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

[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

[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

[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

[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

[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

[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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> >http://lists.digium.com/mailman/listinfo/asterisk-users
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> >
> >
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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

[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

[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

[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

[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]

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

[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

[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

[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]

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]

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

[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

[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

[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

[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

[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

[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

[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

[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

[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

[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

[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

[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

[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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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 MWI

[Peter Bowyer]

On 10/07/05, Mark Edwards <[EMAIL PROTECTED]> wrote:
> anyone managed to get MWI going on the GXP-2000 with * CVS-HEAD? I have set
> up the "mailbox" in the sip.conf entries but no flashing lights... SIP
> NOTIFY seems to be being sent out...

Yes, works like a charm here. Firmware 1.0.1.9.

Peter

-- 
Peter Bowyer
Email: [EMAIL PROTECTED]
Tel: +44 1296 768003
VoIP: sip:[EMAIL PROTECTED]
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Re: [Asterisk-Users] GXP-2000 MWI

[Mark Edwards]

Thanks mate - I had my voicemail context set up wrong
 
cheers - works a treat for me too! ;-)
 
Mark 
On 7/11/05, Peter Bowyer <[EMAIL PROTECTED]> wrote:
On 10/07/05, Mark Edwards <[EMAIL PROTECTED]> wrote:
> anyone managed to get MWI going on the GXP-2000 with * CVS-HEAD? I have set> up the "mailbox" in the sip.conf entries but no flashing lights... SIP> NOTIFY seems to be being sent out...
Yes, works like a charm here. Firmware 1.0.1.9.Peter--Peter BowyerEmail: [EMAIL PROTECTED]Tel: +44 1296 768003VoIP: 
sip:[EMAIL PROTECTED]___Asterisk-Users mailing listAsterisk-Users@lists.digium.com
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-- regards,Mark P. EdwardsFWD: 667917
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Re: [Asterisk-Users] GXP-2000 presence

[Harald Holzer]

> Hi All,
>
> Just wondering if anyone has managed to get line presence working on the
> 7 indicator lights on a grandstream gxp-2000 with asterisk? If so, what
> is the trick? :)

last week i asked the grandstream support for this, and got this short answer:

>> This feature is not supported yet, it will be supported in the future.


> I have simple presence working with my polycom phones but cant seem to
> get it working with the gxp-2000 - is it available in the latest
> firmware or is it something that will be released later on? Or is there
> something tricky i need to do on teh * side?
>
> Cheers,
>
> Ben
>
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RE: [Asterisk-Users] GXP-2000 presence

[Anton Krall]

Speaking of GS..

I know polycom phones can eb rebooted with some script using sip_notify.

Can GS phones do this also? 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Harald Holzer
|Sent: Lunes, 29 de Agosto de 2005 01:09 p.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] GXP-2000 presence
|
|> Hi All,
|>
|> Just wondering if anyone has managed to get line presence working on 
|> the
|> 7 indicator lights on a grandstream gxp-2000 with asterisk? If so, 
|> what is the trick? :)
|
|last week i asked the grandstream support for this, and got 
|this short answer:
|
|>> This feature is not supported yet, it will be supported in 
|the future.
|
|
|> I have simple presence working with my polycom phones but 
|cant seem to 
|> get it working with the gxp-2000 - is it available in the latest 
|> firmware or is it something that will be released later on? Or is 
|> there something tricky i need to do on teh * side?
|>
|> Cheers,
|>
|> Ben
|>
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|
|
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RE: [Asterisk-Users] GXP-2000 presence

[Peter Svensson]

On Tue, 30 Aug 2005, Anton Krall wrote:

> Speaking of GS..
> 
> I know polycom phones can eb rebooted with some script using sip_notify.
> 
> Can GS phones do this also? 

You can reset the phones by requesting the right page from their built in 
web server as long as you know the admin password.

Peter

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RE: [Asterisk-Users] GXP-2000 presence

[Anton Krall]

I got that under control :) but was wondering if it could be done using
sip_notify :( 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Peter Svensson
|Sent: Martes, 30 de Agosto de 2005 12:54 p.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: RE: [Asterisk-Users] GXP-2000 presence
|
|On Tue, 30 Aug 2005, Anton Krall wrote:
|
|> Speaking of GS..
|> 
|> I know polycom phones can eb rebooted with some script using 
|sip_notify.
|> 
|> Can GS phones do this also? 
|
|You can reset the phones by requesting the right page from 
|their built in web server as long as you know the admin password.
|
|Peter
|
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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 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

[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 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 tftp cfg

[Peter Svensson]

On Tue, 7 Jun 2005, marek cervenka wrote:

> can you someone post tftp template for gxp-2000?
> like 
> http://www.grandstream.com/DOWNLOAD/Configuration_Tool/Windows/Grandstream_Configuration_File_Template_1.0.6.x.txt

I think it will be released with the 1.0.1.9 firmware. You may be able to 
get it by asking their support for it. YMMW.

Peter

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RE: [Asterisk-Users] gxp-2000 tftp cfg

[David Phelan]

 If you download the "configuration tool" which I couldn't get working on my
systemthere is a cfg template in there for 1.0.1.8


Dave


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter Svensson
Sent: Wednesday, 8 June 2005 7:26 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] gxp-2000 tftp cfg

On Tue, 7 Jun 2005, marek cervenka wrote:

> can you someone post tftp template for gxp-2000?
> like
> http://www.grandstream.com/DOWNLOAD/Configuration_Tool/Windows/Grandst
> ream_Configuration_File_Template_1.0.6.x.txt

I think it will be released with the 1.0.1.9 firmware. You may be able to
get it by asking their support for it. YMMW.

Peter

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RE: [Asterisk-Users] gxp-2000 tftp cfg

[Peter Svensson]

On Wed, 8 Jun 2005, David Phelan wrote:

>  If you download the "configuration tool" which I couldn't get working on my
> systemthere is a cfg template in there for 1.0.1.8

Oh, then they have added it, or we missed it the first time around. We 
have it running. We had to tweak the paths in the file "encode.sh" a bit 
to match our setup. 

Peter


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Re: [Asterisk-Users] gxp-2000 tftp cfg

[Kristof Hardy]

The VoIP Connection wrote:

It's here:
http://www.thevoipconnection.com/Downloads/GXP2000_1.0.1.9/gxp2000_config.txt


Very interesting, it wasn't available at Grandstream's site :-)
Thanks! I will adjust some things on the page now I have the new 
template.. http://voip-info.org/tiki-index.php?page=GXP-2000


Cheers
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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 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 Firmware 1.0.1.2

[Lee Archer]

Hi, do you have an on-site NTP server?  I found that 
after the firmware update NTP from the * server stopped 
working.
 
Regards
 
Lee


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Jesus 
MogollonSent: 24 August 2005 22:11To: 
asterisk-users@lists.digium.comSubject: [Asterisk-Users] GXP 2000 
Firmware 1.0.1.2
Greetings all   Grandstream released a new firmware 
and it seems like the speaker phone problem has been fixed. However we updated 
to firmware 1.0.1.12 to fix the echo problem but found other 
problems were now created. The worst 
of these new problems is that the whole phone starts degrading, the volume starts 
getting lower and lower. The ringing 
starts fading and the calls start stuttering. The only way this can 
be fixed is by rebooting the phone. 
We were able to replicate this problem 
in all phones while some Polycoms we have do not suffer from this problem. Again, this problem happened 
AFTER we upgraded to the new 
firmware.   Has anyone seen this?Jesus 
MogollonGlobal IP Systems, LLChttp://www.globalipsystems.com/###This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange.For more information, connect to http://www.f-secure.com/
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Re: [Asterisk-Users] GXP 2000 Firmware 1.0.1.2

[Jesus Mogollon]

Hi Lee:

  NTP is working as expected, but it does take a couple of minutes (!) to get the date from the server


Jesus Mogollon
2005/8/25, Lee Archer <[EMAIL PROTECTED]>:





Hi, do you have an on-site NTP server?  I found that 
after the firmware update NTP from the * server stopped 
working.
 
Regards
 
Lee


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED]] On Behalf Of Jesus 
MogollonSent: 24 August 2005 22:11To: 
asterisk-users@lists.digium.comSubject:  [Asterisk-Users] 
GXP 2000 
Firmware 1.0.1.2
Greetings all   Grandstream released a new firmware 
and it seems like the speaker phone problem has been fixed. However we updated 
to firmware 1.0.1.12 to fix the echo problem but found other 
problems were now created. The worst 
of these new problems is that the whole phone starts degrading, the volume starts 
getting lower and lower. The ringing 
starts fading and the calls start stuttering. The only way this can 
be fixed is by rebooting the phone. 
We were able to replicate this problem 
in all phones while some Polycoms we have do not suffer from this problem. Again, this problem happened 
AFTER we upgraded to the new 
firmware.   Has anyone seen this?Jesus 
MogollonGlobal IP Systems, LLChttp://www.globalipsystems.com/###
This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange.For more information, connect to 
http://www.f-secure.com/

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RE: [Asterisk-Users] GXP 2000 Firmware 1.0.1.2

[Lee Archer]

Well it's only worked once and I've left the phones several 
hours.  I've done various debugs and the phone is asking for NTP and the 
server is answering but its not getting set.
 
Lee


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Jesus 
MogollonSent: 25 August 2005 12:54To: Asterisk Users 
Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] 
GXP 2000 Firmware 1.0.1.2
Hi Lee:  NTP is working as expected, but it does take a 
couple of minutes (!) to get the date from the serverJesus 
Mogollon
2005/8/25, Lee Archer <[EMAIL PROTECTED]>:

  Hi, do you 
  have an on-site NTP server?  I found that after the firmware update NTP 
  from the * server stopped working.
   
  Regards
   
  Lee
  
  
  From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of 
  Jesus MogollonSent: 24 August 2005 22:11To: asterisk-users@lists.digium.comSubject: 
  [Asterisk-Users] GXP 2000 Firmware 1.0.1.2
  
  Greetings all   Grandstream released a new 
  firmware and it seems like the speaker phone problem has been fixed. However 
  we updated to firmware 1.0.1.12 to fix the echo problem but found other problems 
  were now created. The worst of 
  these new problems is that the whole phone starts degrading, the volume starts 
  getting lower and lower. The ringing 
  starts fading and the calls start stuttering. The only way this can 
  be fixed is by rebooting the 
  phone. We were able to replicate this problem in all phones while some Polycoms we 
  have do not suffer from this 
  problem. Again, this problem happened AFTER we upgraded to the new firmware.   Has anyone 
  seen this?Jesus MogollonGlobal IP Systems, LLChttp://www.globalipsystems.com/### 
  This message has been scanned by F-Secure Anti-Virus for Microsoft 
  Exchange.For more information, connect to http://www.f-secure.com/ 
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  UNSUBSCRIBE or update options visit:  http://lists.digium.com/mailman/listinfo/asterisk-users###This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange.For more information, connect to http://www.f-secure.com/
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RE: [Asterisk-Users] GXP 2000 Firmware 1.0.1.2

[Lee Archer]

I can time sync with time.nist.gov but not with any 
internal servers.  I read in the changelog about them fixing something 
related to NTP on the same subnet but it doesn't say whether it should work or 
shouldn't.
 
Regards
 
Lee


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Jesus 
MogollonSent: 25 August 2005 12:54To: Asterisk Users 
Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] 
GXP 2000 Firmware 1.0.1.2
Hi Lee:  NTP is working as expected, but it does take a 
couple of minutes (!) to get the date from the serverJesus 
Mogollon
2005/8/25, Lee Archer <[EMAIL PROTECTED]>:

  Hi, do you 
  have an on-site NTP server?  I found that after the firmware update NTP 
  from the * server stopped working.
   
  Regards
   
  Lee
  
  
  From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of 
  Jesus MogollonSent: 24 August 2005 22:11To: asterisk-users@lists.digium.comSubject: 
  [Asterisk-Users] GXP 2000 Firmware 1.0.1.2
  
  Greetings all   Grandstream released a new 
  firmware and it seems like the speaker phone problem has been fixed. However 
  we updated to firmware 1.0.1.12 to fix the echo problem but found other problems 
  were now created. The worst of 
  these new problems is that the whole phone starts degrading, the volume starts 
  getting lower and lower. The ringing 
  starts fading and the calls start stuttering. The only way this can 
  be fixed is by rebooting the 
  phone. We were able to replicate this problem in all phones while some Polycoms we 
  have do not suffer from this 
  problem. Again, this problem happened AFTER we upgraded to the new firmware.   Has anyone 
  seen this?Jesus MogollonGlobal IP Systems, LLChttp://www.globalipsystems.com/### 
  This message has been scanned by F-Secure Anti-Virus for Microsoft 
  Exchange.For more information, connect to http://www.f-secure.com/ 
  ___--Bandwidth and 
  Colocation sponsored by Easynews.com 
  --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo 
  UNSUBSCRIBE or update options visit:  http://lists.digium.com/mailman/listinfo/asterisk-users###This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange.For more information, connect to http://www.f-secure.com/
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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 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

[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

[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

[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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> >>http://lists.digium.com/mailman/listinfo/asterisk-users
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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 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 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 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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> > 
> > 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 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-
> > > 
> > > ___
> > > --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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> > 
> > 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 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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> > > > 
> > > > Asterisk-Users mailing list
> > > > To UNSUBSCRIBE or update options visit:
> > > >http://lists.digium.com/mailman/listinfo/asterisk-users
> > > 
> > > ___
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> > > 
> > > Asterisk-Users mailing list
> > > To UNSUBSCRIBE or update options visit:
> > >http://lists.digium.com/mailman/listinfo/asterisk-users
> > > 
> > 
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> > 
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> >http://lists.digium.com/mailman/listinfo/asterisk-users
> > 
> 
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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 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

[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

[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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[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 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

[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 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 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 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

[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]

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 any good with * ?

[trixter aka Bret McDanel]

On Fri, 2005-12-30 at 15:01 -0800, [EMAIL PROTECTED] wrote:
> Anyone using the GXP-2000 with * ?
> 
> Any showstopper problems?
> 
> The echo issues, is it speakerphone only?
> 
I have a gxp and dont have the echo issue.  I got it from
www.thevoipconnection.com and its a good phone I think anyway.  No
manual was included but the PDF is quickly accessable via
www.grandstream.com (click on the gxp2000 image on the main page then
its on the right side).

I think the echo problem was with older firmware versions.  It takes
about 5 seconds to set up, so its not that hard.  I dont use it as a
router or anything though, but that didnt look terribly difficult to set
up, infact at most  that should only add an additional 5 seconds to
configuration.

People have said that it doesnt do call forwarding hwoever according to
the manual and testing it does (*72 ...)

Software Version:Program-- 1.0.1.9Bootloader-- 1.0.1.2
-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378
http://www.sacaug.org/ Sacramento Asterisk Users Group


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

[Michiel van Baak]

On 15:01, Fri 30 Dec 05, [EMAIL PROTECTED] wrote:
> Anyone using the GXP-2000 with * ?
> 
> Any showstopper problems?
> 
> The echo issues, is it speakerphone only?
> 

Hi,

We have some of these phones in production.
They work ok.

The echo issues are fixed in recent fw versions.

The speeddial buttons come with a led, but hinting is not
working. And I'm used to cisco call quality and then the GXP
just is lowbudget. The phone is good for an office that
doesn't focus on phonecalls. Our programmers use them cause
they don't make that many calls.

Overall they are not bad but don't expect a real business
phone for your money.
-- 
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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Re: [Asterisk-Users] GXP-2000 any good with * ?

[Tom Vile]

Hinting works fine for me with the latest firmware.
On 12/30/05, Michiel van Baak <[EMAIL PROTECTED]> wrote:
> On 15:01, Fri 30 Dec 05, [EMAIL PROTECTED] wrote:
> > Anyone using the GXP-2000 with * ?
> >
> > Any showstopper problems?
> >
> > The echo issues, is it speakerphone only?
> >
>
> Hi,
>
> We have some of these phones in production.
> They work ok.
>
> The echo issues are fixed in recent fw versions.
>
> The speeddial buttons come with a led, but hinting is not
> working. And I'm used to cisco call quality and then the GXP
> just is lowbudget. The phone is good for an office that
> doesn't focus on phonecalls. Our programmers use them cause
> they don't make that many calls.
>
> Overall they are not bad but don't expect a real business
> phone for your money.
> --
> 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 19:35, Fri 30 Dec 05, Tom Vile 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
blink when one of the speeddials is ringing/in a call.
The same hint stuff in the dailplan works great with snom
and cisco.

Please enlighten me.
-- 
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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RE: [Asterisk-Users] GXP-2000 any good with * ?

[Ross C]

I like them.  IMO, you can't beat them for the money.  My first SIP phone
order was a bunch of GXP-2000's, some Polycom 601's, and 2 Snom 320's (now
discontinued I think).  The Polycom has the best feel/quality, but it was
hundreds more.
The speakerphone on the GXP-2000 leaves a bit to be desired. And they also
need to work on getting the volume louder (from both the speakerphone and
the handset) without it getting distorted or causing echo or blowing up.
I've had trouble getting call forwarding to work reliably.  I can enable
call forwarding, but when I try to disable it, it doesn't work right; when I
dial *73 or whatever it is to disable the call forwarding, after I press the
"3", I get a dial tone again (when I should get a response from the Asterisk
server telling me that call forwarding has been disabled).  Maybe there's a
fix for thisI dunno.
Another problem I have sometimes is the phone locking up on reboot.  This
kinda puts the hurt on the whole 'remote management' concept because you're
forced to go power cycle the phone when it locks up.  Since I've had all of
mine setup, I've had to do this a large handful of times; not enough to
justify throwing them out the window, but enough to be very annoying.

Ultra simple to configure.  Looking at the PDF manual didn't even cross my
mind when I started configuring them.  The default password is "admin";
that's about the only non-obvious piece of information when doing the basic
configuration.  The built in switch doesn't require any configuration to
use; just connect a computer to the 'PC' port on the back of the phone.

FYI--I'm using 1.0.1.13 for the firmware version (beta, but works well--at
least better than other versions)

All in all, I'd recommend them if you're on a budget.  At <$100, they're a
good deal IMHO.  Lots of features and it seems like they're actively working
on improving the firmware and featuresat least some.


-ross

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Friday, December 30, 2005 5:02 PM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] GXP-2000 any good with * ?

Anyone using the GXP-2000 with * ?

Any showstopper problems?

The echo issues, is it speakerphone only?

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

[Ken D'Ambrosio]

> Anyone using the GXP-2000 with * ?

Yup.  I like the phone -- works nicely, easy to configure, four lines, etc.

> Any showstopper problems?

Well... see below.

> The echo issues, is it speakerphone only?

The speakerphone kinda sucks.  Maybe someone with a newer firmware or
different config options has found differently.  I was all ready to make
the GXP-2000 my "default" desk phone, but nixed it after trying to use the
speakerphone.  IMHO, it's not so much echo as it is feedback -- the
speaker seems to be awfully close to the microphone.  [Note: I may be dead
wrong, but that's my hunch.]  Since I don't have one working right now, I
can't comment further...

-Ken


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

[Tom Vile]

I am using the .12 firmware

How are you using hint with the cisco phones?

On 12/30/05, Ross C <[EMAIL PROTECTED]> wrote:
> I like them.  IMO, you can't beat them for the money.  My first SIP phone
> order was a bunch of GXP-2000's, some Polycom 601's, and 2 Snom 320's (now
> discontinued I think).  The Polycom has the best feel/quality, but it was
> hundreds more.
> The speakerphone on the GXP-2000 leaves a bit to be desired. And they also
> need to work on getting the volume louder (from both the speakerphone and
> the handset) without it getting distorted or causing echo or blowing up.
> I've had trouble getting call forwarding to work reliably.  I can enable
> call forwarding, but when I try to disable it, it doesn't work right; when I
> dial *73 or whatever it is to disable the call forwarding, after I press the
> "3", I get a dial tone again (when I should get a response from the Asterisk
> server telling me that call forwarding has been disabled).  Maybe there's a
> fix for thisI dunno.
> Another problem I have sometimes is the phone locking up on reboot.  This
> kinda puts the hurt on the whole 'remote management' concept because you're
> forced to go power cycle the phone when it locks up.  Since I've had all of
> mine setup, I've had to do this a large handful of times; not enough to
> justify throwing them out the window, but enough to be very annoying.
>
> Ultra simple to configure.  Looking at the PDF manual didn't even cross my
> mind when I started configuring them.  The default password is "admin";
> that's about the only non-obvious piece of information when doing the basic
> configuration.  The built in switch doesn't require any configuration to
> use; just connect a computer to the 'PC' port on the back of the phone.
>
> FYI--I'm using 1.0.1.13 for the firmware version (beta, but works well--at
> least better than other versions)
>
> All in all, I'd recommend them if you're on a budget.  At <$100, they're a
> good deal IMHO.  Lots of features and it seems like they're actively working
> on improving the firmware and featuresat least some.
>
>
> -ross
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> [EMAIL PROTECTED]
> Sent: Friday, December 30, 2005 5:02 PM
> To: Asterisk-Users@lists.digium.com
> Subject: [Asterisk-Users] GXP-2000 any good with * ?
>
> Anyone using the GXP-2000 with * ?
>
> Any showstopper problems?
>
> The echo issues, is it speakerphone only?
>
> -Dan
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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 * ?

[Bjorn Asmul]

You have to disable the local call features to let Asterisk deal with
them.
Otherwise the phone will try to accommodate all those features.

It's important to update to the latest firmware to get the best use.
Using .13 works well with all features. If you're on speakerphone you
should not crank it up to the loudest volume.

Overall the phone beats any other phone in the same price category, and
you're not bound to stupid licensing/software limitations as the others.
It should also be said that Grandstream is working closely with
Asterisk. That can NOT be said about other, and more expensive, phones.
Last time I spoke to Polycom they didn't want to help me when I told
them I was connecting the phone to Asterisk. They tend to support only
(expensive) commercial PBX's.

Bjorn 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ross C
Sent: Saturday, December 31, 2005 1:57 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] GXP-2000 any good with * ?

I like them.  IMO, you can't beat them for the money.  My first SIP
phone order was a bunch of GXP-2000's, some Polycom 601's, and 2 Snom
320's (now discontinued I think).  The Polycom has the best
feel/quality, but it was hundreds more.
The speakerphone on the GXP-2000 leaves a bit to be desired. And they
also need to work on getting the volume louder (from both the
speakerphone and the handset) without it getting distorted or causing
echo or blowing up.
I've had trouble getting call forwarding to work reliably.  I can enable
call forwarding, but when I try to disable it, it doesn't work right;
when I dial *73 or whatever it is to disable the call forwarding, after
I press the "3", I get a dial tone again (when I should get a response
from the Asterisk server telling me that call forwarding has been
disabled).  Maybe there's a fix for thisI dunno.
Another problem I have sometimes is the phone locking up on reboot.
This kinda puts the hurt on the whole 'remote management' concept
because you're forced to go power cycle the phone when it locks up.
Since I've had all of mine setup, I've had to do this a large handful of
times; not enough to justify throwing them out the window, but enough to
be very annoying.

Ultra simple to configure.  Looking at the PDF manual didn't even cross
my mind when I started configuring them.  The default password is
"admin"; that's about the only non-obvious piece of information when
doing the basic configuration.  The built in switch doesn't require any
configuration to use; just connect a computer to the 'PC' port on the
back of the phone.

FYI--I'm using 1.0.1.13 for the firmware version (beta, but works
well--at least better than other versions)

All in all, I'd recommend them if you're on a budget.  At <$100, they're
a good deal IMHO.  Lots of features and it seems like they're actively
working on improving the firmware and featuresat least some.


-ross

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Friday, December 30, 2005 5:02 PM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] GXP-2000 any good with * ?

Anyone using the GXP-2000 with * ?

Any showstopper problems?

The echo issues, is it speakerphone only?

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

[Michiel van Baak]

On 20:02, Fri 30 Dec 05, Tom Vile wrote:
> I am using the .12 firmware
> 
> How are you using hint with the cisco phones?

In my dialplan I have (the default stuff):
exten => 6000,hint,SCCP/6000
exten => 6000,1,Macro(stdexten,mainvoicemail,SCCP/6000)

Then on the cisco side I did setup a speeddial for SCCP/6000

I think I need to mention that I use chan_sccp.so for all my
cisco phones. It's so much better then the SIP image.

You can see some image of the speeddial/hint stuff here:
http://lunteren.vanbaak.info/~michiel/voor_alex/
I made those images for a 7960 SIP setup at a partners
place. The images are taken of my home 7960.
The livingroom speeddial you see there is a 7905 in my
livingroom ;)
-- 
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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RE: [Asterisk-Users] GXP-2000 any good with * ?

[Ross C]

Do I disable the local call features in the web admin of the phone?  If so,
how?

Thanks  :)

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bjorn Asmul
Sent: Friday, December 30, 2005 7:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] GXP-2000 any good with * ?

You have to disable the local call features to let Asterisk deal with
them.
Otherwise the phone will try to accommodate all those features.

It's important to update to the latest firmware to get the best use.
Using .13 works well with all features. If you're on speakerphone you
should not crank it up to the loudest volume.

Overall the phone beats any other phone in the same price category, and
you're not bound to stupid licensing/software limitations as the others.
It should also be said that Grandstream is working closely with
Asterisk. That can NOT be said about other, and more expensive, phones.
Last time I spoke to Polycom they didn't want to help me when I told
them I was connecting the phone to Asterisk. They tend to support only
(expensive) commercial PBX's.

Bjorn 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ross C
Sent: Saturday, December 31, 2005 1:57 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] GXP-2000 any good with * ?

I like them.  IMO, you can't beat them for the money.  My first SIP
phone order was a bunch of GXP-2000's, some Polycom 601's, and 2 Snom
320's (now discontinued I think).  The Polycom has the best
feel/quality, but it was hundreds more.
The speakerphone on the GXP-2000 leaves a bit to be desired. And they
also need to work on getting the volume louder (from both the
speakerphone and the handset) without it getting distorted or causing
echo or blowing up.
I've had trouble getting call forwarding to work reliably.  I can enable
call forwarding, but when I try to disable it, it doesn't work right;
when I dial *73 or whatever it is to disable the call forwarding, after
I press the "3", I get a dial tone again (when I should get a response
from the Asterisk server telling me that call forwarding has been
disabled).  Maybe there's a fix for thisI dunno.
Another problem I have sometimes is the phone locking up on reboot.
This kinda puts the hurt on the whole 'remote management' concept
because you're forced to go power cycle the phone when it locks up.
Since I've had all of mine setup, I've had to do this a large handful of
times; not enough to justify throwing them out the window, but enough to
be very annoying.

Ultra simple to configure.  Looking at the PDF manual didn't even cross
my mind when I started configuring them.  The default password is
"admin"; that's about the only non-obvious piece of information when
doing the basic configuration.  The built in switch doesn't require any
configuration to use; just connect a computer to the 'PC' port on the
back of the phone.

FYI--I'm using 1.0.1.13 for the firmware version (beta, but works
well--at least better than other versions)

All in all, I'd recommend them if you're on a budget.  At <$100, they're
a good deal IMHO.  Lots of features and it seems like they're actively
working on improving the firmware and featuresat least some.


-ross

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Friday, December 30, 2005 5:02 PM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] GXP-2000 any good with * ?

Anyone using the GXP-2000 with * ?

Any showstopper problems?

The echo issues, is it speakerphone only?

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

[Tom Vile]

I see,  I am not using skinny but maybe I should give it a try.  Why
is it better than the SIP image?

On 12/30/05, Michiel van Baak <[EMAIL PROTECTED]> wrote:
> On 20:02, Fri 30 Dec 05, Tom Vile wrote:
> > I am using the .12 firmware
> >
> > How are you using hint with the cisco phones?
>
> In my dialplan I have (the default stuff):
> exten => 6000,hint,SCCP/6000
> exten => 6000,1,Macro(stdexten,mainvoicemail,SCCP/6000)
>
> Then on the cisco side I did setup a speeddial for SCCP/6000
>
> I think I need to mention that I use chan_sccp.so for all my
> cisco phones. It's so much better then the SIP image.
>
> You can see some image of the speeddial/hint stuff here:
> http://lunteren.vanbaak.info/~michiel/voor_alex/
> I made those images for a 7960 SIP setup at a partners
> place. The images are taken of my home 7960.
> The livingroom speeddial you see there is a 7905 in my
> livingroom ;)
> --
> 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?"
>
> ___
> --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
>


--
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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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?"
>
> ___
> --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
>


--
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 * ?

[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 * ?

[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 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)]

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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-- 
---
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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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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-- 
---
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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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
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> --
> --
> -
> 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
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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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> 
> --
> --
> -
> 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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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 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 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 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

[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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