM, Dustin Wildes wrote:



shadowym wrote:

That feature is called Bridged (or Shared) line appearance.  That  
is one of
the things Asterisk cannot do and nobody seems very interested in  
making it
do that because it is apparently not easy.  There has been some  
talk about

implementing it but so far there does not seem to be any progress.



http://forums.digium.com/viewtopic.php?p=23974#23974
I will be posting the code later today.


--Dustin Wildes
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RE: [Asterisk-Users] GXP-2000 and Shared Line Appearances

[shadowym]

Here is the latest word on SLA that I could find.  Looks like it is quite a
ways off but at least it is on the radar screen.

http://lists.digium.com/pipermail/asterisk-users/2006-May/153385.html

> -Original Message-
> From: Daniel Salama [mailto:[EMAIL PROTECTED] 
> Sent: Friday, June 23, 2006 2:00 AM
> To: Non-Commercial Discussion Asterisk
> Subject: [Asterisk-Users] GXP-2000 and Shared Line Appearances
> 
> I have a client with 20 GXP-2000s. Everything seems to be 
> working fine. However, after a couple of weeks of use, the 
> client is having a hard time adjusting to the new IP based 
> phone systems and only misses one feature from their old 
> Lucent system.
> 
> That is, they had 8 analog lines before and all their old 
> Lucent phones showed a button for each line. So, it was easy 
> for anyone to say, pick up line 2 or anyone to see which 
> lines were in use.
> 
> Is it possible to use the GXP-2000 line buttons or extension 
> buttons to show the lines in use, shared by all phones. Since 
> the client is purchasing 8 "virtual lines", I have them 
> restricted in a call group and also with incoming and 
> outgoing call limits. Is it possible for all the GXP-2000s to 
> show that line 1 is in use, and so on?
> 
> Thanks,
> Daniel
> 
> 
> 
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Re: [Asterisk-Users] GXP-2000 and Shared Line Appearances

[Dustin Wildes]

shadowym wrote:


That feature is called Bridged (or Shared) line appearance.  That is one of
the things Asterisk cannot do and nobody seems very interested in making it
do that because it is apparently not easy.  There has been some talk about
implementing it but so far there does not seem to be any progress.
 



http://forums.digium.com/viewtopic.php?p=23974#23974
I will be posting the code later today.


--Dustin Wildes
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RE: [Asterisk-Users] GXP-2000 and Shared Line Appearances

[shadowym]

That feature is called Bridged (or Shared) line appearance.  That is one of
the things Asterisk cannot do and nobody seems very interested in making it
do that because it is apparently not easy.  There has been some talk about
implementing it but so far there does not seem to be any progress.

I was actually quite surprised to learn that it didn't do that. I am also
surprised that there doesn't seem to be much interest by developers in
general to make it do that even thought it is a deal killer for a lot of
people.

I would really like to see that feature myself as I have had to turn away
potential business because of the lack of that one feature.  If I could
write code I'd try do it myself. 

> -Original Message-
> From: Daniel Salama [mailto:[EMAIL PROTECTED] 
> Sent: Friday, June 23, 2006 2:00 AM
> To: Non-Commercial Discussion Asterisk
> Subject: [Asterisk-Users] GXP-2000 and Shared Line Appearances
> 
> I have a client with 20 GXP-2000s. Everything seems to be 
> working fine. However, after a couple of weeks of use, the 
> client is having a hard time adjusting to the new IP based 
> phone systems and only misses one feature from their old 
> Lucent system.
> 
> That is, they had 8 analog lines before and all their old 
> Lucent phones showed a button for each line. So, it was easy 
> for anyone to say, pick up line 2 or anyone to see which 
> lines were in use.
> 
> Is it possible to use the GXP-2000 line buttons or extension 
> buttons to show the lines in use, shared by all phones. Since 
> the client is purchasing 8 "virtual lines", I have them 
> restricted in a call group and also with incoming and 
> outgoing call limits. Is it possible for all the GXP-2000s to 
> show that line 1 is in use, and so on?
> 
> Thanks,
> Daniel
> 
> 
> 
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Re: [Asterisk-Users] GXP 2000 - BLF and Hold/Hangup Answering

[Daniel Salama]

I had the same problem some time ago. Make sure call waiting is NOT  
disabled. This will make the phone receive more calls on the other  
lines.


- Daniel

On Jun 23, 2006, at 1:29 AM, Corporate IT Solutions - Michael Dunne  
wrote:


I have a network of GXP 2000 phones and would like to know if there  
is a
way to configure the phones so that if there is one person talking,  
and

another call comes in then they can hold/hangup that call and take the
incoming call.

At the moment, when a call comes in and the phone is offhook, then  
that

phone is completely unavailable for that ring session, any call coming
in after that call will of course ring.

Is this limited to the GXP series or does the SNOM phones fix this,  
etc.


Any advice is appreciated of course.
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Re: [Asterisk-Users] GXP-2000

[Mike Fedyk]

Kristian Kielhofner wrote:

Mike Fedyk wrote:
I happen to have asterisk running as a router, so I use it doing QoS 
with tc (traffic control) and wondershaper set to prioritize based on 
port ranges.  I sent a patch to the debian bug tracking system a 
while back with a few improvements -- I should check on that.  It 
basically prioritizes smaller packets before larger packets with ~8 
levels of priority and groups of sizes for the packets.  Just doing 
that automatically handles 80% of the need for prioritization without 
specifying port ranges for the sip rtp packets.


Mike



Mike,

Have you tried AstShape?  Shapping based on port ranges is totally 
hit or miss.  TOS is the way to go:


http://www.krisk.org/files/astlinux-i586/usr/sbin/astshape

Comment out the . /etc/rc.conf and you should be okay!


Actually the above is wrong.  I don't use port ranges at all, just 
packet sizes.  It allows me to blast away with p2p, interactive ssh and 
scp file copies all while having two g.711 and one g.729 voip 
conversation going on a dsn connection with a 384Kbps upload speed.


It is based on the premise that smaller packets should have higher 
priority.  There will be exceptions of course, and empty classes have 
are there for that also.  For the common case, no configuration is 
necessary.


Give this one a try:
http://mikefedyk.com/wondershaper-pkt-size-classes

Mike
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Re: [Asterisk-Users] GXP-2000 1.1.0.13 Issues

[drew-asterisk-users]

Grandstream have acknowledged that there is a problem with 1.1.0.13 on
later phones (MAC's 00:0B:82:09:xx:xx I assume) and have advised me to
wait for the next firmware release.  So anyone with later phones (MAC's
00:0B:82:09:xx:xx), do not upgrade to 1.1.0.13.

On Wed, 14 Jun 2006 [EMAIL PROTECTED] wrote:

> I have had 2 GXP-2000 for a while now and been slowly following the 
> firmware releases made by Grandstream and am now up to 1.1.0.13.  This 
> version works really well on these 2 original phones (MAC's 
> 00:0B:82:06:xx:xx), so I went ahead and ordered another 2 phones (MAC's 
> 00:0B:82:09:xx:xx).  One of these I upgraded to 1.1.0.13 (it came with 
> 1.1.0.5) and pressed it into use.
> The Speaker phone does not work at all (no sound from the Speaker) and the 
> phone completely hangs doing a soft-reboot, other than that the phone 
> seems to work well.
> Unfortunatly I do not have a copy of 1.1.0.5 so cannot downgrade the 
> phone.
> Has anyone else noticed these problems, or does anyone have a copy of 
> 1.1.0.5.
> 
> -Drew-
> 
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Re: [Asterisk-Users] GXP-2000 Audio Quality

[Tzafrir Cohen]

On Sat, Jun 17, 2006 at 11:14:33AM +0100, Tim Panton wrote:
> 
> On 17 Jun 2006, at 07:53, Kristian Kielhofner wrote:
> 
> >Tim Panton wrote:
> >>Well, with 16 phones, it might be worth putting a
> >>'satellite' asterisk in their office, have it handle local
> >>transfers, and act as a protocol converter, talking sip to the
> >>phones and (trunked) IAX2 to the outside world.
> >>An embedded low power system would do fine.
> >>You might even get away with an nslu2, but I'm not sure
> >>it has the RAM for 16 calls.
> >>A better alternative is to get them to upgrade the DSL to 512 uplink.
> >>Tim.
> >
> > Neither the unslung nor the wrt support IAX trunking.  Zaptel does  
> >not compile on either of these architectures.
> >
> > No zaptel = no timer = no trunking/meetme/etc.
> 
> Just out of curiosity, is ztdummy on kernel 2.6.12.2 architecture  
> specific? i.e.
> would it care if it were on an armv5teb not on x86 ?

ztdummy on kernel 2.6 has tw implementations:

with USE_RTC defined (the default on x86, at least) it uses the rtc
clock of the system. This is availble on x86 and amd64. I don't know if
other architectures have anything equivalent.

Without it, it relies on HZ=1000 . That was the only possible value up
until 2.6.13 , so I guess that in the specific kernel you refer to it
should hold.

> 
> I understand that the _real_ zaptel modules will be much harder to port,
> I just figured that ztdummy might be easier.

Most other modules are PCI cards. Two others are USB. I don't know how
much architecture-specific are PCI and USB.

There are also ztdynamic and friends. In theory nothing prevents them
from being portable.

-- 
Tzafrir Cohen  sip:[EMAIL PROTECTED]
icq#16849755   iax:[EMAIL PROTECTED]
+972-50-7952406   
[EMAIL PROTECTED]  http://www.xorcom.com
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Re: [Asterisk-Users] GXP-2000 Audio Quality

[Tim Panton]

On 17 Jun 2006, at 07:53, Kristian Kielhofner wrote:


Tim Panton wrote:

Well, with 16 phones, it might be worth putting a
'satellite' asterisk in their office, have it handle local
transfers, and act as a protocol converter, talking sip to the
phones and (trunked) IAX2 to the outside world.
An embedded low power system would do fine.
You might even get away with an nslu2, but I'm not sure
it has the RAM for 16 calls.
A better alternative is to get them to upgrade the DSL to 512 uplink.
Tim.


	Neither the unslung nor the wrt support IAX trunking.  Zaptel does  
not compile on either of these architectures.


No zaptel = no timer = no trunking/meetme/etc.


Just out of curiosity, is ztdummy on kernel 2.6.12.2 architecture  
specific? i.e.

would it care if it were on an armv5teb not on x86 ?

I understand that the _real_ zaptel modules will be much harder to port,
I just figured that ztdummy might be easier.

Tim.


Tim Panton
[EMAIL PROTECTED]



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Re: [Asterisk-Users] GXP-2000 Audio Quality

[Kristian Kielhofner]

Tim Panton wrote:

Well, with 16 phones, it might be worth putting a
'satellite' asterisk in their office, have it handle local
transfers, and act as a protocol converter, talking sip to the
phones and (trunked) IAX2 to the outside world.

An embedded low power system would do fine.

You might even get away with an nslu2, but I'm not sure
it has the RAM for 16 calls.

A better alternative is to get them to upgrade the DSL to 512 uplink.

Tim.



	Neither the unslung nor the wrt support IAX trunking.  Zaptel does not 
compile on either of these architectures.


No zaptel = no timer = no trunking/meetme/etc.

--
Kristian Kielhofner
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Re: [Asterisk-Users] GXP-2000 addressbook

[Mike Fedyk]

Matthias Fechner wrote:

Hi Gareth,

Gareth Blades wrote:
  

No I dont believe so. The address book is a new feature as it is very
basic in my opinion and even editing it on the phone is difficult.

I would expect a web based editing feature to be implemented at some
point and once that is done it should be possible to do a mass update of
the phones.



ah ok, then I will wait for a new firmware :)
This is one of those times where you should be contacting the supplier 
you bought the phones from.  They should be able to get your message 
over to grandstream so they know what people want.  Other than better 
phones of course. ;)

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Re: [Asterisk-Users] GXP-2000 addressbook

[Matthias Fechner]

Hi Gareth,

Gareth Blades wrote:
> No I dont believe so. The address book is a new feature as it is very
> basic in my opinion and even editing it on the phone is difficult.
> 
> I would expect a web based editing feature to be implemented at some
> point and once that is done it should be possible to do a mass update of
> the phones.

ah ok, then I will wait for a new firmware :)

Best regards,
Matthias

-- 

"Programming today is a race between software engineers striving to
build bigger and better idiot-proof programs, and the universe trying to
produce bigger and better idiots. So far, the universe is winning." --
Rich Cook

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Re: [Asterisk-Users] GXP-2000 addressbook

[Gareth Blades]

No I dont believe so. The address book is a new feature as it is very
basic in my opinion and even editing it on the phone is difficult.

I would expect a web based editing feature to be implemented at some
point and once that is done it should be possible to do a mass update of
the phones.

On Thu, 2006-06-15 at 02:24, Matthias Fechner wrote:
> Hi,
> 
> is it possible to have one central phonebook and install it on the
> phone or using ldap?
> 
> Best regards,
> Matthias

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Re: [Asterisk-Users] GXP-2000 Audio Quality

[Daniel Salama]

That may not be such a bad idea. I've read people trying to put  
Asterisk on a WRTG54 or something like that. Would that be good? I  
guess I could do SIP in the office and trunk via IAX2 and save on  
bandwidth plus internal calls would be local.


I tried to upgrade them to 512K but because they're borderline to the  
18K feet, the best BellSouth can offer them is 256K. I'm talking to  
Comcast to see if they can get their broadband service which can go  
up to 768K.


Thanks,
Daniel

On Jun 14, 2006, at 12:45 PM, Tim Panton wrote:


Well, with 16 phones, it might be worth putting a
'satellite' asterisk in their office, have it handle local
transfers, and act as a protocol converter, talking sip to the
phones and (trunked) IAX2 to the outside world.

An embedded low power system would do fine.

You might even get away with an nslu2, but I'm not sure
it has the RAM for 16 calls.

A better alternative is to get them to upgrade the DSL to 512 uplink.

Tim.

On 14 Jun 2006, at 17:11, Daniel Salama wrote:

Wow! 22Kbps of overhead? Are you sure? That sounds like way too  
much overhead. I can't use IAX2 because the GXP-2000 are SIP  
phones :( Any other suggestion?


Thanks,
Daniel

On Jun 14, 2006, at 4:37 AM, Gareth Blades wrote:

G729 uses 8kbps but with the IP overhead it actually uses 30kbps  
so for
256k upstream you should be able to handle 8 calls but this is in  
ideal

conditions.

If you were to use IAX and enable trunking then you would use  
30kbps for

the 1st call and 10kbps for each additional call.
See http://www.voip-info.org/wiki/index.php?page=Asterisk 
+bandwidth+iax2


On Wed, 2006-06-14 at 04:17, Daniel Salama wrote:
I have a client with about 16 GXP-2000. They complain that the  
audio
quality is terrible after 2 or 3 simultaneous conversations.  
They are
behind DSL 1.5Mbps down and 256Kbps up. Because they are using  
G711.u
codec, I know they upstream bandwidth is the limiting factor and  
they

most likely won't be able to have more than 3 simultaneous
conversations, and if they're surfing the net and/or checking  
email,

things will only get worse.

So, I purchased some g729 codec licenses and forced their sip peer
configuration to g729 codec. We made sample test calls and were  
able

to make 8 simultaneous calls. On the eighth call, the audio started
to sound choppy. Then we dropped the eighth call and tested with 7.
We could hear just fine on the GXP-2000 but the remote end heard  
us a
bit choppy and/or with a robot-like voice. So, we kept dropping  
calls

until they were of acceptable quality.

My question is, if they were using g729 which, in theory uses 8kbps
plus overhead, they should have been just fine handling eight  
calls.
All the computers were turned off on the network, so there  
shouldn't

have been any other traffic but VoIP. Does anyone have any ideas?

How can I improve their audio quality? I requested BellSouth to
upgrade their capacity but because of where they are located, the
best they can get is 900Kbps/256Kbps, so the upstream continues  
to be

the limiting factor.

I purchased a Dlink-1226G switch to allow me to control QoS on the
LAN. I also upgraded their Netopia DSL router to the latest  
firmware

which allows me to configure VLANs and DiffServ. All the computers
are connected to the PC port on the phone because there is no
available second wiring. Can anyone suggest how to configure the  
QoS

settings on the phones, the Dlink and the Netopia?

While there was "no traffic" on the wire, pinging from/to the
Asterisk box gave me about 47ms latency. When we went passed the  
4th
call, the latency started increasing significantly and when we  
got to

8 calls, the latency was up in the 2000ms. Obviously, if anything I
did in the QoS configuration gave VoIP a priority, then ICMP  
packets
would have the lowest priority and I could understand that to be  
the

reason for such result. However, I'm not sure I configured QoS
properly and that's why I'm asking for help.

Thanks,
Daniel
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Tim Panton
[EMAIL PROTECTED]



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Re: [Asterisk-Users] GXP-2000 Audio Quality

[Tim Panton]

Well, with 16 phones, it might be worth putting a
'satellite' asterisk in their office, have it handle local
transfers, and act as a protocol converter, talking sip to the
phones and (trunked) IAX2 to the outside world.

An embedded low power system would do fine.

You might even get away with an nslu2, but I'm not sure
it has the RAM for 16 calls.

A better alternative is to get them to upgrade the DSL to 512 uplink.

Tim.

On 14 Jun 2006, at 17:11, Daniel Salama wrote:

Wow! 22Kbps of overhead? Are you sure? That sounds like way too  
much overhead. I can't use IAX2 because the GXP-2000 are SIP  
phones :( Any other suggestion?


Thanks,
Daniel

On Jun 14, 2006, at 4:37 AM, Gareth Blades wrote:

G729 uses 8kbps but with the IP overhead it actually uses 30kbps  
so for
256k upstream you should be able to handle 8 calls but this is in  
ideal

conditions.

If you were to use IAX and enable trunking then you would use  
30kbps for

the 1st call and 10kbps for each additional call.
See http://www.voip-info.org/wiki/index.php?page=Asterisk+bandwidth 
+iax2


On Wed, 2006-06-14 at 04:17, Daniel Salama wrote:

I have a client with about 16 GXP-2000. They complain that the audio
quality is terrible after 2 or 3 simultaneous conversations. They  
are
behind DSL 1.5Mbps down and 256Kbps up. Because they are using  
G711.u
codec, I know they upstream bandwidth is the limiting factor and  
they

most likely won't be able to have more than 3 simultaneous
conversations, and if they're surfing the net and/or checking email,
things will only get worse.

So, I purchased some g729 codec licenses and forced their sip peer
configuration to g729 codec. We made sample test calls and were able
to make 8 simultaneous calls. On the eighth call, the audio started
to sound choppy. Then we dropped the eighth call and tested with 7.
We could hear just fine on the GXP-2000 but the remote end heard  
us a
bit choppy and/or with a robot-like voice. So, we kept dropping  
calls

until they were of acceptable quality.

My question is, if they were using g729 which, in theory uses 8kbps
plus overhead, they should have been just fine handling eight calls.
All the computers were turned off on the network, so there shouldn't
have been any other traffic but VoIP. Does anyone have any ideas?

How can I improve their audio quality? I requested BellSouth to
upgrade their capacity but because of where they are located, the
best they can get is 900Kbps/256Kbps, so the upstream continues  
to be

the limiting factor.

I purchased a Dlink-1226G switch to allow me to control QoS on the
LAN. I also upgraded their Netopia DSL router to the latest firmware
which allows me to configure VLANs and DiffServ. All the computers
are connected to the PC port on the phone because there is no
available second wiring. Can anyone suggest how to configure the QoS
settings on the phones, the Dlink and the Netopia?

While there was "no traffic" on the wire, pinging from/to the
Asterisk box gave me about 47ms latency. When we went passed the 4th
call, the latency started increasing significantly and when we  
got to

8 calls, the latency was up in the 2000ms. Obviously, if anything I
did in the QoS configuration gave VoIP a priority, then ICMP packets
would have the lowest priority and I could understand that to be the
reason for such result. However, I'm not sure I configured QoS
properly and that's why I'm asking for help.

Thanks,
Daniel
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Tim Panton
[EMAIL PROTECTED]



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Re: [Asterisk-Users] GXP-2000 Audio Quality

[Steve Underwood]

Welcome to the wonderful world of VoIP, where people are eager to move 
from 8kbps G.729 to 6.3kbps G.723.1, and accept a substantial drop in 
voice quality, and then throw over 20kbps of RTP, IP and related 
overhead on top of them. Isn't IP wonderful? :-)


Regards,
Steve

Daniel Salama wrote:

Wow! 22Kbps of overhead? Are you sure? That sounds like way too much  
overhead. I can't use IAX2 because the GXP-2000 are SIP phones :( Any  
other suggestion?


Thanks,
Daniel

On Jun 14, 2006, at 4:37 AM, Gareth Blades wrote:


G729 uses 8kbps but with the IP overhead it actually uses 30kbps so  for
256k upstream you should be able to handle 8 calls but this is in  ideal
conditions.

If you were to use IAX and enable trunking then you would use  30kbps 
for

the 1st call and 10kbps for each additional call.
See http://www.voip-info.org/wiki/index.php?page=Asterisk+bandwidth 
+iax2


On Wed, 2006-06-14 at 04:17, Daniel Salama wrote:


I have a client with about 16 GXP-2000. They complain that the audio
quality is terrible after 2 or 3 simultaneous conversations. They are
behind DSL 1.5Mbps down and 256Kbps up. Because they are using G711.u
codec, I know they upstream bandwidth is the limiting factor and they
most likely won't be able to have more than 3 simultaneous
conversations, and if they're surfing the net and/or checking email,
things will only get worse.

So, I purchased some g729 codec licenses and forced their sip peer
configuration to g729 codec. We made sample test calls and were able
to make 8 simultaneous calls. On the eighth call, the audio started
to sound choppy. Then we dropped the eighth call and tested with 7.
We could hear just fine on the GXP-2000 but the remote end heard us a
bit choppy and/or with a robot-like voice. So, we kept dropping calls
until they were of acceptable quality.

My question is, if they were using g729 which, in theory uses 8kbps
plus overhead, they should have been just fine handling eight calls.
All the computers were turned off on the network, so there shouldn't
have been any other traffic but VoIP. Does anyone have any ideas?

How can I improve their audio quality? I requested BellSouth to
upgrade their capacity but because of where they are located, the
best they can get is 900Kbps/256Kbps, so the upstream continues to be
the limiting factor.

I purchased a Dlink-1226G switch to allow me to control QoS on the
LAN. I also upgraded their Netopia DSL router to the latest firmware
which allows me to configure VLANs and DiffServ. All the computers
are connected to the PC port on the phone because there is no
available second wiring. Can anyone suggest how to configure the QoS
settings on the phones, the Dlink and the Netopia?

While there was "no traffic" on the wire, pinging from/to the
Asterisk box gave me about 47ms latency. When we went passed the 4th
call, the latency started increasing significantly and when we got to
8 calls, the latency was up in the 2000ms. Obviously, if anything I
did in the QoS configuration gave VoIP a priority, then ICMP packets
would have the lowest priority and I could understand that to be the
reason for such result. However, I'm not sure I configured QoS
properly and that's why I'm asking for help.

Thanks,
Daniel




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Re: [Asterisk-Users] GXP-2000 Audio Quality

[Daniel Salama]

Wow! 22Kbps of overhead? Are you sure? That sounds like way too much  
overhead. I can't use IAX2 because the GXP-2000 are SIP phones :( Any  
other suggestion?


Thanks,
Daniel

On Jun 14, 2006, at 4:37 AM, Gareth Blades wrote:

G729 uses 8kbps but with the IP overhead it actually uses 30kbps so  
for
256k upstream you should be able to handle 8 calls but this is in  
ideal

conditions.

If you were to use IAX and enable trunking then you would use  
30kbps for

the 1st call and 10kbps for each additional call.
See http://www.voip-info.org/wiki/index.php?page=Asterisk+bandwidth 
+iax2


On Wed, 2006-06-14 at 04:17, Daniel Salama wrote:

I have a client with about 16 GXP-2000. They complain that the audio
quality is terrible after 2 or 3 simultaneous conversations. They are
behind DSL 1.5Mbps down and 256Kbps up. Because they are using G711.u
codec, I know they upstream bandwidth is the limiting factor and they
most likely won't be able to have more than 3 simultaneous
conversations, and if they're surfing the net and/or checking email,
things will only get worse.

So, I purchased some g729 codec licenses and forced their sip peer
configuration to g729 codec. We made sample test calls and were able
to make 8 simultaneous calls. On the eighth call, the audio started
to sound choppy. Then we dropped the eighth call and tested with 7.
We could hear just fine on the GXP-2000 but the remote end heard us a
bit choppy and/or with a robot-like voice. So, we kept dropping calls
until they were of acceptable quality.

My question is, if they were using g729 which, in theory uses 8kbps
plus overhead, they should have been just fine handling eight calls.
All the computers were turned off on the network, so there shouldn't
have been any other traffic but VoIP. Does anyone have any ideas?

How can I improve their audio quality? I requested BellSouth to
upgrade their capacity but because of where they are located, the
best they can get is 900Kbps/256Kbps, so the upstream continues to be
the limiting factor.

I purchased a Dlink-1226G switch to allow me to control QoS on the
LAN. I also upgraded their Netopia DSL router to the latest firmware
which allows me to configure VLANs and DiffServ. All the computers
are connected to the PC port on the phone because there is no
available second wiring. Can anyone suggest how to configure the QoS
settings on the phones, the Dlink and the Netopia?

While there was "no traffic" on the wire, pinging from/to the
Asterisk box gave me about 47ms latency. When we went passed the 4th
call, the latency started increasing significantly and when we got to
8 calls, the latency was up in the 2000ms. Obviously, if anything I
did in the QoS configuration gave VoIP a priority, then ICMP packets
would have the lowest priority and I could understand that to be the
reason for such result. However, I'm not sure I configured QoS
properly and that's why I'm asking for help.

Thanks,
Daniel
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Re: [Asterisk-Users] GXP-2000 and Configdownload via TFTP

[Matthias Fechner]

Hi Gareth,

Gareth Blades wrote:
> You need to run the java based tool from the grandstream website to
> convert the template to a format the phone understands.

thx that was the problem. Now it works fine.


Best regards,
Matthias

-- 

"Programming today is a race between software engineers striving to
build bigger and better idiot-proof programs, and the universe trying to
produce bigger and better idiots. So far, the universe is winning." --
Rich Cook

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RE: [Asterisk-Users] GXP-2000 and Configdownload via TFTP

[Mimmus]

You need to encode txt configuration file using tool provided on GS site.  

DV


> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Matthias Fechner
> Sent: Wednesday, June 14, 2006 3:06 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] GXP-2000 and Configdownload via TFTP
> 
> Hi,
> 
> i got my Grandstream GXP-2000 phone today and want to 
> configure it with TFTP. I downloaded the firmware 1.1.0.13 
> and put it into my tftp-server directory.
> Then I downloaded the template from:
> http://www.grandstream.com/DOWNLOAD/Configuration_Tool/Linux_U
> nix/Grandstream_Configuration_File_Template_1.0.6.x.txt
> 
> renamed it to cfg
> 
> Did the configuration in the new file and rebooted my phone.
> I can see in the log file from my tftp server that all files 
> are loaded, the phone did a firmware upgrade.
> 
> But it doesn't seems that the configuration file is loaded.
> 
> Is it necessary to define on any place something that the 
> phone use the config-file via tftp?
> 
> Best regards,
> Matthias

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Re: [Asterisk-Users] GXP-2000 and Configdownload via TFTP

[Patrick]

On Wed, 2006-06-14 at 15:46 +0200, Matthias Fechner wrote:
> Hi,
> 
> I was now successful in getting syslog messages.
> Syslog says the following:
> Jun 14 15:43:57 192.168.0.117 GS_LOG: [][708][FF71][0101000D] ERROR 4099 
> GET cfg
> 
> What does errorcode 4099 mean?

I don't know but it looks like it can't download the cfg file from your
tft server. I've seen this with Cisco phones and boxes booting via PXE.
Make sure the cfg file has the right read permissions ie with
chmod 644 cfg.

You can probably run the tftpserver with one or more -v arguments so you
may get more info. Worth a try.

Regards,
Patrick


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Re: [Asterisk-Users] GXP-2000 and Configdownload via TFTP

[Gareth Blades]

You need to run the java based tool from the grandstream website to
convert the template to a format the phone understands.

On Wed, 2006-06-14 at 14:05, Matthias Fechner wrote:
> Hi,
> 
> i got my Grandstream GXP-2000 phone today and want to configure it
> with TFTP. I downloaded the firmware 1.1.0.13 and put it into my
> tftp-server directory.
> Then I downloaded the template from:
> http://www.grandstream.com/DOWNLOAD/Configuration_Tool/Linux_Unix/Grandstream_Configuration_File_Template_1.0.6.x.txt
> 
> renamed it to cfg
> 
> Did the configuration in the new file and rebooted my phone.
> I can see in the log file from my tftp server that all files are
> loaded, the phone did a firmware upgrade.
> 
> But it doesn't seems that the configuration file is loaded.
> 
> Is it necessary to define on any place something that the phone use
> the config-file via tftp?
> 
> Best regards,
> Matthias

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Re: [Asterisk-Users] GXP-2000 and Configdownload via TFTP

[Matthias Fechner]

Hi,

I was now successful in getting syslog messages.
Syslog says the following:
Jun 14 15:43:57 192.168.0.117 GS_LOG: [][708][FF71][0101000D] ERROR 4099 
GET cfg

What does errorcode 4099 mean?

Best regards,
Matthias

-- 

"Programming today is a race between software engineers striving to
build bigger and better idiot-proof programs, and the universe trying to
produce bigger and better idiots. So far, the universe is winning." --
Rich Cook
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RE: [Asterisk-Users] GXP-2000 1.1.0.13 Issues

[drew-asterisk-users]

Thanks for the offer, but I have just tried 1.1.0.11, it is available 
publicly and it has the same problems on these 2 phones.

On Wed, 14 Jun 2006, Mimmus wrote:

> If can help, I have 80 "00:0b:82:08 :xx:xx" GXP-2000 phones and they works
> well with 1.1.0.11 firmware.
> 
> I can send you this firmware, if you mail me off-list.
> 
> Bye
> DV
> 
> 
> > -Original Message-
> > From: [EMAIL PROTECTED] 
> > [mailto:[EMAIL PROTECTED] On Behalf Of 
> > [EMAIL PROTECTED]
> > Sent: Wednesday, June 14, 2006 1:49 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] GXP-2000 1.1.0.13 Issues
> > 
> > Thats what I thought the problem might be, so I have just now 
> > upgraded the other phone to 1.1.0.13 and its exactly the 
> > same, no speaker phone and hangs from a soft reboot.
> > I also tried the audio loopback in the factory functions 
> > menu, this loopback's fine with the older 1.1.0.13 phones but 
> > does not with the newer ones (by older I mean MAC's 
> > 00:0B:82:06:xx:xx and newer I mean MAC's 00:0B:82:09:xx:xx).
> > 
> > -Drew-
> > 
> >  On Wed, 14 Jun 2006, Gareth Blades wrote:
> > 
> > > The only issue with 1.1.0.13 which affects only certain versions of 
> > > the gxp-2000 is the display blanking issue on very early phones.
> > > It sounds like you have a faulty phone and should return it for a 
> > > replacement.
> > > 
> > > On Wed, 2006-06-14 at 11:57, 
> > [EMAIL PROTECTED] wrote:
> > > > I have had 2 GXP-2000 for a while now and been slowly 
> > following the 
> > > > firmware releases made by Grandstream and am now up to 1.1.0.13.  
> > > > This version works really well on these 2 original phones (MAC's 
> > > > 00:0B:82:06:xx:xx), so I went ahead and ordered another 2 phones 
> > > > (MAC's 00:0B:82:09:xx:xx).  One of these I upgraded to 
> > 1.1.0.13 (it 
> > > > came with
> > > > 1.1.0.5) and pressed it into use.
> > > > The Speaker phone does not work at all (no sound from the 
> > Speaker) 
> > > > and the phone completely hangs doing a soft-reboot, other 
> > than that 
> > > > the phone seems to work well.
> > > > Unfortunatly I do not have a copy of 1.1.0.5 so cannot 
> > downgrade the 
> > > > phone.
> > > > Has anyone else noticed these problems, or does anyone 
> > have a copy 
> > > > of 1.1.0.5.
> > > > 
> > > > -Drew-
> > > > 
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RE: [Asterisk-Users] GXP-2000 1.1.0.13 Issues

[Mimmus]

If can help, I have 80 "00:0b:82:08 :xx:xx" GXP-2000 phones and they works
well with 1.1.0.11 firmware.

I can send you this firmware, if you mail me off-list.

Bye
DV


> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> [EMAIL PROTECTED]
> Sent: Wednesday, June 14, 2006 1:49 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] GXP-2000 1.1.0.13 Issues
> 
> Thats what I thought the problem might be, so I have just now 
> upgraded the other phone to 1.1.0.13 and its exactly the 
> same, no speaker phone and hangs from a soft reboot.
> I also tried the audio loopback in the factory functions 
> menu, this loopback's fine with the older 1.1.0.13 phones but 
> does not with the newer ones (by older I mean MAC's 
> 00:0B:82:06:xx:xx and newer I mean MAC's 00:0B:82:09:xx:xx).
> 
> -Drew-
> 
>  On Wed, 14 Jun 2006, Gareth Blades wrote:
> 
> > The only issue with 1.1.0.13 which affects only certain versions of 
> > the gxp-2000 is the display blanking issue on very early phones.
> > It sounds like you have a faulty phone and should return it for a 
> > replacement.
> > 
> > On Wed, 2006-06-14 at 11:57, 
> [EMAIL PROTECTED] wrote:
> > > I have had 2 GXP-2000 for a while now and been slowly 
> following the 
> > > firmware releases made by Grandstream and am now up to 1.1.0.13.  
> > > This version works really well on these 2 original phones (MAC's 
> > > 00:0B:82:06:xx:xx), so I went ahead and ordered another 2 phones 
> > > (MAC's 00:0B:82:09:xx:xx).  One of these I upgraded to 
> 1.1.0.13 (it 
> > > came with
> > > 1.1.0.5) and pressed it into use.
> > > The Speaker phone does not work at all (no sound from the 
> Speaker) 
> > > and the phone completely hangs doing a soft-reboot, other 
> than that 
> > > the phone seems to work well.
> > > Unfortunatly I do not have a copy of 1.1.0.5 so cannot 
> downgrade the 
> > > phone.
> > > Has anyone else noticed these problems, or does anyone 
> have a copy 
> > > of 1.1.0.5.
> > > 
> > > -Drew-
> > > 
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Re: [Asterisk-Users] GXP-2000 1.1.0.13 Issues

[drew-asterisk-users]

Thats what I thought the problem might be, so I have just now upgraded the 
other phone to 1.1.0.13 and its exactly the same, no speaker phone and 
hangs from a soft reboot.
I also tried the audio loopback in the factory functions menu, this 
loopback's fine with the older 1.1.0.13 phones but does not with the newer 
ones (by older I mean MAC's 00:0B:82:06:xx:xx and newer I mean MAC's 
00:0B:82:09:xx:xx).

-Drew-

 On Wed, 14 Jun 2006, Gareth Blades wrote:

> The only issue with 1.1.0.13 which affects only certain versions of the
> gxp-2000 is the display blanking issue on very early phones.
> It sounds like you have a faulty phone and should return it for a
> replacement.
> 
> On Wed, 2006-06-14 at 11:57, [EMAIL PROTECTED] wrote:
> > I have had 2 GXP-2000 for a while now and been slowly following the 
> > firmware releases made by Grandstream and am now up to 1.1.0.13.  This 
> > version works really well on these 2 original phones (MAC's 
> > 00:0B:82:06:xx:xx), so I went ahead and ordered another 2 phones (MAC's 
> > 00:0B:82:09:xx:xx).  One of these I upgraded to 1.1.0.13 (it came with 
> > 1.1.0.5) and pressed it into use.
> > The Speaker phone does not work at all (no sound from the Speaker) and the 
> > phone completely hangs doing a soft-reboot, other than that the phone 
> > seems to work well.
> > Unfortunatly I do not have a copy of 1.1.0.5 so cannot downgrade the 
> > phone.
> > Has anyone else noticed these problems, or does anyone have a copy of 
> > 1.1.0.5.
> > 
> > -Drew-
> > 
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Re: [Asterisk-Users] GXP-2000 1.1.0.13 Issues

[Gareth Blades]

The only issue with 1.1.0.13 which affects only certain versions of the
gxp-2000 is the display blanking issue on very early phones.
It sounds like you have a faulty phone and should return it for a
replacement.

On Wed, 2006-06-14 at 11:57, [EMAIL PROTECTED] wrote:
> I have had 2 GXP-2000 for a while now and been slowly following the 
> firmware releases made by Grandstream and am now up to 1.1.0.13.  This 
> version works really well on these 2 original phones (MAC's 
> 00:0B:82:06:xx:xx), so I went ahead and ordered another 2 phones (MAC's 
> 00:0B:82:09:xx:xx).  One of these I upgraded to 1.1.0.13 (it came with 
> 1.1.0.5) and pressed it into use.
> The Speaker phone does not work at all (no sound from the Speaker) and the 
> phone completely hangs doing a soft-reboot, other than that the phone 
> seems to work well.
> Unfortunatly I do not have a copy of 1.1.0.5 so cannot downgrade the 
> phone.
> Has anyone else noticed these problems, or does anyone have a copy of 
> 1.1.0.5.
> 
> -Drew-
> 
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Re: [Asterisk-Users] GXP-2000 Audio Quality

[Gareth Blades]

G729 uses 8kbps but with the IP overhead it actually uses 30kbps so for
256k upstream you should be able to handle 8 calls but this is in ideal
conditions.

If you were to use IAX and enable trunking then you would use 30kbps for
the 1st call and 10kbps for each additional call.
See http://www.voip-info.org/wiki/index.php?page=Asterisk+bandwidth+iax2

On Wed, 2006-06-14 at 04:17, Daniel Salama wrote:
> I have a client with about 16 GXP-2000. They complain that the audio  
> quality is terrible after 2 or 3 simultaneous conversations. They are  
> behind DSL 1.5Mbps down and 256Kbps up. Because they are using G711.u  
> codec, I know they upstream bandwidth is the limiting factor and they  
> most likely won't be able to have more than 3 simultaneous  
> conversations, and if they're surfing the net and/or checking email,  
> things will only get worse.
> 
> So, I purchased some g729 codec licenses and forced their sip peer  
> configuration to g729 codec. We made sample test calls and were able  
> to make 8 simultaneous calls. On the eighth call, the audio started  
> to sound choppy. Then we dropped the eighth call and tested with 7.  
> We could hear just fine on the GXP-2000 but the remote end heard us a  
> bit choppy and/or with a robot-like voice. So, we kept dropping calls  
> until they were of acceptable quality.
> 
> My question is, if they were using g729 which, in theory uses 8kbps  
> plus overhead, they should have been just fine handling eight calls.  
> All the computers were turned off on the network, so there shouldn't  
> have been any other traffic but VoIP. Does anyone have any ideas?
> 
> How can I improve their audio quality? I requested BellSouth to  
> upgrade their capacity but because of where they are located, the  
> best they can get is 900Kbps/256Kbps, so the upstream continues to be  
> the limiting factor.
> 
> I purchased a Dlink-1226G switch to allow me to control QoS on the  
> LAN. I also upgraded their Netopia DSL router to the latest firmware  
> which allows me to configure VLANs and DiffServ. All the computers  
> are connected to the PC port on the phone because there is no  
> available second wiring. Can anyone suggest how to configure the QoS  
> settings on the phones, the Dlink and the Netopia?
> 
> While there was "no traffic" on the wire, pinging from/to the  
> Asterisk box gave me about 47ms latency. When we went passed the 4th  
> call, the latency started increasing significantly and when we got to  
> 8 calls, the latency was up in the 2000ms. Obviously, if anything I  
> did in the QoS configuration gave VoIP a priority, then ICMP packets  
> would have the lowest priority and I could understand that to be the  
> reason for such result. However, I'm not sure I configured QoS  
> properly and that's why I'm asking for help.
> 
> Thanks,
> Daniel
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Re: [Asterisk-Users] GXP-2000

[Daniel Salama]

Would you mind telling me how to setup the GXP-2000's VLAN/QoS  
settings with the DES-1226G? I just purchased the DES-1226G and want  
to make sure I setup it up right. I don't have the ability to run  
separate wiring for the PC and the phone and that's why I need this  
help.


Thanks,
Daniel

On Jun 7, 2006, at 9:52 PM, Mike Fedyk wrote:

I have heard good things about the D-Link DES-1226G switch ($150 at  
newegg).  If you can run a separate cable to the computer and  
phone.  If you can't run the extra cables, then configure your  
phone to tag itself as part of the voip vlan and let the switch tag  
everything else as the computer vlan.


I happen to have asterisk running as a router, so I use it doing  
QoS with tc (traffic control) and wondershaper set to prioritize  
based on port ranges.  I sent a patch to the debian bug tracking  
system a while back with a few improvements -- I should check on  
that.  It basically prioritizes smaller packets before larger  
packets with ~8 levels of priority and groups of sizes for the  
packets.  Just doing that automatically handles 80% of the need for  
prioritization without specifying port ranges for the sip rtp packets.


Mike

Daniel Salama wrote:
They are extremely casual web surfers. Just have their Outlook  
client opened checking email every minute. Email traffic is very low.


They are all connected to the same switch. It's a Netopia DSL  
router/modem/switch for the BellSouth DSL service. The computers  
are connected to the PC port behind the GXP-2000.


Any suggestions?

Thanks,
Daniel

On Jun 7, 2006, at 8:49 PM, list mail wrote:

What do they do on the internet? Heavy surfing, large transfers,  
myspace. How are these units connected to the network? Are they  
passing through the same switch?

I don't think it is the phones...

On Jun 7, 2006, at 12:32 PM, Daniel Salama wrote:


Mike,

I added a qualify=500 on those phones. My client has peers  
100218 thru 100222 (a total of 5 phones). Below is the messages  
log since I activated it this morning at 8:30AM:


Jun  7 10:59:21 NOTICE[3648] chan_sip.c: Peer '100219' is now  
TOO LAGGED! (1075ms / 500ms)
Jun  7 10:59:31 NOTICE[3648] chan_sip.c: Peer '100219' is now  
REACHABLE! (66ms / 500ms)
Jun  7 11:02:32 NOTICE[3648] chan_sip.c: Peer '100219' is now  
TOO LAGGED! (1075ms / 500ms)
Jun  7 11:02:42 NOTICE[3648] chan_sip.c: Peer '100219' is now  
REACHABLE! (68ms / 500ms)
Jun  7 11:35:15 NOTICE[3648] chan_sip.c: Peer '100222' is now  
TOO LAGGED! (1114ms / 500ms)
Jun  7 11:35:25 NOTICE[3648] chan_sip.c: Peer '100222' is now  
REACHABLE! (90ms / 500ms)
Jun  7 11:48:20 NOTICE[3648] chan_sip.c: Peer '100219' is now  
TOO LAGGED! (1077ms / 500ms)
Jun  7 11:48:30 NOTICE[3648] chan_sip.c: Peer '100219' is now  
REACHABLE! (72ms / 500ms)
Jun  7 12:24:51 NOTICE[3648] chan_sip.c: Peer '100221' is now  
TOO LAGGED! (1077ms / 500ms)
Jun  7 12:25:01 NOTICE[3648] chan_sip.c: Peer '100221' is now  
REACHABLE! (73ms / 500ms)


As you can see, it only happens to a couple of their phones and  
at random times. They're behind a DSL circuit. I don't know if  
it's because their DSL line is going up/down. They don't  
necessarily claim their Internet goes down, however, they are  
not constantly check it.


What would you (or anyone else) suggest?

Thanks,
Daniel

On Jun 7, 2006, at 8:07 AM, Mike Fedyk wrote:

Do you have multiple phones going down at the same time?  If  
so, monitor them with "qualify=500" in sip.conf to see if they  
hit that limit.  If you see more than one go down within a  
short period of time, you have network problems.  Check the  
quality of the network switches they have.


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- 
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Re: [Asterisk-Users] GXP-2000

[Matthias Fechner]

Hi,

is it possible to update the phonebook of the gxp-2000 via tftp?
So I can maintain the phonebook central or using ldap etc.?

Best regards,
Matthias
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Re: [Asterisk-Users] GXP-2000 MultiPurpose Keys

[Daniel Salama]

That's great. GS support people are great, but I had asked him how to  
set other parameters that I see on the web and they told me they  
didn't know. That I should look through the wiki or other web sources.


Anyway, that's great to know.

Thanks,
Daniel

On Jun 10, 2006, at 5:16 AM, Phil Blundell wrote:


For future reference, I think the Grandstream config files can program
any parameter that's included in the web interface.  If you want to  
set

something that isn't in the template, you can use "view source" on the
web form to figure out the name of the option: the field names in the
HTML are the same as the ones that go in the config file.

p.

On Sat, 2006-06-10 at 02:06 -0400, Daniel Salama wrote:

Wow! Awesome. This template is much more complete than the one on
GS's download page.

Thanks,
Daniel

On Jun 9, 2006, at 10:26 AM, Gareth Blades wrote:

Yes you can as long as you have at least the 1.0.2.13 firmware. I  
have
attached the template. The multi-purpose key settings are at the  
end.


On Fri, 2006-06-09 at 14:41, Daniel Salama wrote:

Is it possible to program the multi-purpose keys on a GXP-2000
remotely via a TFTP configuration file? If so, what are the
parameters to put in the configuration file?

Thanks,
Daniel

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Re: [Asterisk-Users] GXP-2000 MultiPurpose Keys

[Phil Blundell]

For future reference, I think the Grandstream config files can program
any parameter that's included in the web interface.  If you want to set
something that isn't in the template, you can use "view source" on the
web form to figure out the name of the option: the field names in the
HTML are the same as the ones that go in the config file.

p.

On Sat, 2006-06-10 at 02:06 -0400, Daniel Salama wrote:
> Wow! Awesome. This template is much more complete than the one on  
> GS's download page.
> 
> Thanks,
> Daniel
> 
> On Jun 9, 2006, at 10:26 AM, Gareth Blades wrote:
> 
> > Yes you can as long as you have at least the 1.0.2.13 firmware. I have
> > attached the template. The multi-purpose key settings are at the end.
> >
> > On Fri, 2006-06-09 at 14:41, Daniel Salama wrote:
> >> Is it possible to program the multi-purpose keys on a GXP-2000
> >> remotely via a TFTP configuration file? If so, what are the
> >> parameters to put in the configuration file?
> >>
> >> Thanks,
> >> Daniel
> >>
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Re: [Asterisk-Users] GXP-2000 MultiPurpose Keys

[Daniel Salama]

Wow! Awesome. This template is much more complete than the one on  
GS's download page.


Thanks,
Daniel

On Jun 9, 2006, at 10:26 AM, Gareth Blades wrote:


Yes you can as long as you have at least the 1.0.2.13 firmware. I have
attached the template. The multi-purpose key settings are at the end.

On Fri, 2006-06-09 at 14:41, Daniel Salama wrote:

Is it possible to program the multi-purpose keys on a GXP-2000
remotely via a TFTP configuration file? If so, what are the
parameters to put in the configuration file?

Thanks,
Daniel

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Re: [Asterisk-Users] GXP-2000 MultiPurpose Keys

[Gareth Blades]

Yes you can if you are running 1.0.2.13 or later. I have the template
which I tried posting here as an attachment but it has not arrived yet.
If it does not arrive you can email me directly or contact grandstream
support.

On Fri, 2006-06-09 at 14:41, Daniel Salama wrote:
> Is it possible to program the multi-purpose keys on a GXP-2000  
> remotely via a TFTP configuration file? If so, what are the  
> parameters to put in the configuration file?
> 
> Thanks,
> Daniel
> 
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Re: [Asterisk-Users] GXP-2000 MultiPurpose Keys

[Gareth Blades]

Yes you can as long as you have at least the 1.0.2.13 firmware. I have
attached the template. The multi-purpose key settings are at the end.

On Fri, 2006-06-09 at 14:41, Daniel Salama wrote:
> Is it possible to program the multi-purpose keys on a GXP-2000  
> remotely via a TFTP configuration file? If so, what are the  
> parameters to put in the configuration file?
> 
> Thanks,
> Daniel
> 
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## Configuration template for GXP-2000 firmware version 1.0.2.13


##
##  Advanced/System-wide Options
##

# Admin password for web interface
P2 = admin

# Silence Suppression. 0 - no, 1 - yes
P50 = 0

# Voice Frames per TX (up to 10/20/32/64 frames for G711/G726/G723/other codecs 
respectively)
P37 = 2

# Layer 3 QoS (IP Diff-Serv or Precedence value for RTP)
P38 = 48

# Layer 2 QoS. 802.1Q/VLAN Tag (VLAN classification for RTP)
P51 = 0

# Layer 2 QoS. 802.1p priority value (0 - 7)
P87 = 0

# No Key Entry Timeout. Default - 4 seconds.
P85 = 4

# Use # as Dial Key (if set to Yes, "#" will function as the "(Re-)Dial" key). 
0 - no, 1 - yes
P72 = 1

# Local RTP port (1024-65535, default 5004)
P39 = 5004 

# Use Random Port. 0 - no, 1 - yes
P78 = 0

# Keep-alive interval (in seconds. default 20 seconds)
P84 = 20

# Use NAT IP.  This will enable our SIP client to use this IP in the SIP 
message. Example 64.3.153.50.
P101 =

# STUN server
P76 = stun.mycompany.com

#-
# Firmware Upgrade 
#-

# Firmware Upgrade. 0 - TFTP Upgrade,  1 - HTTP Upgrade.
P212 = 0

# Firmware Server Path
P192 =

# Config Server Path
P237 =

# Firmware File Prefix
P232 =

# Firmware File Postfix
P233 =

# Config File Prefix
P234 =

# Config File Postfix
P235 =

# Allow DHCP Option 66 to override server. 0 - No, 1 - Yes. Default is No.
# When set to Yes(1), it will override the configured provision path and method.
P145 = 0

# Automatic Upgrade. 0 - No, 1 - Yes (checking every defined days). Default is 
No.
P194 = 0

# Check for new firmware every () minutes, unit is in minute, default is 7 days.
P193 = 10080

# Use firmware pre/postfix to determine if f/w is required
# 0 = Always Check for New Firmware 
# 1 = Check New Firmware only when F/W pre/suffix changes 
P238 = 0

# DTMF Payload Type
P79 = 101

# Syslog Server (name of the server, max length is 64 charactors)
P207 = 

# Syslog Level (Default setting is NONE)
# 0 - NONE, 1 - DEBUG, 2 - INFO, 3 - WARNING, 4 - ERROR
P208 = 0

# NTP Server
P30 = time.nist.gov

# Allow DHCP Option 42 to override NTP server. 0 - No, 1 - Yes. Default is No.
# When set to Yes(1), it will override the configured NTP server.
P144 = 0

# Distinctive Ring Tone
# Use custom ring tone 1 if incoming caller ID is the following:
P105 =

# Use custom ring tone 2 if incoming caller ID is the following:
P106 =

# Use custom ring tone 3 if incoming caller ID is the following:
P107 =

# Disable Call Waiting. 0 - no, 1 - yes
P91 = 0

# Lock Keypad Update. 0 - no, 1 - yes
P88 = 0


# Primary Account (Account 1) Settings


# Account Active (In Use). 0 - no, 1 - yes
P271 = 1

# Account Name
P270 =

# SIP Server
P47 = sip.mycompany.com

# Outbound Proxy
P48 = proxy.mycompany.com

# SIP User ID
P35 = 8000

# Authenticate ID
P36 = 8000

# Authenticate password
P34 = 

# Display Name (John Doe)
P3 = 

# Use DNS SRV. 0 - No, 1 - Yes.
P103 = 0

# SIP User ID is phone number. 0 - no, 1 - yes
P63 = 0

# SIP Registration. 0 - no, 1 - yes
P31 = 1

# Unregister On Reboot. 0 - no, 1 - yes
P81 = 0

# Register Expiration (in minutes. default 1 hour, max 45 days)
P32 = 60

# Local SIP port (default 5060)
P40 = 5060

# SIP T1 Timeout. RFC 3261 T1 value (RTT estimate)
# 50 - 0.5 sec, 100 - 1 sec, 200 - 2 sec. Default 100.
P209 = 100

# SIP T2 Interval. RFC 3261 T2 value. The maximum retransmit interval for 
non-INVITE requests and INVITE responses.
# 200 - 2 sec, 400 - 4 sec, 800 - 8 sec. Default 400.
P250 = 400

# NAT Traversal. 0 - yes, 1 - no, 2 - No, but send keep-alive
P52 = 0

# SUBSCRIBE for MWI. (Whether or not send SUBSCRIBE for Message Waiting 
Indication) 0 - No, 1 - Yes.
P99 = 0

# Proxy-Require (A SIP extension to enable firewall penetration)
P197 =

# Voice Mail UserID (User ID/extension for 3rd party voi

RE: [Asterisk-Users] GXP-2000 MultiPurpose Keys

[Rick Smith]

good question!  I'd like to know too, so keep it public please !:) 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Daniel
Salama
Sent: Friday, June 09, 2006 9:42 AM
To: Non-Commercial Discussion Asterisk
Subject: [Asterisk-Users] GXP-2000 MultiPurpose Keys

Is it possible to program the multi-purpose keys on a GXP-2000 remotely
via a TFTP configuration file? If so, what are the parameters to put in
the configuration file?

Thanks,
Daniel


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Re: [Asterisk-Users] GXP-2000

[Patrick]

On Thu, 2006-06-08 at 13:21 -0400, list mail wrote:
> I'm willing to bet the phones that are stalling have the most active
> computer users attatched to them. I wouldn't advise having the
> computer running through the phones port. To me that is asking too
> much out of the <$100 phone.
> Run each device from it's own port on your switch.

I've seen that too on ACTEL P103 phones. Actually didn't even need a
lot of network traffic to crash the phone. They just went poof at the
blink of an eye. You can test this by hooking up a box with a traffic
generator to one of these phones and let it blast for a while.
The opposite worked also. I couldn't make them crash anymore after I
forced the link to the LAN and the computer to 10Mbit half-duplex.

Regards,
Patrick

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Re: [Asterisk-Users] GXP-2000

[list mail]

I'm willing to bet the phones that are stalling have the most active computer users attatched to them. I wouldn't advise having the computer running through the phones port. To me that is asking too much out of the <$100 phone.Run each device from it's own port on your switch.On Jun 7, 2006, at 9:36 PM, Daniel Salama wrote:They are extremely casual web surfers. Just have their Outlook client opened checking email every minute. Email traffic is very low.They are all connected to the same switch. It's a Netopia DSL router/modem/switch for the BellSouth DSL service. The computers are connected to the PC port behind the GXP-2000.Any suggestions?Thanks,DanielOn Jun 7, 2006, at 8:49 PM, list mail wrote:What do they do on the internet? Heavy surfing, large transfers, myspace. How are these units connected to the network? Are they passing through the same switch?I don't think it is the phones...On Jun 7, 2006, at 12:32 PM, Daniel Salama wrote:Mike,I added a qualify=500 on those phones. My client has peers 100218 thru 100222 (a total of 5 phones). Below is the messages log since I activated it this morning at 8:30AM:Jun  7 10:59:21 NOTICE[3648] chan_sip.c: Peer '100219' is now TOO LAGGED! (1075ms / 500ms)Jun  7 10:59:31 NOTICE[3648] chan_sip.c: Peer '100219' is now REACHABLE! (66ms / 500ms)Jun  7 11:02:32 NOTICE[3648] chan_sip.c: Peer '100219' is now TOO LAGGED! (1075ms / 500ms)Jun  7 11:02:42 NOTICE[3648] chan_sip.c: Peer '100219' is now REACHABLE! (68ms / 500ms)Jun  7 11:35:15 NOTICE[3648] chan_sip.c: Peer '100222' is now TOO LAGGED! (1114ms / 500ms)Jun  7 11:35:25 NOTICE[3648] chan_sip.c: Peer '100222' is now REACHABLE! (90ms / 500ms)Jun  7 11:48:20 NOTICE[3648] chan_sip.c: Peer '100219' is now TOO LAGGED! (1077ms / 500ms)Jun  7 11:48:30 NOTICE[3648] chan_sip.c: Peer '100219' is now REACHABLE! (72ms / 500ms)Jun  7 12:24:51 NOTICE[3648] chan_sip.c: Peer '100221' is now TOO LAGGED! (1077ms / 500ms)Jun  7 12:25:01 NOTICE[3648] chan_sip.c: Peer '100221' is now REACHABLE! (73ms / 500ms)As you can see, it only happens to a couple of their phones and at random times. They're behind a DSL circuit. I don't know if it's because their DSL line is going up/down. They don't necessarily claim their Internet goes down, however, they are not constantly check it.What would you (or anyone else) suggest?Thanks,DanielOn Jun 7, 2006, at 8:07 AM, Mike Fedyk wrote:Do you have multiple phones going down at the same time?  If so, monitor them with "qualify=500" in sip.conf to see if they hit that limit.  If you see more than one go down within a short period of time, you have network problems.  Check the quality of the network switches they have.  ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users ___
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Re: [Asterisk-Users] GXP-2000

[Kristian Kielhofner]

Mike Fedyk wrote:
I have heard good things about the D-Link DES-1226G switch ($150 at 
newegg).  If you can run a separate cable to the computer and phone.  If 
you can't run the extra cables, then configure your phone to tag itself 
as part of the voip vlan and let the switch tag everything else as the 
computer vlan.


I happen to have asterisk running as a router, so I use it doing QoS 
with tc (traffic control) and wondershaper set to prioritize based on 
port ranges.  I sent a patch to the debian bug tracking system a while 
back with a few improvements -- I should check on that.  It basically 
prioritizes smaller packets before larger packets with ~8 levels of 
priority and groups of sizes for the packets.  Just doing that 
automatically handles 80% of the need for prioritization without 
specifying port ranges for the sip rtp packets.


Mike



Mike,

	Have you tried AstShape?  Shapping based on port ranges is totally hit 
or miss.  TOS is the way to go:


http://www.krisk.org/files/astlinux-i586/usr/sbin/astshape

Comment out the . /etc/rc.conf and you should be okay!

--
Kristian Kielhofner
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RE: [Asterisk-Users] GXP-2000

[Nabeel Jafferali]

> Is the 94x any better? seems without backlighting, any are 
> next to useless.

The SPA-9x2 have backlit displays.

Nabeel

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RE: [Asterisk-Users] GXP-2000

[mustardman29]

What about Aastra 480i, 9133i? 

> -Original Message-
> From: Kerry Garrison [mailto:[EMAIL PROTECTED] 
> Sent: Wednesday, June 07, 2006 1:28 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] GXP-2000
> 
> With hundreds of installed phones now, here are my choices in order
> 
> Linksys SPA-941/942
> Polycom 501/601
> Cisco 7960
> Polycom 301
> Snom 320/360
> 
> I would never ever ever sell a client on a SPA-841 or heaven 
> forbid the GXP-2000. All the clients who bought those 
> originally sold them off and went for better phones very quickly.
> 
> Kerry Garrison
> Director of Technical Services
> Tech Data Pros - Orange County's Mobile IT Service Provider
> (949) 502-7819 x200 - [EMAIL PROTECTED] 
> http://www.techdatapros.com
> 
> 
> 
> 
> > Polycom 501
> > Linksys spa-941
> > Polycom 301
> > Sipura/Linksys spa-841
> > Grandstream GXP-2000
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> >http://lists.digium.com/mailman/listinfo/asterisk-users
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> 
> 
> 
> 
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Re: [Asterisk-Users] GXP-2000

[Erick Baum]

We had not used these phones before, which I will admit was my first mistake.  However, I did do research online to see what other peoples experiences were but the major problems with the phone started surfacing online almost immediately after we installed them.  Before that, there were the usual like them/don't like them posts.  But there was nothing about the multitude of problems, serious problems.  The only issue that I knew about when we got them was the speakerphone problem which required a (beta) firmware upgrade to resolve, didn't know it was "beta" until after we got them.  In fact their firmware is still stored in a BETATEST folder to this day... which really gives me a warm fuzzy feeling.

Erick 
On 6/6/06, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote:
Erick Baum wrote:> We setup a company with 50 of these phones and had my client not been as> understanding as they were, that could have put me out of business.
> What an unbelievable nightmare.  This was about 8 months ago when the> firmware was so bad the phone was a better paper weight than anything else.You did not experience these problems when you set up your prototype
problems and did not see people reporting these issues when you searchedthe mailing lists?--Now accepting new clients in Birmingham, Atlanta, Huntsville,Chattanooga, and Montgomery.___
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Re: [Asterisk-Users] GXP-2000

[Mike Fedyk]

I have heard good things about the D-Link DES-1226G switch ($150 at 
newegg).  If you can run a separate cable to the computer and phone.  If 
you can't run the extra cables, then configure your phone to tag itself 
as part of the voip vlan and let the switch tag everything else as the 
computer vlan.


I happen to have asterisk running as a router, so I use it doing QoS 
with tc (traffic control) and wondershaper set to prioritize based on 
port ranges.  I sent a patch to the debian bug tracking system a while 
back with a few improvements -- I should check on that.  It basically 
prioritizes smaller packets before larger packets with ~8 levels of 
priority and groups of sizes for the packets.  Just doing that 
automatically handles 80% of the need for prioritization without 
specifying port ranges for the sip rtp packets.


Mike

Daniel Salama wrote:
They are extremely casual web surfers. Just have their Outlook client 
opened checking email every minute. Email traffic is very low.


They are all connected to the same switch. It's a Netopia DSL 
router/modem/switch for the BellSouth DSL service. The computers are 
connected to the PC port behind the GXP-2000.


Any suggestions?

Thanks,
Daniel

On Jun 7, 2006, at 8:49 PM, list mail wrote:

What do they do on the internet? Heavy surfing, large transfers, 
myspace. 
How are these units connected to the network? Are they passing 
through the same switch?

I don't think it is the phones...

On Jun 7, 2006, at 12:32 PM, Daniel Salama wrote:


Mike,

I added a qualify=500 on those phones. My client has peers 100218 
thru 100222 (a total of 5 phones). Below is the messages log since I 
activated it this morning at 8:30AM:


Jun  7 10:59:21 NOTICE[3648] chan_sip.c: Peer '100219' is now TOO 
LAGGED! (1075ms / 500ms)
Jun  7 10:59:31 NOTICE[3648] chan_sip.c: Peer '100219' is now 
REACHABLE! (66ms / 500ms)
Jun  7 11:02:32 NOTICE[3648] chan_sip.c: Peer '100219' is now TOO 
LAGGED! (1075ms / 500ms)
Jun  7 11:02:42 NOTICE[3648] chan_sip.c: Peer '100219' is now 
REACHABLE! (68ms / 500ms)
Jun  7 11:35:15 NOTICE[3648] chan_sip.c: Peer '100222' is now TOO 
LAGGED! (1114ms / 500ms)
Jun  7 11:35:25 NOTICE[3648] chan_sip.c: Peer '100222' is now 
REACHABLE! (90ms / 500ms)
Jun  7 11:48:20 NOTICE[3648] chan_sip.c: Peer '100219' is now TOO 
LAGGED! (1077ms / 500ms)
Jun  7 11:48:30 NOTICE[3648] chan_sip.c: Peer '100219' is now 
REACHABLE! (72ms / 500ms)
Jun  7 12:24:51 NOTICE[3648] chan_sip.c: Peer '100221' is now TOO 
LAGGED! (1077ms / 500ms)
Jun  7 12:25:01 NOTICE[3648] chan_sip.c: Peer '100221' is now 
REACHABLE! (73ms / 500ms)


As you can see, it only happens to a couple of their phones and at 
random times. They're behind a DSL circuit. I don't know if it's 
because their DSL line is going up/down. They don't necessarily 
claim their Internet goes down, however, they are not constantly 
check it.


What would you (or anyone else) suggest?

Thanks,
Daniel

On Jun 7, 2006, at 8:07 AM, Mike Fedyk wrote:

Do you have multiple phones going down at the same time?  If so, 
monitor them with "qualify=500" in sip.conf to see if they hit that 
limit.  If you see more than one go down within a short period of 
time, you have network problems.  Check the quality of the network 
switches they have. 


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Re: [Asterisk-Users] GXP-2000

[Daniel Salama]

They are extremely casual web surfers. Just have their Outlook client opened checking email every minute. Email traffic is very low.They are all connected to the same switch. It's a Netopia DSL router/modem/switch for the BellSouth DSL service. The computers are connected to the PC port behind the GXP-2000.Any suggestions?Thanks,DanielOn Jun 7, 2006, at 8:49 PM, list mail wrote:What do they do on the internet? Heavy surfing, large transfers, myspace. How are these units connected to the network? Are they passing through the same switch?I don't think it is the phones...On Jun 7, 2006, at 12:32 PM, Daniel Salama wrote:Mike,I added a qualify=500 on those phones. My client has peers 100218 thru 100222 (a total of 5 phones). Below is the messages log since I activated it this morning at 8:30AM:Jun  7 10:59:21 NOTICE[3648] chan_sip.c: Peer '100219' is now TOO LAGGED! (1075ms / 500ms)Jun  7 10:59:31 NOTICE[3648] chan_sip.c: Peer '100219' is now REACHABLE! (66ms / 500ms)Jun  7 11:02:32 NOTICE[3648] chan_sip.c: Peer '100219' is now TOO LAGGED! (1075ms / 500ms)Jun  7 11:02:42 NOTICE[3648] chan_sip.c: Peer '100219' is now REACHABLE! (68ms / 500ms)Jun  7 11:35:15 NOTICE[3648] chan_sip.c: Peer '100222' is now TOO LAGGED! (1114ms / 500ms)Jun  7 11:35:25 NOTICE[3648] chan_sip.c: Peer '100222' is now REACHABLE! (90ms / 500ms)Jun  7 11:48:20 NOTICE[3648] chan_sip.c: Peer '100219' is now TOO LAGGED! (1077ms / 500ms)Jun  7 11:48:30 NOTICE[3648] chan_sip.c: Peer '100219' is now REACHABLE! (72ms / 500ms)Jun  7 12:24:51 NOTICE[3648] chan_sip.c: Peer '100221' is now TOO LAGGED! (1077ms / 500ms)Jun  7 12:25:01 NOTICE[3648] chan_sip.c: Peer '100221' is now REACHABLE! (73ms / 500ms)As you can see, it only happens to a couple of their phones and at random times. They're behind a DSL circuit. I don't know if it's because their DSL line is going up/down. They don't necessarily claim their Internet goes down, however, they are not constantly check it.What would you (or anyone else) suggest?Thanks,DanielOn Jun 7, 2006, at 8:07 AM, Mike Fedyk wrote:Do you have multiple phones going down at the same time?  If so, monitor them with "qualify=500" in sip.conf to see if they hit that limit.  If you see more than one go down within a short period of time, you have network problems.  Check the quality of the network switches they have.  ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users ___
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Re: [Asterisk-Users] GXP-2000

[list mail]

What do they do on the internet? Heavy surfing, large transfers, myspace. How are these units connected to the network? Are they passing through the same switch?I don't think it is the phones...On Jun 7, 2006, at 12:32 PM, Daniel Salama wrote:Mike,I added a qualify=500 on those phones. My client has peers 100218 thru 100222 (a total of 5 phones). Below is the messages log since I activated it this morning at 8:30AM:Jun  7 10:59:21 NOTICE[3648] chan_sip.c: Peer '100219' is now TOO LAGGED! (1075ms / 500ms)Jun  7 10:59:31 NOTICE[3648] chan_sip.c: Peer '100219' is now REACHABLE! (66ms / 500ms)Jun  7 11:02:32 NOTICE[3648] chan_sip.c: Peer '100219' is now TOO LAGGED! (1075ms / 500ms)Jun  7 11:02:42 NOTICE[3648] chan_sip.c: Peer '100219' is now REACHABLE! (68ms / 500ms)Jun  7 11:35:15 NOTICE[3648] chan_sip.c: Peer '100222' is now TOO LAGGED! (1114ms / 500ms)Jun  7 11:35:25 NOTICE[3648] chan_sip.c: Peer '100222' is now REACHABLE! (90ms / 500ms)Jun  7 11:48:20 NOTICE[3648] chan_sip.c: Peer '100219' is now TOO LAGGED! (1077ms / 500ms)Jun  7 11:48:30 NOTICE[3648] chan_sip.c: Peer '100219' is now REACHABLE! (72ms / 500ms)Jun  7 12:24:51 NOTICE[3648] chan_sip.c: Peer '100221' is now TOO LAGGED! (1077ms / 500ms)Jun  7 12:25:01 NOTICE[3648] chan_sip.c: Peer '100221' is now REACHABLE! (73ms / 500ms)As you can see, it only happens to a couple of their phones and at random times. They're behind a DSL circuit. I don't know if it's because their DSL line is going up/down. They don't necessarily claim their Internet goes down, however, they are not constantly check it.What would you (or anyone else) suggest?Thanks,DanielOn Jun 7, 2006, at 8:07 AM, Mike Fedyk wrote:Do you have multiple phones going down at the same time?  If so, monitor them with "qualify=500" in sip.conf to see if they hit that limit.  If you see more than one go down within a short period of time, you have network problems.  Check the quality of the network switches they have.  ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users ___
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Re: [Asterisk-Users] GXP-2000

[Mike Fedyk]

Kerry Garrison wrote:

I would never ever ever sell a client on a SPA-841 or heaven forbid the
GXP-2000. All the clients who bought those originally sold them off and went
for better phones very quickly.
Let me say that when suggesting the spa-841 it is only in the context of 
sub-$100 phones.


I hadn't worked with any spa-841s before, but when my client wanted 
cheaper phones than the 941s that I suggested, I strongly warned them 
that from what I had seen, about 50% of them are returned.  But they 
insisted and I have to say that the phones are not *that* bad.  There a 
lot of things I like about them that I don't like about my polycom 301 
(though most of my gripes with the 301 could be fixed by remapping some 
of the buttons and make call lists available with one button press, so 
it's not a hardware deficiency except for the lack of speakerphone and 
backlight).


Mike
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RE: [Asterisk-Users] GXP-2000

[Kerry Garrison]

With hundreds of installed phones now, here are my choices in order

Linksys SPA-941/942
Polycom 501/601
Cisco 7960
Polycom 301
Snom 320/360

I would never ever ever sell a client on a SPA-841 or heaven forbid the
GXP-2000. All the clients who bought those originally sold them off and went
for better phones very quickly.

Kerry Garrison
Director of Technical Services
Tech Data Pros - Orange County's Mobile IT Service Provider
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com




> Polycom 501
> Linksys spa-941
> Polycom 301
> Sipura/Linksys spa-841
> Grandstream GXP-2000
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Re: [Asterisk-Users] GXP-2000

[Daniel Salama]

No changes whatsoever. Unplugged the spa and replaced it with a gxp.  
I haven't tweaked any RTP or QoS parameters for I don't have any  
documentation on it :(


Thanks,
Daniel

On Jun 7, 2006, at 3:44 PM, Mike Fedyk wrote:

Did you try setting the RTP packet time size to 0.020?  Also I  
would look at the trunk, provider or internet connection before the  
phones I started suspecting the phones.


I have had the same problems with providers, and the conversations  
sound great from one location to another over the internet, but  
once it hits a provider, the sound quality drops.  That is not the  
fault of the phones.  Are you sure you didn't change anything else  
when you switched from the spa-841 phones?


Daniel Salama wrote:
The complete opposite. The user complaints that either they cannot  
hear the remote party well or the remote party cannot hear them  
well. Sometimes it works and sometimes the volume is very low and  
that's why they cannot hear.


- Daniel

On Jun 7, 2006, at 1:35 PM, Mike Fedyk wrote:

What specifically were the voice quality complaints about the  
spa-841 phones?  The only thing I have noticed is calls can be  
louder than expected.  What else have you seen?




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Re: [Asterisk-Users] GXP-2000

[Mike Fedyk]

John Novack wrote:
Is the 94x any better? seems without backlighting, any are next to 
useless.
Yes, I like the 941 better than the Polycom 301 and the display is much 
improved (no backlight, but one of the guys at voipsupply told me that 
the 942 has a backlight which sounds very promising).  The base for the 
941 is more angled like the polycom phones and it is bigger and heavier 
so it doesn't move around as much.  And the buttons have a very nice feel.


With the list of phones I have used, here is how I would choose them 
(first being better):


Polycom 501
Linksys spa-941
Polycom 301
Sipura/Linksys spa-841
Grandstream GXP-2000
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Re: [Asterisk-Users] GXP-2000

[Daniel Salama]

Latest firmware installed and problem with handset. They don't use  
headset nor speakerphone.


Thanks,
Daniel

On Jun 7, 2006, at 3:14 PM, John Novack wrote:




Daniel Salama wrote:



As for the SPA-841, I have a client with a few of them and he  
cannot stop complaining about the bad audio quality.


Latest/last firmware upgrade?
Handset?
speaker phone?
headset?

I find the handset quite acceptable
Speaker phones are a can of worms, with so many issues not related  
to the phones

the SPA-841 might as well not have a display.
Is the 94x any better? seems without backlighting, any are next to  
useless.


John Novack

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Re: [Asterisk-Users] GXP-2000

[Mike Fedyk]

Did you try setting the RTP packet time size to 0.020?  Also I would 
look at the trunk, provider or internet connection before the phones I 
started suspecting the phones.


I have had the same problems with providers, and the conversations sound 
great from one location to another over the internet, but once it hits a 
provider, the sound quality drops.  That is not the fault of the 
phones.  Are you sure you didn't change anything else when you switched 
from the spa-841 phones?


Daniel Salama wrote:
The complete opposite. The user complaints that either they cannot 
hear the remote party well or the remote party cannot hear them well. 
Sometimes it works and sometimes the volume is very low and that's why 
they cannot hear.


- Daniel

On Jun 7, 2006, at 1:35 PM, Mike Fedyk wrote:

What specifically were the voice quality complaints about the spa-841 
phones?  The only thing I have noticed is calls can be louder than 
expected.  What else have you seen?




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Re: [Asterisk-Users] GXP-2000

[John Novack]

Daniel Salama wrote:



As for the SPA-841, I have a client with a few of them and he cannot 
stop complaining about the bad audio quality.


Latest/last firmware upgrade?
Handset?
speaker phone?
headset?

I find the handset quite acceptable
Speaker phones are a can of worms, with so many issues not related to 
the phones

the SPA-841 might as well not have a display.
Is the 94x any better? seems without backlighting, any are next to useless.

John Novack

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Re: [Asterisk-Users] GXP-2000

[Daniel Salama]

The complete opposite. The user complaints that either they cannot hear the remote party well or the remote party cannot hear them well. Sometimes it works and sometimes the volume is very low and that's why they cannot hear.- DanielOn Jun 7, 2006, at 1:35 PM, Mike Fedyk wrote:What specifically were the voice quality complaints about the spa-841 phones?  The only thing I have noticed is calls can be louder than expected.  What else have you seen? ___
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Re: [Asterisk-Users] GXP-2000

[Mike Fedyk]

What specifically were the voice quality complaints about the spa-841 
phones?  The only thing I have noticed is calls can be louder than 
expected.  What else have you seen?


Daniel Salama wrote:
They don't all go down at the same time, or at least, my client hasn't 
noticed. I just added the qualify option. Let's see how that goes.


As for the SPA-841, I have a client with a few of them and he cannot 
stop complaining about the bad audio quality. I replace a couple with 
a PAP-2 and another one with the GXP-2000 and he claims the quality to 
be incredibly better for both the PAP2 and the GXP-2000. He hasn't 
complained about the problems I mentioned on the GXP-2000 - yet :)


Thanks,
Daniel

On Jun 7, 2006, at 8:07 AM, Mike Fedyk wrote:

Do you have multiple phones going down at the same time?  If so, 
monitor them with "qualify=500" in sip.conf to see if they hit that 
limit.  If you see more than one go down within a short period of 
time, you have network problems.  Check the quality of the network 
switches they have. 

Also I have heard some phones have trouble with broadcast packets (at 
least this has been said about the spa-841 on the wiki).  You should 
strongly consider putting them on a separate vlan to avoid any issues 
like that.  In the future, for phones under $100 then look at the 
spa-841 phones.






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Re: [Asterisk-Users] GXP-2000

[Daniel Salama]

Mike,I added a qualify=500 on those phones. My client has peers 100218 thru 100222 (a total of 5 phones). Below is the messages log since I activated it this morning at 8:30AM:Jun  7 10:59:21 NOTICE[3648] chan_sip.c: Peer '100219' is now TOO LAGGED! (1075ms / 500ms)Jun  7 10:59:31 NOTICE[3648] chan_sip.c: Peer '100219' is now REACHABLE! (66ms / 500ms)Jun  7 11:02:32 NOTICE[3648] chan_sip.c: Peer '100219' is now TOO LAGGED! (1075ms / 500ms)Jun  7 11:02:42 NOTICE[3648] chan_sip.c: Peer '100219' is now REACHABLE! (68ms / 500ms)Jun  7 11:35:15 NOTICE[3648] chan_sip.c: Peer '100222' is now TOO LAGGED! (1114ms / 500ms)Jun  7 11:35:25 NOTICE[3648] chan_sip.c: Peer '100222' is now REACHABLE! (90ms / 500ms)Jun  7 11:48:20 NOTICE[3648] chan_sip.c: Peer '100219' is now TOO LAGGED! (1077ms / 500ms)Jun  7 11:48:30 NOTICE[3648] chan_sip.c: Peer '100219' is now REACHABLE! (72ms / 500ms)Jun  7 12:24:51 NOTICE[3648] chan_sip.c: Peer '100221' is now TOO LAGGED! (1077ms / 500ms)Jun  7 12:25:01 NOTICE[3648] chan_sip.c: Peer '100221' is now REACHABLE! (73ms / 500ms)As you can see, it only happens to a couple of their phones and at random times. They're behind a DSL circuit. I don't know if it's because their DSL line is going up/down. They don't necessarily claim their Internet goes down, however, they are not constantly check it.What would you (or anyone else) suggest?Thanks,DanielOn Jun 7, 2006, at 8:07 AM, Mike Fedyk wrote:Do you have multiple phones going down at the same time?  If so, monitor them with "qualify=500" in sip.conf to see if they hit that limit.  If you see more than one go down within a short period of time, you have network problems.  Check the quality of the network switches they have.  ___
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Re: [Asterisk-Users] GXP-2000

[Daniel Salama]

They don't all go down at the same time, or at least, my client hasn't noticed. I just added the qualify option. Let's see how that goes.As for the SPA-841, I have a client with a few of them and he cannot stop complaining about the bad audio quality. I replace a couple with a PAP-2 and another one with the GXP-2000 and he claims the quality to be incredibly better for both the PAP2 and the GXP-2000. He hasn't complained about the problems I mentioned on the GXP-2000 - yet :)Thanks,DanielOn Jun 7, 2006, at 8:07 AM, Mike Fedyk wrote:Do you have multiple phones going down at the same time?  If so, monitor them with "qualify=500" in sip.conf to see if they hit that limit.  If you see more than one go down within a short period of time, you have network problems.  Check the quality of the network switches they have.  Also I have heard some phones have trouble with broadcast packets (at least this has been said about the spa-841 on the wiki).  You should strongly consider putting them on a separate vlan to avoid any issues like that.  In the future, for phones under $100 then look at the spa-841 phones. ___
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Re: [Asterisk-Users] GXP-2000

[Mike Fedyk]

I have a client who has about six of these phones.  Luckily (for me, not 
for them) they were purchased before I came into the picture.


Daniel Salama wrote:
I have heard complaints from my client about the speakerphone and they 
are now
You don't notice any problems when using the speaker-phone, but the 
person on the other end hears echo, and quite a lot of it.

, I guess, getting used to picking up the handset :).
My client uses them exclusively with headsets (in a call center) so the 
quality of the speaker-phone isn't an issue for them.
I have heard any echo problems so far. What bothers me the most is 
that the phone stops working often (multiple times per day). By this I 
mean that my client won't be able to dial anything successfully. As 
soon as 3 or 4 digits are entered, they get a fast busy. To solve it, 
they need to reboot it. It sounds as if these phones were running 
Windows instead of Linux :)
Do you have multiple phones going down at the same time?  If so, monitor 
them with "qualify=500" in sip.conf to see if they hit that limit.  If 
you see more than one go down within a short period of time, you have 
network problems.  Check the quality of the network switches they have. 

Also I have heard some phones have trouble with broadcast packets (at 
least this has been said about the spa-841 on the wiki).  You should 
strongly consider putting them on a separate vlan to avoid any issues 
like that.  In the future, for phones under $100 then look at the 
spa-841 phones.


Anyway, what firmware did you use that solved so many of your problems?

http://www.voip-info.org/wiki/view/GXP-2000

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Re: [Asterisk-Users] GXP-2000

[Gareth Blades]

I am running 1.1.0.13 and there are no issues which are causing a
problem for us. The speakerphone is not much use but we can live with
that.

1.0.1.9 would stop registering after a while causing incoming calls to
go straight to voicemail. 
1.0.2.13 fixed this but had a bug where sometimes reviewing the missed
call list caused the phone to crash.

We have 35 handsets in use.

On Tue, 2006-06-06 at 21:11, Daniel Salama wrote:
> I'm using a few GXP-2000 with firmware 1.0.2.13 and everything seems  
> to be working fine. However, there are a couple of issues I'd like to  
> know if are possible:
> 
> 1) Even though the phone has 4 line appearances, if I am speaking on  
> a line, the phone can no longer receive phone calls. I can manually  
> select another line and make calls, but when Asterisk tries to send a  
> call to it, I see Got SIP response 486 "Busy" back on the console. Is  
> there a way to make the phone receive calls on all 4 lines?
> 
> 2) Is there any more documentation as to the tftp configuration file?
> 
> Thanks,
> Daniel
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Re: [Asterisk-Users] GXP-2000

[Thomas Kenyon]

Erick Baum wrote:
> The worst ongoing issue has been the echo and the really crappy
> speakerphone.  The customer is pretty much used to it now.  But we're
> slowly replacing them with Polycom's as new people come on and as
> others just get fed up.  Unfortunately one of the phones met it's
> doom by way of a hammer.  But I guess, what do you expect for under a
> hundred bucks.
Wow, I nearly bought some of these, but since the customer wouldn't pay
that much ended up getting some £30 chinese phones instead (not quite as
good spec. but sounds like they work at least as well).

Had no problems with the 2 at home, and so far (touch wood) the other 18
haven't had any major problems. Mind you no-one uses the speaker phone,
now if only I could get a headset for them.
>  
> Erick
>
>
>  
> On 6/6/06, *Daniel Salama* <[EMAIL PROTECTED]
> > wrote:
>
> I enabled call-waiting from the tftp configuration and it now works.
> What firmware are you using and where can I get it?
>
> My client complaints that the phone stops working every once in a
> while with no explanation. My client says that he could be using the
> phone with no problem and a few minutes later, when he wants to make
> a call, the phone will always give a fast busy after pressing the
> fourth digit. My workaround to him was to reboot the phone. That
> seems to solve the problem, however, it's not practical to have that
> problem in an office environment with 18 GXP-2000. Any ideas what the
> problem could be?
>
> Thanks,
> Daniel
>
> On Jun 6, 2006, at 6:26 PM, Mike wrote:
>
> > I can't say why you're having this problem, but I can tell you that
> > my phone
> > can receive (and make) multiple calls easily.  It might have more
> > to do with
> > Asterisk than the GXP2000.
> >
> > I am using the latest release firmware, not a beta.
> >
> > Mike
> >
> > -Original Message-
> > From: [EMAIL PROTECTED]
> 
> > [mailto:[EMAIL PROTECTED]
> ] On Behalf Of
> > Daniel Salama
> > Sent: June 6, 2006 4:12 PM
> > To: Non-Commercial Discussion Asterisk
> > Subject: [Asterisk-Users] GXP-2000
> >
> > I'm using a few GXP-2000 with firmware *MailScanner warning:
> numerical links are often malicious:* 1.0.2.13 
> and everything
> > seems to be
> > working fine. However, there are a couple of issues I'd like to
> > know if are
> > possible:
> >
> > 1) Even though the phone has 4 line appearances, if I am speaking
> > on a line,
> > the phone can no longer receive phone calls. I can manually select
> > another
> > line and make calls, but when Asterisk tries to send a call to it,
> > I see Got
> > SIP response 486 "Busy" back on the console. Is there a way to make
> > the
> > phone receive calls on all 4 lines?
> >
> > 2) Is there any more documentation as to the tftp configuration
> file?
> >
> > Thanks,
> > Daniel
> > ___
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> >
> >
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Re: [Asterisk-Users] GXP-2000

[Shaun Hofer]

I suggest you contact grandstream about this. Only thing I can suggest is look 
at feature's Early Dial (I have set to no) and No Key Entry Timeout (set to 
10-15 seconds). As for all these other problems of phone stop working, etc., 
we haven't come across these in office (then again we don't have 50 phones 
deployed). We have some phones running out of the box, stable release and 1 
with latest unstable (lots of nice features).

-Shaun

On Wednesday 07 June 2006 13:26, Daniel Salama wrote:
> Well, these are encouraging words :)
>
> You're basically telling me that I should tell my client to buy other
> phones. I agree that you cannot compare these phones with Cisco or
> Polycom. After all, like you said, what do you expect for under $90.
> However, the fact is that my client just recently invested in these
> and it will be hard, if not impossible, for me to tell my client to
> swap them for Polycoms or something else at a much higher cost.
>
> I have heard complaints from my client about the speakerphone and
> they are now, I guess, getting used to picking up the handset :). I
> have heard any echo problems so far. What bothers me the most is that
> the phone stops working often (multiple times per day). By this I
> mean that my client won't be able to dial anything successfully. As
> soon as 3 or 4 digits are entered, they get a fast busy. To solve it,
> they need to reboot it. It sounds as if these phones were running
> Windows instead of Linux :)
>
> Anyway, what firmware did you use that solved so many of your problems?
>
> Thanks,
> Daniel
>
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Re: [Asterisk-Users] GXP-2000

[Daniel Salama]

Well, these are encouraging words :)You're basically telling me that I should tell my client to buy other phones. I agree that you cannot compare these phones with Cisco or Polycom. After all, like you said, what do you expect for under $90. However, the fact is that my client just recently invested in these and it will be hard, if not impossible, for me to tell my client to swap them for Polycoms or something else at a much higher cost.I have heard complaints from my client about the speakerphone and they are now, I guess, getting used to picking up the handset :). I have heard any echo problems so far. What bothers me the most is that the phone stops working often (multiple times per day). By this I mean that my client won't be able to dial anything successfully. As soon as 3 or 4 digits are entered, they get a fast busy. To solve it, they need to reboot it. It sounds as if these phones were running Windows instead of Linux :)Anyway, what firmware did you use that solved so many of your problems?Thanks,DanielOn Jun 6, 2006, at 10:31 PM, Erick Baum wrote:We setup a company with 50 of these phones and had my client not been as understanding as they were, that could have put me out of business.  What an unbelievable nightmare.  This was about 8 months ago when the firmware was so bad the phone was a better paper weight than anything else.    Since then, they've fixed a lot of problems and made a lot of the features work like they're supposed to.  But we still have issues with them quite frequently.  From phones that need to be rebooted occationally, to ones that just drop calls, or do nothing when you pickup the receiver... lots of little qwerks.  We even experience their poor grounding problem every once in a while when you get a small static shock from the phone which cases it to reboot.  I don't think there's any firmware that can fix that.  We had to get several phones RMA'd because they just plain died.  The worst ongoing issue has been the echo and the really crappy speakerphone.  The customer is pretty much used to it now.  But we're slowly replacing them with Polycom's as new people come on and as others just get fed up.  Unfortunately one of the phones met it's doom by way of a hammer.  But I guess, what do you expect for under a hundred bucks.    Erick   On 6/6/06, Daniel Salama <[EMAIL PROTECTED]> wrote: I enabled call-waiting from the tftp configuration and it now works.What firmware are you using and where can I get it? My client complaints that the phone stops working every once in awhile with no explanation. My client says that he could be using thephone with no problem and a few minutes later, when he wants to make a call, the phone will always give a fast busy after pressing thefourth digit. My workaround to him was to reboot the phone. Thatseems to solve the problem, however, it's not practical to have thatproblem in an office environment with 18 GXP-2000. Any ideas what the problem could be?Thanks,DanielOn Jun 6, 2006, at 6:26 PM, Mike wrote:> I can't say why you're having this problem, but I can tell you that> my phone> can receive (and make) multiple calls easily.  It might have more > to do with> Asterisk than the GXP2000.>> I am using the latest release firmware, not a beta.>> Mike>> -Original Message-> From:  [EMAIL PROTECTED]> [mailto:[EMAIL PROTECTED]] On Behalf Of> Daniel Salama> Sent: June 6, 2006 4:12 PM > To: Non-Commercial Discussion Asterisk> Subject: [Asterisk-Users] GXP-2000>> I'm using a few GXP-2000 with firmware 1.0.2.13 and everything> seems to be > working fine. However, there are a couple of issues I'd like to> know if are> possible:>> 1) Even though the phone has 4 line appearances, if I am speaking> on a line,> the phone can no longer receive phone calls. I can manually select > another> line and make calls, but when Asterisk tries to send a call to it,> I see Got> SIP response 486 "Busy" back on the console. Is there a way to make> the> phone receive calls on all 4 lines? >> 2) Is there any more documentation as to the tftp configuration file?>> Thanks,> Daniel> ___> --Bandwidth and Colocation provided by Easynews.com -->> Asterisk-Users mailing list> To UNSUBSCRIBE or update options visit:>http://lists.digium.com/mailman/listinfo/asterisk-users >>> ___> --Bandwidth and Colocation provided by Easynews.com -->> Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit:>http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/a

Re: [Asterisk-Users] GXP-2000

[Eric \"ManxPower\" Wieling]

Erick Baum wrote:
We setup a company with 50 of these phones and had my client not been as 
understanding as they were, that could have put me out of business.  
What an unbelievable nightmare.  This was about 8 months ago when the 
firmware was so bad the phone was a better paper weight than anything else.


You did not experience these problems when you set up your prototype 
problems and did not see people reporting these issues when you searched 
the mailing lists?


--
Now accepting new clients in Birmingham, Atlanta, Huntsville, 
Chattanooga, and Montgomery.

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Re: [Asterisk-Users] GXP-2000

[Erick Baum]

We setup a company with 50 of these phones and had my client not been as understanding as they were, that could have put me out of business.  What an unbelievable nightmare.  This was about 8 months ago when the firmware was so bad the phone was a better paper weight than anything else.

 
Since then, they've fixed a lot of problems and made a lot of the features work like they're supposed to.  But we still have issues with them quite frequently.  From phones that need to be rebooted occationally, to ones that just drop calls, or do nothing when you pickup the receiver... lots of little qwerks.  We even experience their poor grounding problem every once in a while when you get a small static shock from the phone which cases it to reboot.  I don't think there's any firmware that can fix that.  We had to get several phones RMA'd because they just plain died.  The worst ongoing issue has been the echo and the really crappy speakerphone.  The customer is pretty much used to it now.  But we're slowly replacing them with Polycom's as new people come on and as others just get fed up.  Unfortunately one of the phones met it's doom by way of a hammer.  But I guess, what do you expect for under a hundred bucks.

 
Erick
 
On 6/6/06, Daniel Salama <[EMAIL PROTECTED]> wrote:
I enabled call-waiting from the tftp configuration and it now works.What firmware are you using and where can I get it?
My client complaints that the phone stops working every once in awhile with no explanation. My client says that he could be using thephone with no problem and a few minutes later, when he wants to make
a call, the phone will always give a fast busy after pressing thefourth digit. My workaround to him was to reboot the phone. Thatseems to solve the problem, however, it's not practical to have thatproblem in an office environment with 18 GXP-2000. Any ideas what the
problem could be?Thanks,DanielOn Jun 6, 2006, at 6:26 PM, Mike wrote:> I can't say why you're having this problem, but I can tell you that> my phone> can receive (and make) multiple calls easily.  It might have more
> to do with> Asterisk than the GXP2000.>> I am using the latest release firmware, not a beta.>> Mike>> -Original Message-> From: 
[EMAIL PROTECTED]> [mailto:[EMAIL PROTECTED]] On Behalf Of> Daniel Salama> Sent: June 6, 2006 4:12 PM
> To: Non-Commercial Discussion Asterisk> Subject: [Asterisk-Users] GXP-2000>> I'm using a few GXP-2000 with firmware 1.0.2.13 and everything> seems to be
> working fine. However, there are a couple of issues I'd like to> know if are> possible:>> 1) Even though the phone has 4 line appearances, if I am speaking> on a line,> the phone can no longer receive phone calls. I can manually select
> another> line and make calls, but when Asterisk tries to send a call to it,> I see Got> SIP response 486 "Busy" back on the console. Is there a way to make> the> phone receive calls on all 4 lines?
>> 2) Is there any more documentation as to the tftp configuration file?>> Thanks,> Daniel> ___> --Bandwidth and Colocation provided by 
Easynews.com -->> Asterisk-Users mailing list> To UNSUBSCRIBE or update options visit:>http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] GXP-2000

[Daniel Salama]

I enabled call-waiting from the tftp configuration and it now works.  
What firmware are you using and where can I get it?


My client complaints that the phone stops working every once in a  
while with no explanation. My client says that he could be using the  
phone with no problem and a few minutes later, when he wants to make  
a call, the phone will always give a fast busy after pressing the  
fourth digit. My workaround to him was to reboot the phone. That  
seems to solve the problem, however, it's not practical to have that  
problem in an office environment with 18 GXP-2000. Any ideas what the  
problem could be?


Thanks,
Daniel

On Jun 6, 2006, at 6:26 PM, Mike wrote:

I can't say why you're having this problem, but I can tell you that  
my phone
can receive (and make) multiple calls easily.  It might have more  
to do with

Asterisk than the GXP2000.

I am using the latest release firmware, not a beta.

Mike

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of  
Daniel Salama

Sent: June 6, 2006 4:12 PM
To: Non-Commercial Discussion Asterisk
Subject: [Asterisk-Users] GXP-2000

I'm using a few GXP-2000 with firmware 1.0.2.13 and everything  
seems to be
working fine. However, there are a couple of issues I'd like to  
know if are

possible:

1) Even though the phone has 4 line appearances, if I am speaking  
on a line,
the phone can no longer receive phone calls. I can manually select  
another
line and make calls, but when Asterisk tries to send a call to it,  
I see Got
SIP response 486 "Busy" back on the console. Is there a way to make  
the

phone receive calls on all 4 lines?

2) Is there any more documentation as to the tftp configuration file?

Thanks,
Daniel
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RE: [Asterisk-Users] GXP-2000

[Mike]

I can't say why you're having this problem, but I can tell you that my phone
can receive (and make) multiple calls easily.  It might have more to do with
Asterisk than the GXP2000.

I am using the latest release firmware, not a beta.

Mike

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Daniel Salama
Sent: June 6, 2006 4:12 PM
To: Non-Commercial Discussion Asterisk
Subject: [Asterisk-Users] GXP-2000

I'm using a few GXP-2000 with firmware 1.0.2.13 and everything seems to be
working fine. However, there are a couple of issues I'd like to know if are
possible:

1) Even though the phone has 4 line appearances, if I am speaking on a line,
the phone can no longer receive phone calls. I can manually select another
line and make calls, but when Asterisk tries to send a call to it, I see Got
SIP response 486 "Busy" back on the console. Is there a way to make the
phone receive calls on all 4 lines?

2) Is there any more documentation as to the tftp configuration file?

Thanks,
Daniel
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RE: [Asterisk-Users] GXP-2000 w/ 1.1.0.11 firmware

[Boris Bakchiev]

I had the same problem!
You have in your PXXX in your configs that 1.1.0.11 does not support.
Took me an hour to go through my configs and the web page to find what
PXXX in my configs unset the phone :)

Once its done, the phone will be accept the configs with no problems.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Ringwald
Sent: Wednesday, 17 May 2006 10:50
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] GXP-2000 w/ 1.1.0.11 firmware

I had provisioning via tftp working on this phone. I have verified that 
after the firmware upgrade, it contacts the tftp server and downloads 
the cfgMACADDR file, and the ring/etc files successfully. Unfortunately,

changes made to the config file don't make it to the phone (SIP account 
info/server info, etc).

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Re: [Asterisk-Users] GXP-2000 Message Waiting Light

[Peter Bowyer]

On 01/05/06, Jeffrey Macko <[EMAIL PROTECTED]> wrote:



Does anyone know the secret to get the GXP-2000 Message waiting lamp to
illuminate?


No secret - just set a 'mailbox' line in the appropriate peer entry in
sip.conf. Later GXP-2000 firmware shows the number of messages waiting
on the LCD display as well as flashing the MWI lamp (can't remember
which firmware version introduced this).

Peter

--
Peter Bowyer
Email: [EMAIL PROTECTED]
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RE: [Asterisk-Users] GXP-2000 phones stop registering

[Gareth Blades]

Mark,
Do you have the Flash Operator Panel or anything else installed?
I only had 1 phone stop registering in the first 2 weeks that I used
them and then after I installed FOP I had 3 phones stop registering in
the next couple of days.
I have now disabled FOP and have gone just over 2 days without any
problems.

Its probably just a coincidence but I am going to run without FOP for
another week and then try enabling it again.


On Mon, 2006-04-10 at 12:14, Mark Edwards wrote:
> Yes. Me.
> 
> I don't have a fix unfortunately - like you I seek one, however I have had a
> better experience by far though with the new 102x firmware branch. 
> 
> I would definitely recommend it to you.
> 
> Mark
> 
> -Original Message-
> From: Gareth Blades [mailto:[EMAIL PROTECTED] 
> Sent: Monday, 10 April 2006 8:49 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] GXP-2000 phones stop registering
> 
> I have about 30 GXP-2000 phones running 1.0.1.9 which have all been
> configured using the provisioning feature so the configuration is all
> identical.
> 
> The problem I am having is that they randomly seem to stop registering
> with asterisk. When they stop registering they can still make calls but
> oviously asterisk cannot ring the phone so all incoming calls go to
> voicemail.
> 
> Has anyone else had similar problems?
> 
> example sip.conf entry:-
> 
> 6015]
> type=friend
> secret=x
> username=6015
> callerid="users name" <6015>
> host=dynamic
> nat=no
> canreinvite=yes
> disallow=all
> allow=ulaw
> allow=alaw
> deny=0.0.0.0/0.0.0.0
> permit=10.0.0.0/255.0.0.0
> context=voipuk
> mailbox=6015
> 
> The phone config is fairly standard. the registration expiry is set to
> 60 minutes
> 
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RE: [Asterisk-Users] GXP-2000 phones stop registering

[Gareth Blades]

So the bug still exists in the 1.0.2 branch?

Thanks

On Mon, 2006-04-10 at 12:14, Mark Edwards wrote:
> Yes. Me.
> 
> I don't have a fix unfortunately - like you I seek one, however I have had a
> better experience by far though with the new 102x firmware branch. 
> 
> I would definitely recommend it to you.
> 
> Mark
> 
> -Original Message-
> From: Gareth Blades [mailto:[EMAIL PROTECTED] 
> Sent: Monday, 10 April 2006 8:49 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] GXP-2000 phones stop registering
> 
> I have about 30 GXP-2000 phones running 1.0.1.9 which have all been
> configured using the provisioning feature so the configuration is all
> identical.
> 
> The problem I am having is that they randomly seem to stop registering
> with asterisk. When they stop registering they can still make calls but
> oviously asterisk cannot ring the phone so all incoming calls go to
> voicemail.
> 
> Has anyone else had similar problems?
> 
> example sip.conf entry:-
> 
> 6015]
> type=friend
> secret=x
> username=6015
> callerid="users name" <6015>
> host=dynamic
> nat=no
> canreinvite=yes
> disallow=all
> allow=ulaw
> allow=alaw
> deny=0.0.0.0/0.0.0.0
> permit=10.0.0.0/255.0.0.0
> context=voipuk
> mailbox=6015
> 
> The phone config is fairly standard. the registration expiry is set to
> 60 minutes
> 
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RE: [Asterisk-Users] GXP-2000 phones stop registering

[Mark Edwards]

Yes. Me.

I don't have a fix unfortunately - like you I seek one, however I have had a
better experience by far though with the new 102x firmware branch. 

I would definitely recommend it to you.

Mark

-Original Message-
From: Gareth Blades [mailto:[EMAIL PROTECTED] 
Sent: Monday, 10 April 2006 8:49 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] GXP-2000 phones stop registering

I have about 30 GXP-2000 phones running 1.0.1.9 which have all been
configured using the provisioning feature so the configuration is all
identical.

The problem I am having is that they randomly seem to stop registering
with asterisk. When they stop registering they can still make calls but
oviously asterisk cannot ring the phone so all incoming calls go to
voicemail.

Has anyone else had similar problems?

example sip.conf entry:-

6015]
type=friend
secret=x
username=6015
callerid="users name" <6015>
host=dynamic
nat=no
canreinvite=yes
disallow=all
allow=ulaw
allow=alaw
deny=0.0.0.0/0.0.0.0
permit=10.0.0.0/255.0.0.0
context=voipuk
mailbox=6015

The phone config is fairly standard. the registration expiry is set to
60 minutes

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Re: [Asterisk-Users] GXP-2000 and Voicemail

[Waldo Rubinstein]

Thanks

Waldo

On Apr 9, 2006, at 2:19 PM, Tim Litwiller wrote:


it dials the userid that you put in that field as an extension.
at home I have it set to 100

and then I have this in the extensions.conf

exten => 100,1,Answer
exten => 100,2,Wait(1)
exten => 100,3,VoicemailMain,s${CALLERIDNUM}
exten => 100,4,Macro(hangupcall)

so the user doesn't need to put in a password when they press the  
MSG button



Waldo Rubinstein wrote:
Right, but it's asking for a user id not a number to dial. So, how  
would I set it to dial extension ?


Thanks,
Waldo

On Apr 9, 2006, at 12:21 PM, Harald Holzer wrote:


Look at the Account Settings for "Voice Mail UserID".



Hi,

I have a few GXP-2000 working fine with Asterisk. The one thing I
have not been able to do is to program the MSG button to dial the
Voicemail extension. How can I program that button? I normally use
extension  for voicemail. Can anyone shed any light?

Thanks,
Waldo

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Re: [Asterisk-Users] GXP-2000 and Voicemail

[Tim Litwiller]

it dials the userid that you put in that field as an extension.
at home I have it set to 100

and then I have this in the extensions.conf

exten => 100,1,Answer
exten => 100,2,Wait(1)
exten => 100,3,VoicemailMain,s${CALLERIDNUM}
exten => 100,4,Macro(hangupcall)

so the user doesn't need to put in a password when they press the MSG button


Waldo Rubinstein wrote:
Right, but it's asking for a user id not a number to dial. So, how 
would I set it to dial extension ?


Thanks,
Waldo

On Apr 9, 2006, at 12:21 PM, Harald Holzer wrote:


Look at the Account Settings for "Voice Mail UserID".



Hi,

I have a few GXP-2000 working fine with Asterisk. The one thing I
have not been able to do is to program the MSG button to dial the
Voicemail extension. How can I program that button? I normally use
extension  for voicemail. Can anyone shed any light?

Thanks,
Waldo

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Re: [Asterisk-Users] GXP-2000 and Voicemail

[Waldo Rubinstein]

Right, but it's asking for a user id not a number to dial. So, how  
would I set it to dial extension ?


Thanks,
Waldo

On Apr 9, 2006, at 12:21 PM, Harald Holzer wrote:


Look at the Account Settings for "Voice Mail UserID".



Hi,

I have a few GXP-2000 working fine with Asterisk. The one thing I
have not been able to do is to program the MSG button to dial the
Voicemail extension. How can I program that button? I normally use
extension  for voicemail. Can anyone shed any light?

Thanks,
Waldo

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Re: [Asterisk-Users] GXP-2000 and Voicemail

[Harald Holzer]

Look at the Account Settings for "Voice Mail UserID".


> Hi,
>
> I have a few GXP-2000 working fine with Asterisk. The one thing I
> have not been able to do is to program the MSG button to dial the
> Voicemail extension. How can I program that button? I normally use
> extension  for voicemail. Can anyone shed any light?
>
> Thanks,
> Waldo
>
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Re: [Asterisk-Users] GXP-2000 Volume Issue

[Clint Sharp]

I sent this from the wrong address and I don't think it went through.  I've just done some testing on the phone on 1.0.1.9 and 1.0.2.13.  The one on 
1.0.1.9 has no outbound gain issues, it is nominal with the rest of the phones in out office (Snom 320, Polcyom IP 301, and Budgetone 101).  However, this one on 1.0.2.13 caps at about a third of the meter on ztmonitor.  Is anyone else having this issue, or might this be a hardware issue with this particular phone?
ClintOn 3/1/06, Clint Sharp <[EMAIL PROTECTED]> wrote:
I have one on 1.0.2.13 and one on 
1.0.1.9.  The one on 1.0.2.13 is
the one I can imperically say is too quiet, the other appears to be
better.  I went back to 1.0.1.9 on the other because of a handset
volume issue.

ClintOn 3/1/06, Paul C <
[EMAIL PROTECTED]> wrote:







I had the opposite problem, I had to set txgain 
down as they were too loud and causing problems.


  - Original Message - 
  

From: 
  Clint 
  Sharp 
  To: 

Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Thursday, March 02, 2006 6:56 
  AM
  Subject: [Asterisk-Users] GXP-2000 Volume 
  Issue
  Is anyone else having an issue with GXP-2000s and transmit 
  gain?  All my other phones are fine on my TDM400P with txgain set at 0, 
  but the GXP-2000 caps at about a third of the scale in ztmonitor.  I'm 
  getting people complaining they can't hear me on my GXP-2000s, whereas my Snom 
  320 and Polycom 301 are great, and my Budgetones are overmodulating.  Is 
  there any conceivable fix on the Asterisk side, or does anyone know of any 
  gain adjustments that can be made to the GXP-2000s on either the older 1.0.1 
  series firmwares or the new 1.0.2 branches?Clint
  
  

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Re: [Asterisk-Users] GXP-2000 Volume Issue

[Paul C]

I had the opposite problem, I had to set txgain 
down as they were too loud and causing problems.

  - Original Message - 
  From: 
  Clint 
  Sharp 
  To: Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Thursday, March 02, 2006 6:56 
  AM
  Subject: [Asterisk-Users] GXP-2000 Volume 
  Issue
  Is anyone else having an issue with GXP-2000s and transmit 
  gain?  All my other phones are fine on my TDM400P with txgain set at 0, 
  but the GXP-2000 caps at about a third of the scale in ztmonitor.  I'm 
  getting people complaining they can't hear me on my GXP-2000s, whereas my Snom 
  320 and Polycom 301 are great, and my Budgetones are overmodulating.  Is 
  there any conceivable fix on the Asterisk side, or does anyone know of any 
  gain adjustments that can be made to the GXP-2000s on either the older 1.0.1 
  series firmwares or the new 1.0.2 branches?Clint
  
  

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Re: [Asterisk-Users] GXP-2000 fw 1.0.1.13 and NTP (SOLVED)

[Philip Edelbrock]

Tony Hoyle wrote:

Philip Edelbrock wrote:

 18  17.161118 Grandstr_05:a9:bf -> BroadcastARP Who has 
206.228.191.144?  Gratuitous ARP
 19  17.609869 3com_96:2f:eb -> Grandstr_05:a9:bf ARP 206.228.191.144 
is at 00:10:4b:96:2f:eb
 20  20.155260 206.228.191.144 -> 206.228.191.7 DHCP DHCP Decline  - 
Transaction ID 0xced0



It looks like your DHCP server is in fact broken.  It's passing out 
duplicate addresses - the device 00:10:4b:96:2f:eb already has 
206.228.191.144, so the Grandstream (correctly) declines the offer.


The server then tries to send the same address *again* instead of 
selecting a new one, and the same sequence ensues.  It should give a 
different address if the original one is declined.





Ah, you are close!

I figured it out (*hurray!*).  It was in fact a misconfiguration on my 
part.  144 isn't the end of my subnet, 143 is.  So, packet 18 is the 
phone confirming that it owns IP 144.  Packet 19 is from the router 
saying, "no you don't, I own that" (this is a proxy arp setup).  So, the 
phone declines and requests a new IP.  The head scratcher was that for 
the next request, it requests 144 again, so the DHCP server says (again) 
"OK, you got it" and the loop continues.


Once I adjusted my dhcp config to end my dynamic pool at 143 instead of 
144, all was well.


Additionally, I noticed that the phone requests these pieces of info in 
the dhcp response:


- Subnet
- Router
- DNS server(s)
- Time Server(s) <--- !!

So, I additionally put in the dhcp config a time server (the ip for 
time.nist.gov for now).  And after the first reboot, the phone gets an 
IP, pings the dhcp server once, registers, sets it's time, checks for 
firmware updates, and seems perfectly happy.


Hurray!


Phil
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Re: [Asterisk-Users] GXP-2000 fw 1.0.1.13 and NTP

[Tony Hoyle]

Philip Edelbrock wrote:

 18  17.161118 Grandstr_05:a9:bf -> BroadcastARP Who has 206.228.191.144?  
Gratuitous ARP
 19  17.609869 3com_96:2f:eb -> Grandstr_05:a9:bf ARP 206.228.191.144 is at 
00:10:4b:96:2f:eb
 20  20.155260 206.228.191.144 -> 206.228.191.7 DHCP DHCP Decline  - 
Transaction ID 0xced0


It looks like your DHCP server is in fact broken.  It's passing out 
duplicate addresses - the device 00:10:4b:96:2f:eb already has 
206.228.191.144, so the Grandstream (correctly) declines the offer.


The server then tries to send the same address *again* instead of 
selecting a new one, and the same sequence ensues.  It should give a 
different address if the original one is declined.


Tony
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Re: [Asterisk-Users] GXP-2000 fw 1.0.1.13 and NTP

[Philip Edelbrock]

Kristof Hardy wrote:
Was there a resolution to this issue?  The GXP-2000 seems to be a very 
popular phone, so I can't imagine others on the list not experiencing 
this?  Or is this part of a batch with unresolvable problems that I 
need to send back to the seller?



Well, I'm using dozens of these phones without this problem. What kind 
of DHCP/ntp server are you using? I'm using dnsmasq on a Debian box, 
together with the ntp-server. I'm using a mixture of 1.0.1.13 beta and 
.12 firmwares, both working correct.




The DHCP server is on the same 100BaseT switch as the phone right now 
(they are literally just a few feet away from each other).  DHCP server 
is on Fedora 3 Linux "Internet Systems Consortium DHCP Server V3.0.1" 
(from the rpm: dhcp-3.0.1-44_FC3).


Packet sniffer shows the phone getting in some sort of fight with the 
dhcp server.  I attached a text dump of the sniff.  You can see a 
repeating conversation from packet 20 to 40, and it continues on and on 
like that.


And, my logs are filling up with gazillions of these (pattern repeats 
every 3 seconds):
Jan 23 12:06:41 DrTheopolis dhcpd: DHCPDISCOVER from 00:0b:82:05:a9:bf 
via eth0
Jan 23 12:06:41 DrTheopolis dhcpd: DHCPOFFER on 206.228.191.144 to 
00:0b:82:05:a9:bf via eth0
Jan 23 12:06:41 DrTheopolis dhcpd: DHCPREQUEST for 206.228.191.144 
(206.228.191.7) from 00:0b:82:05:a9:bf via eth0
Jan 23 12:06:41 DrTheopolis dhcpd: DHCPACK on 206.228.191.144 to 
00:0b:82:05:a9:bf via eth0


While I was thinking of logs, I set up remote syslog for the phone, but 
all I see while it is set to dhcp is a single log noting the firmware 
versions on the phone.  With a static IP it logs info about registering 
w/ * (which it does successfully and I can make calls).



Phil
  1   0.00  0.0.0.0 -> 255.255.255.255 DHCP DHCP Discover - Transaction 
ID 0xaabbccdd
  2   0.727622 206.228.191.7 -> 206.228.191.144 DHCP DHCP Offer- 
Transaction ID 0xaabbccdd
  3   0.746653  0.0.0.0 -> 255.255.255.255 DHCP DHCP Request  - Transaction 
ID 0xaabbccde
  4   0.749231 206.228.191.7 -> 206.228.191.144 DHCP DHCP ACK  - 
Transaction ID 0xaabbccde
  5   0.766593 Grandstr_05:a9:bf -> BroadcastARP Who has 206.228.191.1?  
Tell 206.228.191.144
  6   0.997865 3com_96:2f:eb -> Grandstr_05:a9:bf ARP 206.228.191.1 is at 
00:10:4b:96:2f:eb
  7   1.308918 206.228.191.144 -> 206.228.191.7 DHCP DHCP Release  - 
Transaction ID 0xaabbccdf
  8  14.164223  0.0.0.0 -> 255.255.255.255 DHCP DHCP Discover - Transaction 
ID 0xcecb
  9  14.164531 206.228.191.7 -> 206.228.191.144 DHCP DHCP Offer- 
Transaction ID 0xcecb
 10  14.166809  0.0.0.0 -> 255.255.255.255 DHCP DHCP Request  - Transaction 
ID 0xcecc
 11  14.172534 206.228.191.7 -> 206.228.191.144 DHCP DHCP ACK  - 
Transaction ID 0xcecc
 12  14.175408 Grandstr_05:a9:bf -> BroadcastARP Who has 206.228.191.144?  
Gratuitous ARP
 13  14.339375 3com_96:2f:eb -> Grandstr_05:a9:bf ARP 206.228.191.144 is at 
00:10:4b:96:2f:eb
 14  17.155641 206.228.191.144 -> 255.255.255.255 DHCP DHCP Discover - 
Transaction ID 0xcece
 15  17.155975 206.228.191.7 -> 206.228.191.144 DHCP DHCP Offer- 
Transaction ID 0xcece
 16  17.158134 206.228.191.144 -> 255.255.255.255 DHCP DHCP Request  - 
Transaction ID 0xcecf
 17  17.159263 206.228.191.7 -> 206.228.191.144 DHCP DHCP ACK  - 
Transaction ID 0xcecf
 18  17.161118 Grandstr_05:a9:bf -> BroadcastARP Who has 206.228.191.144?  
Gratuitous ARP
 19  17.609869 3com_96:2f:eb -> Grandstr_05:a9:bf ARP 206.228.191.144 is at 
00:10:4b:96:2f:eb
 20  20.155260 206.228.191.144 -> 206.228.191.7 DHCP DHCP Decline  - 
Transaction ID 0xced0
 21  20.155760 206.228.191.144 -> 255.255.255.255 DHCP DHCP Discover - 
Transaction ID 0xced1
 22  20.155981 206.228.191.7 -> 206.228.191.144 DHCP DHCP Offer- 
Transaction ID 0xced1
 23  20.158255 206.228.191.144 -> 255.255.255.255 DHCP DHCP Request  - 
Transaction ID 0xced2
 24  20.159714 206.228.191.7 -> 206.228.191.144 DHCP DHCP ACK  - 
Transaction ID 0xced2
 25  20.161242 Grandstr_05:a9:bf -> BroadcastARP Who has 206.228.191.144?  
Gratuitous ARP
 26  20.640088 3com_96:2f:eb -> Grandstr_05:a9:bf ARP 206.228.191.144 is at 
00:10:4b:96:2f:eb
 27  23.165159 206.228.191.144 -> 206.228.191.7 DHCP DHCP Decline  - 
Transaction ID 0xced3
 28  23.165658 206.228.191.144 -> 255.255.255.255 DHCP DHCP Discover - 
Transaction ID 0xced4
 29  23.165879 206.228.191.7 -> 206.228.191.144 DHCP DHCP Offer- 
Transaction ID 0xced4
 30  23.168148 206.228.191.144 -> 255.255.255.255 DHCP DHCP Request  - 
Transaction ID 0xced5
 31  23.170237 206.228.191.7 -> 206.228.191.144 DHCP DHCP ACK  - 
Transaction ID 0xced5
 32  23.172210 Grandstr_05:a9:bf -> BroadcastARP Who has 206.228.191.144?  
Gratuitous ARP
 33  23.180374 3com_96:2f:eb -> Grandstr_05:a9:bf ARP 206.228.191.144 is at 
00:10:4b:96:2f:eb
 34  26.165097 206.228.191.144 -> 206.228.191.7 DHCP DHCP Decline  - 
T

RE: [Asterisk-Users] GXP-2000 fw 1.0.1.13 and NTP

[Lee Archer]

Odd you should have this problem as I had exactly the same.  In my case
it was a slow DHCP server.  Around 7 seconds in the phones tries to time
sync.  If the phone hasn't got an IP address then this time sync fails
but it doesn't retry.  I emailed Grandstream about it but got nowhere.
I changed my DHCP server from Windows to Linux and now DHCP is much
faster and time sync is working.

Regards

Lee

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Philip
Edelbrock
Sent: 21 January 2006 06:03
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] GXP-2000 fw 1.0.1.13 and NTP


On Dec 31, 2005, at 7:28 AM, Ross C wrote:

> Peter,
>
> After upgrading to 1.0.1.13 I had some miscellaneous problems on one 
> of my GXP-2000's--it would grab an IP address, but it wouldn't get the

> time/date, it wouldn't register, blah blah blah.  I could access the 
> web interface OK, so it wasn't a network issue (I don't think).  
> Anyway...I ended up resetting to factory defaults and all is well now.

> Maybe try that?  That has solved some other problems I've had as well.

I just got a 2000 which does exactly this (our first for evaluation..  
which is somewhat disappointing thus far).  I could see in a packet
sniffer a weird cycle of DHCP requests like it got an IP but kept
retrying?  A power cycle doesn't solve the problem (it's had many, and
dozens of software resets).  A reset with the MAC input doesn't work
either for me.  The phone was at an older FW  when I got it (ending in
.9, I think) and then updated to to the latest stable (.12 I think off
the top of my head).  Btw- the firmware update was a pain.  HTTP updates
were hitting the server (Apache) with 'bad request' results.  I needed
to set up my own tfpt server to make it work.  Off lan updates weren't
working, either, in any case.

The phone will register and work when it has a static address assigned,
but not when set for DHCP.  In all cases, the clock is always wrong.  I
can see with a packet sniffer that the NTP request is sent and received,
but with no effect on the phone display.

Was there a resolution to this issue?  The GXP-2000 seems to be a very
popular phone, so I can't imagine others on the list not experiencing
this?  Or is this part of a batch with unresolvable problems that I need
to send back to the seller?

Thanks! TGIF! :')


Phil
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Re: [Asterisk-Users] GXP-2000 fw 1.0.1.13 and NTP

[Kristof Hardy]

Was there a resolution to this issue?  The GXP-2000 seems to be a very 
popular phone, so I can't imagine others on the list not experiencing 
this?  Or is this part of a batch with unresolvable problems that I need 
to send back to the seller?


Well, I'm using dozens of these phones without this problem. What kind 
of DHCP/ntp server are you using? I'm using dnsmasq on a Debian box, 
together with the ntp-server. I'm using a mixture of 1.0.1.13 beta and 
.12 firmwares, both working correct.



Thanks! TGIF! :')


Hell yeah! :D

cheers

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RE: [Asterisk-Users] GXP-2000 fw 1.0.1.13 and NTP

[Ross C]

I have this problem intermittently.  I haven't found a solution yet.  I
don't suppose you've tried using a different NTP server?  I haven't tried
that either, but I might this afternoon.  For me, it's hard to troubleshoot
because it doesn't happen all the time.

-ross

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Philip
Edelbrock
Sent: Saturday, January 21, 2006 12:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] GXP-2000 fw 1.0.1.13 and NTP


On Dec 31, 2005, at 7:28 AM, Ross C wrote:

> Peter,
>
> After upgrading to 1.0.1.13 I had some miscellaneous problems on  
> one of my
> GXP-2000's--it would grab an IP address, but it wouldn't get the  
> time/date,
> it wouldn't register, blah blah blah.  I could access the web  
> interface OK,
> so it wasn't a network issue (I don't think).  Anyway...I ended up  
> resetting
> to factory defaults and all is well now.  Maybe try that?  That has  
> solved
> some other problems I've had as well.

I just got a 2000 which does exactly this (our first for evaluation..  
which is somewhat disappointing thus far).  I could see in a packet  
sniffer a weird cycle of DHCP requests like it got an IP but kept  
retrying?  A power cycle doesn't solve the problem (it's had many,  
and dozens of software resets).  A reset with the MAC input doesn't  
work either for me.  The phone was at an older FW  when I got it  
(ending in .9, I think) and then updated to to the latest stable (.12  
I think off the top of my head).  Btw- the firmware update was a  
pain.  HTTP updates were hitting the server (Apache) with 'bad  
request' results.  I needed to set up my own tfpt server to make it  
work.  Off lan updates weren't working, either, in any case.

The phone will register and work when it has a static address  
assigned, but not when set for DHCP.  In all cases, the clock is  
always wrong.  I can see with a packet sniffer that the NTP request  
is sent and received, but with no effect on the phone display.

Was there a resolution to this issue?  The GXP-2000 seems to be a  
very popular phone, so I can't imagine others on the list not  
experiencing this?  Or is this part of a batch with unresolvable  
problems that I need to send back to the seller?

Thanks! TGIF! :')


Phil
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Re: [Asterisk-Users] GXP-2000 fw 1.0.1.13 and NTP

[Philip Edelbrock]

On Dec 31, 2005, at 7:28 AM, Ross C wrote:


Peter,

After upgrading to 1.0.1.13 I had some miscellaneous problems on  
one of my
GXP-2000's--it would grab an IP address, but it wouldn't get the  
time/date,
it wouldn't register, blah blah blah.  I could access the web  
interface OK,
so it wasn't a network issue (I don't think).  Anyway...I ended up  
resetting
to factory defaults and all is well now.  Maybe try that?  That has  
solved

some other problems I've had as well.


I just got a 2000 which does exactly this (our first for evaluation..  
which is somewhat disappointing thus far).  I could see in a packet  
sniffer a weird cycle of DHCP requests like it got an IP but kept  
retrying?  A power cycle doesn't solve the problem (it's had many,  
and dozens of software resets).  A reset with the MAC input doesn't  
work either for me.  The phone was at an older FW  when I got it  
(ending in .9, I think) and then updated to to the latest stable (.12  
I think off the top of my head).  Btw- the firmware update was a  
pain.  HTTP updates were hitting the server (Apache) with 'bad  
request' results.  I needed to set up my own tfpt server to make it  
work.  Off lan updates weren't working, either, in any case.


The phone will register and work when it has a static address  
assigned, but not when set for DHCP.  In all cases, the clock is  
always wrong.  I can see with a packet sniffer that the NTP request  
is sent and received, but with no effect on the phone display.


Was there a resolution to this issue?  The GXP-2000 seems to be a  
very popular phone, so I can't imagine others on the list not  
experiencing this?  Or is this part of a batch with unresolvable  
problems that I need to send back to the seller?


Thanks! TGIF! :')


Phil
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RE: [Asterisk-Users] GXP-2000 fw 1.0.1.13 and NTP

[Lee Archer]

I had a problem which I spoke to Grandstream about.  It seemed that
around 7 seconds in it goes for time sync and if it fails it doesn't
retry.  This problem was highlighted by the .12 firmware and a Windows
DHCP server we were using.  Upon moving to a Linux DHCP server the
process was much quicker and NTP worked.  However there isn't an auto
DST mode  This upset a lot of people here where I work as all the
clocks were wrong.  Shame is these are reasonably cheap and fairly
descent phones but we are now moving towards the Aastra range.

I've tried out .13 and NTP worked fine. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter
Bowyer
Sent: 31 December 2005 10:35
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] GXP-2000 fw 1.0.1.13 and NTP

Hi all

Slightly OT but I know a lot of GS experts hang out here - I just
upgraded a GXP-2000 to firmware 1.0.1.13 to try out the BLF
functionality with Asterisk (which so far works as expected), but as a
side-effect the phone won't sync with an NTP server - I've tried
different server names (time.nist.gov and
pool.ntp.org)  and IPs in the config, but it refuses to update the time
on the display.

Anyone heard of this? Any workarounds (other than go back to 1.0.1.12) ?

(Hmmm.. just regressed to 1.0.1.12 and it's still not working -
curiouser and curiouser said Alice...)

Thanks

Peter 

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RE: [Asterisk-Users] GXP-2000 any good with * ?

[Ross C]

Oooh!
So it's on backorder @ voipsupply...

If that's the case,
To the voipsupply folks:
The red text on the VoipSupply site is worded to kind of imply that the 320
isn't available anymore.  It should mention something about "order it now,
and we'll notify you when we have them" or "they're *currently* not
available, but will be soon; if you need something immediately contact us
and we'll recommend an alternative"  not  "they're not available...so look
at some other phone"

Thx Christian!!

-ross
-Original Message-
From: Christian Stredicke [mailto:[EMAIL PROTECTED] 
Sent: Monday, January 02, 2006 2:19 PM
To: Ross C
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] GXP-2000 any good with * ?

My understanding is that there is currently a shortage of phones at
voipsupply (and also in other places). The 320 is selling pretty good :-)
and we are making the biggest production run *ever* this month!

snom does not discontinue the 320! 

Christian

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of Ross C
> Sent: Monday, January 02, 2006 12:46 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] GXP-2000 any good with * ?
> 
> http://www.voipsupply.com/product_info.php?cPath=95_114&produc
> ts_id=883
> am I misreading something?
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Sven Fischer
> (support)
> Sent: Monday, January 02, 2006 11:11 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] GXP-2000 any good with * ?
> 
> This doesn't seem to be correct, too...
> 
> Sven
> 
> On Monday 02 January 2006 17:43, Ross C wrote:
> > Sorry!!
> > Just discontinued @ voipsupply.com I guess.
> > Thx for the correction.
> >
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of Sven 
> > Fischer
> > (support)
> > Sent: Monday, January 02, 2006 2:48 AM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] GXP-2000 any good with * ?
> >
> > On Saturday 31 December 2005 01:57, Ross C wrote:
> > > ... and 2 Snom 320's (now discontinued I think).
> >
> > No, they are not discontinued !!!
> >
> > Regards,
> >
> > Sven
> > ___
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> >
> > Asterisk-Users mailing list
> > To UNSUBSCRIBE or update options visit:
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> > Asterisk-Users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> --
> --
> -
> See our FAQs at: http://www.snom.com/faq0.html?&L=1
> Whitepapers at:  http://www.snom.com/white_papers.html
> --
> -
> snom technology AG   Gradestraße 46 D-12347 Berlin
> Sven Fischer fax +49 30 39833111
> mailto:[EMAIL PROTECTED]   http://www.snom.com
> --
> -
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> 
> 

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RE: [Asterisk-Users] GXP-2000 any good with * ?

[Christian Stredicke]

My understanding is that there is currently a shortage of phones at voipsupply 
(and also in other places). The 320 is selling pretty good :-) and we are 
making the biggest production run *ever* this month!

snom does not discontinue the 320! 

Christian

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of Ross C
> Sent: Monday, January 02, 2006 12:46 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] GXP-2000 any good with * ?
> 
> http://www.voipsupply.com/product_info.php?cPath=95_114&produc
> ts_id=883
> am I misreading something?
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Sven Fischer
> (support)
> Sent: Monday, January 02, 2006 11:11 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] GXP-2000 any good with * ?
> 
> This doesn't seem to be correct, too...
> 
> Sven
> 
> On Monday 02 January 2006 17:43, Ross C wrote:
> > Sorry!!
> > Just discontinued @ voipsupply.com I guess.
> > Thx for the correction.
> >
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of Sven 
> > Fischer
> > (support)
> > Sent: Monday, January 02, 2006 2:48 AM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] GXP-2000 any good with * ?
> >
> > On Saturday 31 December 2005 01:57, Ross C wrote:
> > > ... and 2 Snom 320's (now discontinued I think).
> >
> > No, they are not discontinued !!!
> >
> > Regards,
> >
> > Sven
> > ___
> > --Bandwidth and Colocation provided by Easynews.com --
> >
> > Asterisk-Users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> > ___
> > --Bandwidth and Colocation provided by Easynews.com --
> >
> > Asterisk-Users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> --
> --
> -
> See our FAQs at: http://www.snom.com/faq0.html?&L=1
> Whitepapers at:  http://www.snom.com/white_papers.html
> --
> -
> snom technology AG   Gradestraße 46 D-12347 Berlin
> Sven Fischer fax +49 30 39833111
> mailto:[EMAIL PROTECTED]   http://www.snom.com
> --
> -
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> 
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RE: [Asterisk-Users] GXP-2000 any good with * ?

[Ross C]

http://www.voipsupply.com/product_info.php?cPath=95_114&products_id=883
am I misreading something?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sven Fischer
(support)
Sent: Monday, January 02, 2006 11:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] GXP-2000 any good with * ?

This doesn't seem to be correct, too...

Sven

On Monday 02 January 2006 17:43, Ross C wrote:
> Sorry!!
> Just discontinued @ voipsupply.com I guess.
> Thx for the correction.
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Sven Fischer
> (support)
> Sent: Monday, January 02, 2006 2:48 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] GXP-2000 any good with * ?
>
> On Saturday 31 December 2005 01:57, Ross C wrote:
> > ... and 2 Snom 320's (now discontinued I think).
>
> No, they are not discontinued !!!
>
> Regards,
>
> Sven
> ___
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-- 
---
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Whitepapers at:  http://www.snom.com/white_papers.html
---
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Sven Fischer fax +49 30 39833111
mailto:[EMAIL PROTECTED]   http://www.snom.com
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Re: [Asterisk-Users] GXP-2000 any good with * ?

[Sven Fischer (support)]

This doesn't seem to be correct, too...

Sven

On Monday 02 January 2006 17:43, Ross C wrote:
> Sorry!!
> Just discontinued @ voipsupply.com I guess.
> Thx for the correction.
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Sven Fischer
> (support)
> Sent: Monday, January 02, 2006 2:48 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] GXP-2000 any good with * ?
>
> On Saturday 31 December 2005 01:57, Ross C wrote:
> > ... and 2 Snom 320's (now discontinued I think).
>
> No, they are not discontinued !!!
>
> Regards,
>
> Sven
> ___
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-- 
---
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Whitepapers at:  http://www.snom.com/white_papers.html
---
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RE: [Asterisk-Users] GXP-2000 any good with * ?

[Ross C]

Sorry!!
Just discontinued @ voipsupply.com I guess.  
Thx for the correction.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sven Fischer
(support)
Sent: Monday, January 02, 2006 2:48 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] GXP-2000 any good with * ?

On Saturday 31 December 2005 01:57, Ross C wrote:
> ... and 2 Snom 320's (now discontinued I think).  

No, they are not discontinued !!! 

Regards,

Sven
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Re: [Asterisk-Users] GXP-2000 any good with * ?

[Sven Fischer (support)]

On Saturday 31 December 2005 01:57, Ross C wrote:
> ... and 2 Snom 320's (now discontinued I think).  

No, they are not discontinued !!! 

Regards,

Sven
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Re: [Asterisk-Users] GXP-2000 fw 1.0.1.13 and NTP

[Leif Neland]

 Original Message 
From: "Peter Bowyer" <[EMAIL PROTECTED]>
To: 
Sent: Saturday, December 31, 2005 11:34 AM
Subject: [Asterisk-Users] GXP-2000 fw 1.0.1.13 and NTP


Hi all

Slightly OT but I know a lot of GS experts hang out here - I just
upgraded a GXP-2000 to firmware 1.0.1.13 to try out the BLF
functionality with Asterisk (which so far works as expected), but as
a side-effect the phone won't sync with an NTP server - I've tried
different server names (time.nist.gov and pool.ntp.org)  and IPs in
the config, but it refuses to update the time on the display.

Anyone heard of this? Any workarounds (other than go back to
1.0.1.12) ?
(Hmmm.. just regressed to 1.0.1.12 and it's still not working -
curiouser and curiouser said Alice...)



My GS BT101 have also developed problems with sync'ing to my ntp-server.
I can see, using tcpdump, that the phone asks my server and gets an answer, 
but the display is not updated.
It used to work, but now it usually doesn't, but strangely, sometime it 
does...


Leif

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RE: [Asterisk-Users] GXP-2000 fw 1.0.1.13 and NTP

[Ross C]

Peter,

After upgrading to 1.0.1.13 I had some miscellaneous problems on one of my
GXP-2000's--it would grab an IP address, but it wouldn't get the time/date,
it wouldn't register, blah blah blah.  I could access the web interface OK,
so it wasn't a network issue (I don't think).  Anyway...I ended up resetting
to factory defaults and all is well now.  Maybe try that?  That has solved
some other problems I've had as well.
I dunno if ur familiar with the process, but it's kinda screwy, here's the
info:
http://www.grandstream.com/user_manuals/GXP2000.pdf
scroll down to the very last page.
Do the other phones on the same network get the time and date OK from
time.nist.gov?

-ross

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter Bowyer
Sent: Saturday, December 31, 2005 4:35 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] GXP-2000 fw 1.0.1.13 and NTP

Hi all

Slightly OT but I know a lot of GS experts hang out here - I just upgraded a

GXP-2000 to firmware 1.0.1.13 to try out the BLF functionality with Asterisk

(which so far works as expected), but as a side-effect the phone won't sync 
with an NTP server - I've tried different server names (time.nist.gov and 
pool.ntp.org)  and IPs in the config, but it refuses to update the time on 
the display.

Anyone heard of this? Any workarounds (other than go back to 1.0.1.12) ?

(Hmmm.. just regressed to 1.0.1.12 and it's still not working - curiouser 
and curiouser said Alice...)

Thanks

Peter 

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Re: [Asterisk-Users] GXP-2000 fw 1.0.1.13 and NTP

[Kristof Hardy]

Peter Bowyer wrote:
side-effect the phone won't sync with an NTP server - I've tried 
different server names (time.nist.gov and pool.ntp.org)  and IPs in the 
config, but it refuses to update the time on the display.


No problem here. Using the 1.0.1.13 (very beta:)) also, synching with an 
internal time server on our network. (wich then syncs to pool.ntp.org)


cheers
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Re: [Asterisk-Users] GXP-2000 any good with * ?

[Kristof Hardy]

Michiel van Baak wrote:

Hinting works fine for me with the latest firmware.

What version are you running?
We use 1.0.1.9 but the leds next to the speeddials wont


use latest * and latest gxp firmware, have a look here on how to do it:
http://www.voip-info.org/wiki/view/GXP-2000
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Re: [Asterisk-Users] GXP-2000 any good with * ?

[Tom Vile]

My VM soft key works fine with SIP but I am thinking I will give sccp
a try.  Thanks for the info.
On 12/30/05, Michiel van Baak <[EMAIL PROTECTED]> wrote:
> On 20:25, Fri 30 Dec 05, Tom Vile wrote:
> > I see,  I am not using skinny but maybe I should give it a try.  Why
> > is it better than the SIP image?
>
> There are several reasons why it's better.
>
> 1. It's a lot faster. the phone menus are more responsive
> and a restart is done in less then half the time a SIP image
> takes.
> 2. The XML support is better. SIP doesn't support 100% of
> the documented XML and SCCP/Skinny does.
> 3. the hinting system works like a real hinting should work,
> look at the images on the page I posted in my previous mail
> 4. the TFTP config file is plain XML, no more special
> program to convert the txt config into some weird binary
> file.
> 5. speeddial/hinting is working when provisioned from TFTP.
> when using sip you have to config them on the phone to work
> at all.
>
> There are some drawbacks of course.
> The callforwarding stuff works not as smooth in the SCCP
> version. It works great, but setting callforward works like
> this:
> make a call to the number you want to forward to. While it's
> ringing hit CFWD[ALL|BUSY] to activate it.
> This is the only annoying thing I noticed with the SCCP
> image. the 5 point above (and they are just what I
> experienced) make up for that 100%.
>
> Dont know if sip supports this, but the softkey "toVM"
> actually works in the SCCP image, I never got it to work in
> the SIP image. this would add number 6 to the list above.
>
> There're prolly more reasons why SCCP is better.
> Stefan Gofferje and Sergio can prolly give you more reasons.
>
> Greetz and a happy newyear.
> --
> Michiel van Baak
> http://michiel.vanbaak.info
> [EMAIL PROTECTED]
> GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D
>
> "Why is it drug addicts and computer afficionados are both called users?"
>
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--
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Fax: 518-631-2856
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Re: [Asterisk-Users] GXP-2000 any good with * ?

[Michiel van Baak]

On 20:25, Fri 30 Dec 05, Tom Vile wrote:
> I see,  I am not using skinny but maybe I should give it a try.  Why
> is it better than the SIP image?

There are several reasons why it's better.

1. It's a lot faster. the phone menus are more responsive
and a restart is done in less then half the time a SIP image
takes.
2. The XML support is better. SIP doesn't support 100% of
the documented XML and SCCP/Skinny does.
3. the hinting system works like a real hinting should work,
look at the images on the page I posted in my previous mail
4. the TFTP config file is plain XML, no more special
program to convert the txt config into some weird binary
file.
5. speeddial/hinting is working when provisioned from TFTP.
when using sip you have to config them on the phone to work
at all.

There are some drawbacks of course.
The callforwarding stuff works not as smooth in the SCCP
version. It works great, but setting callforward works like
this:
make a call to the number you want to forward to. While it's
ringing hit CFWD[ALL|BUSY] to activate it.
This is the only annoying thing I noticed with the SCCP
image. the 5 point above (and they are just what I
experienced) make up for that 100%.

Dont know if sip supports this, but the softkey "toVM"
actually works in the SCCP image, I never got it to work in
the SIP image. this would add number 6 to the list above.

There're prolly more reasons why SCCP is better.
Stefan Gofferje and Sergio can prolly give you more reasons.

Greetz and a happy newyear.
-- 
Michiel van Baak
http://michiel.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D

"Why is it drug addicts and computer afficionados are both called users?"

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