[asterisk-users] Which SIP phones to buy?

[Stephen Bosch]

I need to buy some new phones for our own offices.

I've used only Polycom phones until now, but I'd like to broaden my
experience.

I'm trying to decide which phones to experiment with. I have these options:

- A combination of Polycom, Aastra and Snom

- Just Polycom

One the one hand, I'd like to keep things uniform, since it greatly
simplifies provisioning. On the other hand, I don't want to broaden my
knowledge.

Advice, anyone?

-Stephen-

[Stephen Bosch]

Post by Stephen Bosch
I need to buy some new phones for our own offices.
I've used only Polycom phones until now, but I'd like to broaden my
experience.
- A combination of Polycom, Aastra and Snom
- Just Polycom
One the one hand, I'd like to keep things uniform, since it greatly
simplifies provisioning. On the other hand, I don't want to broaden my
knowledge...

...because I like to stay dumb.

Of course, that's not what I meant :)

-Stephen-

[Ronaldo Zacarias Afonso]

Hi Stephen,

I'm using Grandstream and I think is a nice phone, but its the only
one that I've tried.
I bought it to learn voip/asterisk.

Just my 2 cents.
Good luck.

Ronaldo.

Post by Stephen Bosch

Post by Stephen Bosch
I need to buy some new phones for our own offices.
I've used only Polycom phones until now, but I'd like to broaden my
experience.
- A combination of Polycom, Aastra and Snom
- Just Polycom
One the one hand, I'd like to keep things uniform, since it greatly
simplifies provisioning. On the other hand, I don't want to broaden my
knowledge...

...because I like to stay dumb.
Of course, that's not what I meant :)
-Stephen-
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asterisk-users mailing list
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[Drew Gibson]

Post by Stephen Bosch

Post by Stephen Bosch
I need to buy some new phones for our own offices.
I've used only Polycom phones until now, but I'd like to broaden my
experience.
- A combination of Polycom, Aastra and Snom
- Just Polycom
One the one hand, I'd like to keep things uniform, since it greatly
simplifies provisioning. On the other hand, I don't want to broaden my
knowledge...

...because I like to stay dumb.
Of course, that's not what I meant :)

We have Cisco, Aastra 480i and Grandstream GXP2000 phones in house.

I only recommend the Cisco phones to people I don't like, overpriced and
far too much work.

The Aastra 480i is a good quality phone, on par with Cisco and probably
with Polycom (though I've never used them). Voice quality is good, phone
feels robust. Config is well documented and contained in two text files
(one global, one MAC specific). Good web interface on the phone. Aastra
support have been very responsive.

Grandstream phones are lower quality but good value for money. Sound and
feel of phones is not so good as Aastra or Cisco. Configuration is
through a binary file, a bit fiddly, but quite manageable with a few
scripts. Good web interface on the phone. Grandstream support have also
been very responsive.

regards,

Drew

[Stephen Bosch]

Post by Drew Gibson
We have Cisco, Aastra 480i and Grandstream GXP2000 phones in house.
I only recommend the Cisco phones to people I don't like, overpriced and
far too much work.
The Aastra 480i is a good quality phone, on par with Cisco and probably
with Polycom (though I've never used them). Voice quality is good, phone
feels robust. Config is well documented and contained in two text files
(one global, one MAC specific). Good web interface on the phone. Aastra
support have been very responsive.
Grandstream phones are lower quality but good value for money. Sound and
feel of phones is not so good as Aastra or Cisco. Configuration is
through a binary file, a bit fiddly, but quite manageable with a few
scripts. Good web interface on the phone. Grandstream support have also
been very responsive.

Thanks for the comments. I think I might give one or two Aastra sets a
try, just for tire-kicking.

Cheers,

-Stephen-

[Alex Balashov]

The Cisco phones are quite good. The thing that most people don't tend to
appreciate about them is that they all are designed essentially for
mass-provisioning in large environments, and to operate with Call Manager.
Provisioning them using their GUI/configuration interface on a one-off
basis is a pain in the butt, this is absolutely true. They were never
really intended to be used in that manner. If you can take the time to
do the TFTP thing, though, they really are very wonderful, featureful,
reliable, comfortable, and, I would go so far as to say, turn-key.

Just my $.02. No interest in religious debate.

-- Alex

--
Alex Balashov <***@presidium.org>

[David Cook]

Post by Stephen Bosch
I'm trying to decide which phones to experiment with. I have these
- A combination of Polycom, Aastra and Snom
- Just Polycom
One the one hand, I'd like to keep things uniform, since it greatly
simplifies provisioning. On the other hand, I don't want to broaden
my
knowledge.
Advice, anyone?
-Stephen-

You said 'office' so I'm presuming you want business quality. If you
have already tried the Polycom's I'd look at Aastra (just did a 50+
seat implementation with 9133i's & 480i's) and also look at the Cisco
79xx's.

Cisco's & Aastra's both handle multiple appearances differently but both
are excellent. Cisco has superb handsfree quality. Aastra has better BLF
support. You will have to evaluate for yourself. Aastra is significantly
cheaper. That said, there is a 7960 on my desk that isn't going anywhere
soon.

I hear the Grandstream firmware is better now but physically they are
still pretty flimsy. I would stay away from them for anything but
experimentation.

dbc.

[Robert Greene]

Post by Drew Gibson

Post by Stephen Bosch

Post by Stephen Bosch
I need to buy some new phones for our own offices.
I've used only Polycom phones until now, but I'd like to broaden my
experience.
- A combination of Polycom, Aastra and Snom
- Just Polycom
One the one hand, I'd like to keep things uniform, since it greatly
simplifies provisioning. On the other hand, I don't want to broaden my
knowledge...

...because I like to stay dumb.
Of course, that's not what I meant :)

We have Cisco, Aastra 480i and Grandstream GXP2000 phones in house.
I only recommend the Cisco phones to people I don't like, overpriced and
far too much work.
The Aastra 480i is a good quality phone, on par with Cisco and probably
with Polycom (though I've never used them). Voice quality is good, phone
feels robust. Config is well documented and contained in two text files
(one global, one MAC specific). Good web interface on the phone. Aastra
support have been very responsive.
Grandstream phones are lower quality but good value for money. Sound and
feel of phones is not so good as Aastra or Cisco. Configuration is
through a binary file, a bit fiddly, but quite manageable with a few
scripts. Good web interface on the phone. Grandstream support have also
been very responsive.
regards,
Drew
--
Drew Gibson
Systems Administrator
OANDA Corporation
www.oanda.com

I have Cisco 7960G, Polycom SP 501, Aastra 480i, Grandstream GXP2000 and
Budgetone 200 desk phones in my test lab. Overall, I like the Cisco
best. I even bought one for home use. Configuration was no more
difficult than any other.

The Cisco, Aastra and Polycom have similar voice quality. They're all
very good handsets and speakerphones. Of these three, the Aastra is the
only backlit display, but it is hard to read from an angle and the
backlight is not very effective. Aastra is also very vulnerable to
glare. The Cisco and Polycom are easier to read unless you are in a
darkened room. The Grandstream GXP2000 and Budgetone 200 have nice,
bright and easy to read displays, but the phone aesthetics are not up to
par with the others.

For daily use, the Cisco and Polycom buttons are smoothest. The Aastra
is close, but not as comfortable to use. It seems that round buttons
function better. The Grandstream buttons are just heavy and cumbersome.

The Polycom is the biggest pain in the ass to initially configure
because of the extended boot time. All other brands I've used boot
within a minute and are ready to use. The Polycom takes around 4 and if
you are using the web interface for initial configuration, you need to
reboot frequently. Once you've worked out your configuration, new phone
installs are pretty simple with any brand.

The Aastra and Grandstream web interfaces are easy to use and you may
make multiple changes and then reboot when you're done. The Cisco has
no web interface.

For routine provisioning, Cisco only supports tftp and telnet. The
Polycom supports tftp, ftp, sftp, http and https. The Aastra supports
tftp, ftp & http.

Placing a logo on the Cisco display is trivial. I have not been
successful with any other brand so far.

For PoE use, the Polycom and Aastra use 802.3af. Up to the 7970, Cisco
used a proprietary PoE pin configuration and require a special cable to
use with a standards compliant PoE switch. The cable is easy to make,
but you have to ensure that users are aware of the difference.

As for price, Drew is right about the high cost of Cisco. If I hadn't
found one on eBay, my personal phone would likely be Aastra.

[J. Oquendo]

Post by Robert Greene

Post by Drew Gibson
The Aastra 480i is a good quality phone, on par with Cisco and probably
with Polycom (though I've never used them). Voice quality is good, phone
feels robust. Config is well documented and contained in two text files
(one global, one MAC specific). Good web interface on the phone. Aastra
support have been very responsive.

I've programmed who knows how many phones so far, so let me focus on this
year... Cisco ... Overrated. I have one on my desk right now specifically
being used for me to test security on it (writing an exploit against it
to be specific). I don't even bother using it...

Aastra 480i's. I can take a picture of my desk for verification for those
who'd want it of two I have sitting here collecting dust. They're horribly
documented. Their web interface is full of errors (Username/CallerID/Auth)
of which unless you're used to doing it you will have issues programming
these.

Polycoms... The bane of my existence. If you plan on doing NAT, passing
through Netscreens, Sonicwalls, etc., and you don't mind miserably wasting
time, then these are for you! If you're a glutton for XML nonsense, waiting
2 minutes for a reboot after EVERY SINGLE CHANGE. This phone is for you! If
you don't mind explaining the Americans with Disabilities ACT and how
Polycom is the only vendor resetting volumes then this is for you! And yes
I am aware I could make that static via xml so please don't bother with a
"but you can fix that this way..." response.

Snom, although not the greatest, within the past year I've had to deal with
well over I would guesstimate 200 or so. Easiest to deal with.

Grandstream... Sorry, there is only so much garbage I'm willing to keep
around my desk. GXP 2000? Fisher Price toy looking phone I wouldn't
bother with.

Post by Robert Greene
I have Cisco 7960G, Polycom SP 501, Aastra 480i, Grandstream GXP2000 and
Budgetone 200 desk phones in my test lab. Overall, I like the Cisco
best. I even bought one for home use. Configuration was no more
difficult than any other.

This is what is within two feet of me right now. 2 Cisco 7960's, 3
Polycruds,
2 Aasta 480i's, Welltech piece of garbage, 1 Snom 360, 2 320's, unlimited
190's. Guess which one I used on a daily basis... Snom 360.

@Home ... Loading Image... I have about 3 7960's
for my CCVP lab studies... I have a Hitachi WiFi and a Snom 320. Guess
which I use most... Hitachi so I could walk around, followed by Snom. I
don't even want to bother with the Cisco phones.

Post by Robert Greene
The Cisco, Aastra and Polycom have similar voice quality. They're all
very good handsets and speakerphones.

The 480iCT is questionable. The base is alright, nothing to boast about,
the handset... Depends on the environment.

Post by Robert Greene
As for price, Drew is right about the high cost of Cisco. If I hadn't
found one on eBay, my personal phone would likely be Aastra.

I could care less about pricing. I'd be more concerned with
quality and ease of use for both the admin, and the user.
Cisco would rank low on my list, so here goes mine in order...

1) Snom
2) none! (they're all pretty much the same to me)
3) none! (they all have their pros and cons)
4) Cisco
5) ASStra
6) Polycrud

[C F]

Post by J. Oquendo

Post by Robert Greene

Post by Drew Gibson
The Aastra 480i is a good quality phone, on par with Cisco and probably
with Polycom (though I've never used them). Voice quality is good,

phone

Post by Robert Greene

Post by Drew Gibson
feels robust. Config is well documented and contained in two text files
(one global, one MAC specific). Good web interface on the phone. Aastra
support have been very responsive.

I've programmed who knows how many phones so far, so let me focus on this
year... Cisco ... Overrated. I have one on my desk right now specifically
being used for me to test security on it (writing an exploit against it
to be specific). I don't even bother using it...
Aastra 480i's. I can take a picture of my desk for verification for those
who'd want it of two I have sitting here collecting dust. They're horribly
documented. Their web interface is full of errors (Username/CallerID/Auth)
of which unless you're used to doing it you will have issues programming
these.
Polycoms... The bane of my existence. If you plan on doing NAT, passing
through Netscreens, Sonicwalls, etc., and you don't mind miserably wasting
time, then these are for you! If you're a glutton for XML nonsense, waiting
2 minutes for a reboot after EVERY SINGLE CHANGE. This phone is for you! If
you don't mind explaining the Americans with Disabilities ACT and how
Polycom is the only vendor resetting volumes then this is for you! And yes
I am aware I could make that static via xml so please don't bother with a
"but you can fix that this way..." response.

J, Sorry didn't see this email when I wrote the other one (gmail sorts
them on a LIFO order). I can agree with you on everything even with
the terrible pain of getting Polycoms up and running, but once it is
up dont you have less problems with them then with other phones? Isn't
the sound quality of the Polycoms better than any of the other phones?

Post by J. Oquendo
Snom, although not the greatest, within the past year I've had to deal with
well over I would guesstimate 200 or so. Easiest to deal with.

I did not have that good of an experience with Snoms. I guess I should
try again, since it's well over 18 months since I tried last.

Post by J. Oquendo
Grandstream... Sorry, there is only so much garbage I'm willing to keep
around my desk. GXP 2000? Fisher Price toy looking phone I wouldn't
bother with.

Couldn't agree more with you on this.

Post by J. Oquendo

Post by Robert Greene
I have Cisco 7960G, Polycom SP 501, Aastra 480i, Grandstream GXP2000 and
Budgetone 200 desk phones in my test lab. Overall, I like the Cisco
best. I even bought one for home use. Configuration was no more
difficult than any other.

This is what is within two feet of me right now. 2 Cisco 7960's, 3
Polycruds,
2 Aasta 480i's, Welltech piece of garbage, 1 Snom 360, 2 320's, unlimited
190's. Guess which one I used on a daily basis... Snom 360.
@Home ... http://www.infiltrated.net/Mar2520074.jpg I have about 3 7960's
for my CCVP lab studies... I have a Hitachi WiFi and a Snom 320. Guess
which I use most... Hitachi so I could walk around, followed by Snom. I
don't even want to bother with the Cisco phones.

Awesome photo, arn't you having too much fun working?

Post by J. Oquendo

Post by Robert Greene
The Cisco, Aastra and Polycom have similar voice quality. They're all
very good handsets and speakerphones.

The 480iCT is questionable. The base is alright, nothing to boast about,
the handset... Depends on the environment.

Post by Robert Greene
As for price, Drew is right about the high cost of Cisco. If I hadn't
found one on eBay, my personal phone would likely be Aastra.

I could care less about pricing. I'd be more concerned with
quality and ease of use for both the admin, and the user.
Cisco would rank low on my list, so here goes mine in order...

Again I think the Polycom once configure right is quite easy for both
the admin and the user.

Post by J. Oquendo
1) Snom
2) none! (they're all pretty much the same to me)
3) none! (they all have their pros and cons)
4) Cisco
5) ASStra
6) Polycrud
--
====================================================
J. Oquendo
http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x1383A743
The happiness of society is the end of government.
John Adams
_______________________________________________
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asterisk-users mailing list
http://lists.digium.com/mailman/listinfo/asterisk-users

[J. Oquendo]

Post by C F
J, Sorry didn't see this email when I wrote the other one (gmail sorts
them on a LIFO order). I can agree with you on everything even with
the terrible pain of getting Polycoms up and running, but once it is
up dont you have less problems with them then with other phones? Isn't
the sound quality of the Polycoms better than any of the other phones?

Depends on the network sometimes. For clients with anything less than
a dedicated-to-VoIP-T1 I would have to disagree. If you do face this
situation (someone with low bandwidth), Snom's rock.

Post by C F
I did not have that good of an experience with Snoms. I guess I should
try again, since it's well over 18 months since I tried last.

I stated "They aren't the best..." but of the whole lot of phones I deal
with,
they've been thusfar the least problematic.

Post by C F
Awesome photo, arn't you having too much fun working?

Nah ;) that's like a fraction of junk I play with. At work I have a
CC(IE/VP) lab too.
2 3620's 2501, 2522, 3 4500M's, LS1010, Merge ISDN simulator, Pix, Cat
3500's, Netscouts... :D

Post by C F
Again I think the Polycom once configure right is quite easy for both
the admin and the user.

Well, two things come into play so I should have mentioned it. Its best
to get
a complete picture of what the end user would expect. Once you set those
options in XML, unless you're setting up a tftpboot server and can change
it, you're hit. I've had far too many instances where clients have
ordered them
and wanted cosmetic changes that could only be done via the xml files. But
what happens when those phones are not booting via tftp. I'm stuck. I either
have to have them send me back the phone to make the changes, re-do
one and send it back out, or maybe on rare occasions walk someone through
having their phone boot via tftp to one my me servers to make those changes.

Now ponder this for a minute... Executive John calls me: "Can you make
this change for me" ... I respond "Sure can you open up your firewall
for me,
I will also need you to press x button and enter the following..." Even with
some so called "certified" engineers, that becomes cumbersome.

Most of the times if they have their own PBX (I work for a company that
does managed PBX's and sells PBX's), and we administrate it, I will set
up a squid proxy with only my IP space allowed via ACL's and firewall
rules, so I could throw on a proxy on my browser and do it.

[C F]

Post by J. Oquendo

Post by C F
J, Sorry didn't see this email when I wrote the other one (gmail sorts
them on a LIFO order). I can agree with you on everything even with
the terrible pain of getting Polycoms up and running, but once it is
up dont you have less problems with them then with other phones? Isn't
the sound quality of the Polycoms better than any of the other phones?

Depends on the network sometimes. For clients with anything less than
a dedicated-to-VoIP-T1 I would have to disagree. If you do face this
situation (someone with low bandwidth), Snom's rock.

Post by C F
I did not have that good of an experience with Snoms. I guess I should
try again, since it's well over 18 months since I tried last.

I stated "They aren't the best..." but of the whole lot of phones I deal
with,
they've been thusfar the least problematic.

Post by C F
Awesome photo, arn't you having too much fun working?

Nah ;) that's like a fraction of junk I play with. At work I have a
CC(IE/VP) lab too.
2 3620's 2501, 2522, 3 4500M's, LS1010, Merge ISDN simulator, Pix, Cat
3500's, Netscouts... :D

Post by C F
Again I think the Polycom once configure right is quite easy for both
the admin and the user.

Well, two things come into play so I should have mentioned it. Its best
to get
a complete picture of what the end user would expect. Once you set those
options in XML, unless you're setting up a tftpboot server and can change
it, you're hit. I've had far too many instances where clients have
ordered them
and wanted cosmetic changes that could only be done via the xml files. But
what happens when those phones are not booting via tftp. I'm stuck. I either
have to have them send me back the phone to make the changes, re-do
one and send it back out, or maybe on rare occasions walk someone through
having their phone boot via tftp to one my me servers to make those changes.
Now ponder this for a minute... Executive John calls me: "Can you make
this change for me" ... I respond "Sure can you open up your firewall
for me,
I will also need you to press x button and enter the following..." Even with
some so called "certified" engineers, that becomes cumbersome.

This is one point that I have to agree with you, I dread the phone
calls that users call me they want just a simple change on a Polycom
specific to them. However using FTP, it's only a big deal because of
the XML (which also means that I have to document the change, since
there is NO way for me to know otherwise that it has a minor change
compared to the rest of the users), but it should work nicely remotely
as well. All I do before deploying a Polycom phone to a remote site
(which is quite easy to walk someone thru it over the phone) is set
the FTP Server address, username, and password. Which requires just
opening FTP on the server side firewall. That means for security
reasons I can't leave it that way, but I could open it up when the
user needs a change and have them reboot the phone.

Still this is my phone of choice, althoug for the price they should
have had much more features when it comes to remapping buttons, or
PoE.

I must say I have never run into a situation where I had low
bandwidth, I always make sure there is at least 768k up, with a less
than 150ms latency (not always have been able to meet the later, but
never more than 250ms), so can't realy comment on this one.

You are pushing me to test that snom again. Will try it.

Post by J. Oquendo
Most of the times if they have their own PBX (I work for a company that
does managed PBX's and sells PBX's), and we administrate it, I will set
up a squid proxy with only my IP space allowed via ACL's and firewall
rules, so I could throw on a proxy on my browser and do it.
--
====================================================
J. Oquendo
http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x1383A743
The happiness of society is the end of government.
John Adams
_______________________________________________
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asterisk-users mailing list
http://lists.digium.com/mailman/listinfo/asterisk-users

[Bruno De Luca]

I think that the best choice is the snom family...

We use all snom in ower office. We tried the Polycom but the support is
not so good.

Bruno.

Post by C F

Post by J. Oquendo

Post by C F
J, Sorry didn't see this email when I wrote the other one (gmail sorts
them on a LIFO order). I can agree with you on everything even with
the terrible pain of getting Polycoms up and running, but once it is
up dont you have less problems with them then with other phones? Isn't
the sound quality of the Polycoms better than any of the other phones?

Depends on the network sometimes. For clients with anything less than
a dedicated-to-VoIP-T1 I would have to disagree. If you do face this
situation (someone with low bandwidth), Snom's rock.

Post by C F
I did not have that good of an experience with Snoms. I guess I should
try again, since it's well over 18 months since I tried last.

I stated "They aren't the best..." but of the whole lot of phones I deal
with,
they've been thusfar the least problematic.

Post by C F
Awesome photo, arn't you having too much fun working?

Nah ;) that's like a fraction of junk I play with. At work I have a
CC(IE/VP) lab too.
2 3620's 2501, 2522, 3 4500M's, LS1010, Merge ISDN simulator, Pix, Cat
3500's, Netscouts... :D

Post by C F
Again I think the Polycom once configure right is quite easy for both
the admin and the user.

Well, two things come into play so I should have mentioned it. Its best
to get
a complete picture of what the end user would expect. Once you set those
options in XML, unless you're setting up a tftpboot server and can change
it, you're hit. I've had far too many instances where clients have
ordered them
and wanted cosmetic changes that could only be done via the xml files. But
what happens when those phones are not booting via tftp. I'm stuck. I either
have to have them send me back the phone to make the changes, re-do
one and send it back out, or maybe on rare occasions walk someone through
having their phone boot via tftp to one my me servers to make those changes.
Now ponder this for a minute... Executive John calls me: "Can you make
this change for me" ... I respond "Sure can you open up your firewall
for me,
I will also need you to press x button and enter the following..." Even with
some so called "certified" engineers, that becomes cumbersome.

This is one point that I have to agree with you, I dread the phone
calls that users call me they want just a simple change on a Polycom
specific to them. However using FTP, it's only a big deal because of
the XML (which also means that I have to document the change, since
there is NO way for me to know otherwise that it has a minor change
compared to the rest of the users), but it should work nicely remotely
as well. All I do before deploying a Polycom phone to a remote site
(which is quite easy to walk someone thru it over the phone) is set
the FTP Server address, username, and password. Which requires just
opening FTP on the server side firewall. That means for security
reasons I can't leave it that way, but I could open it up when the
user needs a change and have them reboot the phone.
Still this is my phone of choice, althoug for the price they should
have had much more features when it comes to remapping buttons, or
PoE.
I must say I have never run into a situation where I had low
bandwidth, I always make sure there is at least 768k up, with a less
than 150ms latency (not always have been able to meet the later, but
never more than 250ms), so can't realy comment on this one.
You are pushing me to test that snom again. Will try it.

Post by J. Oquendo
Most of the times if they have their own PBX (I work for a company that
does managed PBX's and sells PBX's), and we administrate it, I will set
up a squid proxy with only my IP space allowed via ACL's and firewall
rules, so I could throw on a proxy on my browser and do it.
--
====================================================
J. Oquendo
http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x1383A743
The happiness of society is the end of government.
John Adams
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
http://lists.digium.com/mailman/listinfo/asterisk-users

_______________________________________________
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asterisk-users mailing list
http://lists.digium.com/mailman/listinfo/asterisk-users

[J. Oquendo]

Post by C F
I must say I have never run into a situation where I had low
bandwidth, I always make sure there is at least 768k up, with a less
than 150ms latency (not always have been able to meet the later, but
never more than 250ms), so can't realy comment on this one.

*Ducks the items thrown after this one...*

Don't ask why... For some reason I ended up with a client on a channelized
T w/768 allocated for VoIP. They had an insane packet loss (sometimes 60%)
which some level1 noc monkey told me was normal before I asked for his
supervisor.... They averaged (not kidding) about 170ms and had about 40
Snoms... Why? I asked myself that daily...

On notes of ftp... I wouldn't mind, its almost always the other side
that gets all teary eyed "You want me to open a hole in my firewall!"
(said the recent MCSE shlup)... ;)

[Luca Corti]

Post by J. Oquendo
1) Snom
2) none! (they're all pretty much the same to me)
3) none! (they all have their pros and cons)
4) Cisco
5) ASStra
6) Polycrud

You haven't even mentioned Linksys SPAs. Have you tested them?

ciao

Luca

[Ed Nuñez]

In my asterisk, I have calls coming in on a Zap channel and going out SIP.
My problem is that when I spy on the SIP channel, I hear the calling parting
breaking in and out, and the called party sounds just fine (SIP). If I spy
on the Zap channel , I hear both sides just fine. I am spying from my SIP
extension.

Any ideas?

[J. Oquendo]

Post by Luca Corti

Post by J. Oquendo
1) Snom
2) none! (they're all pretty much the same to me)
3) none! (they all have their pros and cons)
4) Cisco
5) ASStra
6) Polycrud

You haven't even mentioned Linksys SPAs. Have you tested them?
ciao
Luca

We have them in our demo room. No one wants them claiming the too
look cheap. I've programmed like 2 in the last who knows how long
and have no desire to add more stuff to my desk. I've read others'
compliments on their use, but Linksys as a whole... Let's say I
have an opinion on them. So no mentions of them due to the fact
they're not popular with my clients. As for a breakdown... Right
now my clients at least 90% of them are SoHo businesses with
anywhere between 10 and 400 users. I would estimate most PBX's
I build at about 75 clients. They never even pick up Linksys
phones when visiting us. I also have Welltech phones I did not
mention for the same reason. I don't believe there is a phone
on the market right now that hasn't made it to my desk to be
quite honest.

If we went by majority rules... This is what I've been putting
out based on quantity:

Snom
Polycom
Aastra
Cisco
Welltech

With Aastra making a huge gain this year. Last year, people didn't
bother with them. This year when the 480iCT came out, we've had
more orders for them. Snom's have always been ordered over Poly's
with customers even preferring lowly 190's over Poly's.

[map]

Linksys SPAs work well with Asterisk

Post by Luca Corti

Post by J. Oquendo
1) Snom
2) none! (they're all pretty much the same to me)
3) none! (they all have their pros and cons)
4) Cisco
5) ASStra
6) Polycrud

You haven't even mentioned Linksys SPAs. Have you tested them?
ciao
Luca
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
http://lists.digium.com/mailman/listinfo/asterisk-users

[Luca Corti]

Post by map
Linksys SPAs work well with Asterisk

I know, I use them and besides some initial nasty bugs and occasional
quirks they are quite nice. I also think they are not so ugly.

ciao

Luca

[Per Jessen]

Post by Luca Corti

Post by map
Linksys SPAs work well with Asterisk

I know, I use them and besides some initial nasty bugs and occasional
quirks they are quite nice. I also think they are not so ugly.

Luca, what sort of nasty bugs and quirks have you seen with the Linksys
SPA? We've recently started using a few SPA-921, and will probably be
buying some more.


/Per Jessen, Zürich

[Mike Lynchfield]

[pedro noticioso]

Hi there list!

I want to catch all numbers that don't exist, play a
nice message and restart operator, this is different
from dial i because that is for incorrect extensions,
an undefined number will give a busy signal, something
I don't like

You can search for the word irc to see my comments,
the line above is my latest unsuccessful test, thanks!



; #### #### #### #### #### #### #### #### ####
#### #### ####
;
;
;
;
; begin extensions
;
;
;
;
; #### #### #### #### #### #### #### #### ####
#### #### ####
;

[general] ;

language=es
; autofallthrough=yes
clearglobalvars=no

[globals]

; Definiendo variables para usarlas a traves de todo
el
; MINOMBRE=mailinator.net
; MITELEFONOFXO=55555555
; OPERADORA=



;
; Si static esta en no, u omitido, entonces pbx_config
va a sobreescribir
; a este archivo cuando se cambien las extensiones.
Recuerda que todos los
; comentarios de este archivo desapareceran si pasa
eso.
;
; XXX Todavia no ha sido implementado XXX
;
static=yes
;
;
; si stati=yes y writeprotect=no, tambien puedes
guardar al dialplan con
; linea de comandos ejecutando 'save dialplan' y
borrando estos comentarios
;
writeprotect=yes

CONSOLE=Zap/1 ; pendiente entender *
TRUNK=Zap/1 ; Trunk interface *
TRUNKMSD=1 ; MSD digits to strip (usually
1 or 0) *



; #### #### #### #### #### #### #### #### ####
#### #### ####
; Trunks

;[context] ;exten =>
someexten,priority[+offset][(alias)],application(arg1,arg2,...)

[trunkint] ; International long distance
through trunk
exten => _9001.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})

[trunkld] ; Long distance context
accessed through trunk
exten =>
_901ZXXXXXXXXX,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})

[trunklocal] ; Local eight-digit dialing
accessed through trunk interface
exten =>
_9ZXXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) ;
llamada local comun y corriente
exten => _90ZXS0,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
; 020, etc
exten => _9066,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) ;
066, etc

[trunktollfree] ; Long distance context
accessed through trunk interface
exten =>
_901800NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})

[trunkpaypercall] ; Dangerous pay-per call!
exten =>
_901900.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})

[trunkcelular] ; Long distance context
accessed through trunk interface
exten =>
_9044ZZXXXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten =>
_9045ZZXXXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})



; #### #### #### #### #### #### #### #### ####
#### #### ####
; Contexts
[international] ; Master context for
international long distance
ignorepat => 9
include => longdistance
include => trunkint

[longdistance] ; Master context for long
distance
ignorepat => 9
include => local
include => trunkld
include => trunktollfree
include => trunkpaypercall

[mercadotecnia]
ignorepat => 9
include => local

[local] ; Master context for local,
toll-free, and iaxtel calls only
ignorepat => 9
include => default
include => parkedcalls
include => trunklocal


[record]
exten => s,1,Answer
exten => s,2,Read(RECORD|enter4digits|4)
exten => s,3,Playback(record-instructions)
exten =>
s,4,Record(/var/lib/asterisk/sounds/recording/s-${RECORD}|wav)
exten => s,5,Wait(2)
exten =>
s,6,Playback(/var/lib/asterisk/sounds/recording/s-${RECORD})
exten => s,7,ResponseTimeout(10)
exten =>
s,8,Background(1toaccept2torerecord3torecordanother)
exten => 1,1,Hangup
exten => 2,1,Goto(s,3)
exten => 3,1,Goto(s,2)


[macro-stdexten];
;
; Macro de extensiones estandard:
; ${ARG1} - Extension (Pudimos haver usado
${MACRO_EXTEN} tambien aqui
; ${ARG2} - Aparato(s) a marcar
;
exten => s,1,Dial(${ARG2},20) ; Ring the interface,
20 seconds maximum
exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on
status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten => s-NOANSWER,1,Voicemail(${ARG1},u) ; If
unavailable, send to voicemail w/ unavail announce
exten => s-NOANSWER,2,Goto(default,s,1) ; If they
press #, return to start
exten => s-BUSY,1,Voicemail(${ARG1},b) ; If busy,
send to voicemail w/ busy announce
exten => s-BUSY,2,Goto(default,s,1) ; If they press
#, return to start
exten => _s-.,1,Goto(s-NOANSWER,1) ; Treat anything
else as no answer
exten => a,1,VoicemailMain(${ARG1}) ; If they press
*, send the user into VoicemailMain





[macro-stdexten-viejo] ; Standard extension macro:
; ARG1 es el numero de la extension
; ARG2 es sip al cual voy a marcar
exten => s,1,Dial(${ARG2},20,rt) ; Ring the interface,
20 seconds maximum
exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on
status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten => s-NOANSWER,1,Voicemail(u${ARG1}) ; If
unavailable, send to voicemail w/ unavail announce
exten => s-NOANSWER,2,Goto(default,s,1) ; If
they press #, return to start
exten => s-BUSY,1,Voicemail(b${ARG1}) ; If
busy, send to voicemail w/ busy announce
exten => s-BUSY,2,Goto(default,s,1) ; If
they press #, return to start
exten => _s-.,1,Goto(s-NOANSWER,1) ;
Treat anything else as no answer
exten => a,1,VoicemailMain(${ARG1}) ; If
they press *, send the user into VoicemailMain



; #### #### #### #### #### #### #### #### ####
#### #### ####
; Dial in

[default]
exten => s,1,Set(CHANNEL(language)=es)

exten => s,2,Set(TIMEOUT(digit)=5) ; Set Digit Timeout
to 5 seconds
exten => s,n,Set(TIMEOUT(response)=10) ; Set Response
Timeout to 10 seconds

exten => s,n,Answer ; Answer the
line
exten => s,n(restart),BackGround(enter-ext-of-person)
; Play the intoduction message
exten => s,n,WaitExten ; Wait for an extension to
be dialed.

; Si no marcan una extension, termina el tiepo de
timeout
; y marca a la operadora
exten => t,1,Macro(stdexten,603,SIP/sip603)

; Si marcan cero en cualquier momento, contesta la
operadora
exten => 0,1,Macro(stdexten,603,SIP/sip603)


exten => i,1,Playback(pbx-invalid) exten =>
i,n,Goto(s,restart,2)

exten => s,1,Set(CHANNEL(language)=es)


exten => 600,1,Macro(stdexten,600,SIP/sip600)
exten => 601,1,Macro(stdexten,601,SIP/sip601)
exten => 602,1,Macro(stdexten,602,SIP/sip602)
exten => 603,1,Macro(stdexten,603,SIP/sip603)
exten => 604,1,Macro(stdexten,604,SIP/sip604)
exten => 605,1,Macro(stdexten,605,SIP/sip605)

exten => 610,1,Macro(stdexten,610,SIP/sip610)
exten => 650,1,Macro(stdexten,650,SIP/sip650)
exten => 651,1,Macro(stdexten,651,SIP/sip651)
exten => 652,1,Macro(stdexten,652,SIP/sip652)
exten => 653,1,Macro(stdexten,653,SIP/sip653)

exten => 8500,1,VoicemailMain

; Para hacer prubas, ahora voy a llamarle al inicio de
las
; llamadas que entran por el FXS, aqui me esta
respondiendo
; Asterisk como si le estuviera llamando por telefono
desde
; otra linea normal

exten => 700,1,Goto(s,1)
exten => operadora,1,Goto(s,1)

exten => 8200,1,Goto(record,s,1)
exten => 8010,1,MusicOnHold(default)
;
; Esto va al final para no tapar a las anteriores
;


; reinicia el ciclo cuando se equivocan de extension
je
; el numero pelon


exten => _[1-580],1,Playback(pbx-invalid)exten =>
_1,n,Goto(s,restart,2)

;
;
; irc all this crap works on some extensions
;
;
;
;
;

;exten => _1,1,Playback(pbx-invalid)exten =>
_1,n,Goto(s,restart,2)
;exten => _2,1,Playback(pbx-invalid)exten =>
_2,n,Goto(s,restart,2)
;exten => _3,1,Playback(pbx-invalid)exten =>
_3,n,Goto(s,restart,2)
;exten => _4,1,Playback(pbx-invalid)exten =>
_4,n,Goto(s,restart,2)
;exten => _5,1,Playback(pbx-invalid)exten =>
_5,n,Goto(s,restart,2)
;exten => _7,1,Playback(pbx-invalid)exten =>
_7,n,Goto(s,restart,2)
;exten => _8,1,Playback(pbx-invalid)exten =>
_8,Goto(s,restart,2)
;exten => _0,1,Playback(pbx-invalid)exten =>
_0,n,Goto(s,restart,2)

;iniciando con y cualquier cosa mas
;exten => _1X.,1,Playback(pbx-invalid)exten =>
_1.,n,Goto(s,restart,2)
;exten => _2X.,1,Playback(pbx-invalid)exten =>
_2.,n,Goto(s,restart,2)
;exten => _3X.,1,Playback(pbx-invalid)exten =>
_3.,n,Goto(s,restart,2)
;exten => _4X.,1,Playback(pbx-invalid)exten =>
_4.,n,Goto(s,restart,2)
;exten => _5X.,1,Playback(pbx-invalid)exten =>
_5.,n,Goto(s,restart,2)
;exten => _7X.,1,Playback(pbx-invalid)exten =>
_7.,n,Goto(s,restart,2)
;exten => _8X.,1,Playback(pbx-invalid)exten =>
_8.,n,Goto(s,restart,2)
;exten => _0X.,1,Playback(pbx-invalid)exten =>
_0.,n,Goto(s,restart,2)




; Gracias por llamar, marque su extension 0 para que
le atienda la operadora
; busy zap?
;
; #### #### #### #### #### #### #### #### ####
#### #### ####
;
;
;
;
; begin features
;
;
;
;
; #### #### #### #### #### #### #### #### ####
#### #### ####
;
; Sample Call Features (parking, transfer, etc)
configuration
;

[general]
; parkext => 700 ; What extension to dial to park
; parkpos => 701-720 ; What extensions to park calls
on. These needs to be
; numeric, as Asterisk starts from the start
position
; and increments with one for the next parked
call.
; context => parkedcalls ; Which context parked calls
are in
;parkingtime => 45 ; Number of seconds a call can be
parked for
; (default is 45 seconds)
;courtesytone = beep ; Sound file to play to the
parked caller
; when someone dials a parked call
; or the Touch Monitor is activated/deactivated.
;parkedplay = caller ; Who to play the courtesy tone
to when picking up a parked call
; one of: parked, caller, both (default is
caller)
;adsipark = yes ; if you want ADSI parking
announcements
;findslot => next ; Continue to the 'next' free
parking space.
; Defaults to 'first' available
;parkedmusicclass=default ; This is the MOH class to
use for the parked channel
; as long as the class is not set on the channel
directly
; using Set(CHANNEL(musicclass)=whatever) in the
dialplan

;transferdigittimeout => 3 ; Number of seconds to wait
between digits when transferring a call
; (default is 3 seconds)
;xfersound = beep ; to indicate an attended transfer
is complete
;xferfailsound = beeperr ; to indicate a failed
transfer
;pickupexten = *8 ; Configure the pickup extension.
(default is *8)
;featuredigittimeout = 500 ; Max time (ms) between
digits for
; feature activation (default is 500 ms)
;atxfernoanswertimeout = 15 ; Timeout for answer on
attended transfer default is 15 seconds.


[featuremap]
;blindxfer => #1 ; Blind transfer (default is #)
;disconnect => *0 ; Disconnect (default is *)
;automon => *1 ; One Touch Record a.k.a. Touch
Monitor
;atxfer => *2 ; Attended transfer
;parkcall => #72 ; Park call (one step
parking)











atxfer => #

blindxfer => #






[applicationmap]
; Note that the DYNAMIC_FEATURES channel variable must
be set to use the features
; defined here. The value of DYNAMIC_FEATURES should
be the names of the features
; to allow the channel to use separated by '#'. For
example:
;
;
Set(DYNAMIC_FEATURES=myfeature1#myfeature2#myfeature3)
;
;
; The syntax for declaring a dynamic feature is the
following:
;
;<FeatureName> =>
<DTMF_sequence>,<ActivateOn>[/<ActivatedBy>],<Application>[,<AppArguments>[,MOH_Class]]
;
; FeatureName -> This is the name of the feature
used in when setting the
; DYNAMIC_FEATURES variable to
enable usage of this feature.
; DTMF_sequence -> This is the key sequence used to
activate this feature.
; ActivateOn -> This is the channel of the call
that the application will be executed
; on. Valid values are "self" and
"peer". "self" means run the
; application on the same channel
that activated the feature. "peer"
; means run the application on the
opposite channel from the one that
; has activated the feature.
; ActivatedBy -> This is which channel is allowed
to activate this feature. Valid
; values are "caller", "callee", and
"both". "both" is the default.
; The "caller" is the channel that
executed the Dial application, while
; the "callee" is the channel called
by the Dial application.
; Application -> This is the application to
execute.
; AppArguments -> These are the arguments to be
passed into the application.
; MOH_Class -> This is the music on hold class to
play while the idle
; channel waits for the feature to
complete. If left blank,
; no music will be played.
;
;
; IMPORTANT NOTE: The applicationmap is not intended
to be used for all Asterisk
; applications. When applications are used in
extensions.conf, they are executed
; by the PBX core. In this case, these applications
are executed outside of the
; PBX core, so it does *not* make sense to use any
application which has any
; concept of dialplan flow. Examples of this would
be things like Macro, Goto,
; Background, WaitExten, and many more.
;
; Enabling these features means that the PBX needs to
stay in the media flow and
; media will not be re-directed if DTMF is sent in the
media stream.
;
; Example Usage:
;
;testfeature => #9,peer,Playback,tt-monkeys ;Allow
both the caller and callee to play
;
;tt-monkeys to the opposite channel
;
;pauseMonitor => #1,self/callee,Pausemonitor
;Allow the callee to pause monitoring
; ;on
their channel
;unpauseMonitor => #3,self/callee,UnPauseMonitor
;Allow the callee to unpause monitoring
; ;on
their channel
;


;
;
;
;
; begin modules
;
;
;
;
;
; Asterisk configuration file
;
; Module Loader configuration file
;

[modules]
autoload=yes
;
; Any modules that need to be loaded before the
Asterisk core has been initialized
; (just after the logger has been initialized) can be
loaded using 'preload'. This
; will frequently be needed if you wish to map all
module configuration files into
; Realtime storage, since the Realtime driver will
need to be loaded before the
; modules using those configuration files are
initialized.
;
; An example of loading ODBC support would be:
;preload => res_odbc.so
;preload => res_config_odbc.so
;
; If you want, load the GTK console right away.
; Don't load the KDE console since
; it's not as sophisticated right now.
;
noload => pbx_gtkconsole.so
;load => pbx_gtkconsole.so
noload => pbx_kdeconsole.so
;
; Intercom application is obsoleted by
; chan_oss. Don't load it.
;
noload => app_intercom.so
;
; Explicitly load the chan_modem.so early on to be
sure
; it loads before any of the chan_modem_* 's afte rit
;
load => chan_modem.so
load => res_musiconhold.so
;
; Load either OSS or ALSA, not both
; By default, load OSS only (automatically) and do not
load ALSA
;
noload => chan_alsa.so
noload => chan_oss.so
;
; Module names listed in "global" section will have
symbols globally
; exported to modules loaded after them.
;
[global]
chan_modem.so=yes
;
;
;
;
; begin sip
;
;
;
;
; Contexto general
[general]
port = 5060 ; Puerto en el que empezamos
bindaddr = 0.0.0.0 ; dirección o direcciones ip
0.0.0.0 = todas
context=local ; Contexto default para todos
tos=lowdelay
dtmfmode=rfc2833 ; info ; Tonos DTMF
disallow=all ; Deshabilita todos los codecs
allow=ulaw ; Permite el codec ulaw (g711)
10kb/s
allow=ilbc ; Permite el codec ilbc 3kb/s
allow=gsm ; Permite el codec gsm 3kb/s
allow=g729 ; Permite el codec g729 2.5kb/s
(propietario)

; Hacemos login en FWD (registrando) para recibir
llamadas a nuestro numero y enviarlas
; A la extensión 21

; FWD number 77443 pointing to extension 21
; register => 77443:***@fwd.pulver.com/21

; Para poder sacar llamadas por FWD
; FWD account
;[fwd.pulver.com]
; type=peer
; host=fwd.pulver.com
; fromuser=77443
; fromdomain=fwd.pulver.com
; username=77443
; secret=miclave
; dtmfmode=rfc2833

; Extension 600
[sip600]
type=friend
secret=ext600
context=international
callerid="Nombre Apellido" <600>
host=dynamic
reinvite=no
canreinvite=yes
dtmfmode=info
transfer=yes
nat=no
disallow=all
allow=ulaw
allow=alaw
allow=g729
mailbox=***@default,600

; Extension 601
[sip601]
type=friend
secret=ext601
context=international
callerid="Nombre Apellido" <601>
host=dynamic
reinvite=no
canreinvite=yes
dtmfmode=info
transfer=yes
nat=no
disallow=all
allow=ulaw
allow=alaw
allow=g729
mailbox=***@default,601

; Extension 602
[sip602]
type=friend
secret=ext602
context=international
callerid="Nombre correo" <602>
host=dynamic
reinvite=no
canreinvite=yes
dtmfmode=info
transfer=yes
nat=no
disallow=all
allow=ulaw
allow=alaw
allow=g729
mailbox=***@default,602


; Extension 603
[sip603]
type=friend
secret=ext603
context=international
callerid="Nombre correo" <603>
host=dynamic
reinvite=no
canreinvite=yes
dtmfmode=info
transfer=yes
nat=no
disallow=all
allow=ulaw
allow=alaw
allow=g729
mailbox=***@default,603

; Extension 604
[sip604]
type=friend
secret=ext604
context=international
callerid="Nombre correo" <604>
host=dynamic
reinvite=no
canreinvite=yes
dtmfmode=info
transfer=yes
nat=no
disallow=all
allow=ulaw
allow=alaw
allow=g729
mailbox=***@default,604

; Extension 605
[sip605]
type=friend
secret=ext605
context=international
callerid="Nombre correo" <605>
host=dynamic
reinvite=no
canreinvite=yes
dtmfmode=info
transfer=yes
nat=no
disallow=all
allow=ulaw
allow=alaw
allow=g729
mailbox=***@default,605


[sip610]
type=friend
context=international
callerid="Nombre Apellido" <610>
username=sip610
secret=ext610
nat=no
canreinvite=yes
dtmfmode=info
mailbox=***@default
disallow=all
allow=ulaw
allow=alaw
allow=g729

[sip650]
type=friend
callerid="XLite Apellido remote" <650>
host=dynamic ; This device needs to
register
username=sip650
secret=ext650
nat=yes ; X-Lite is behind a
NAT router
canreinvite=no ; Typically set to NO
if behind NAT
disallow=all
allow=gsm ; GSM consumes far less
bandwidth than ulaw
allow=ulaw
allow=alaw



[sip651]
type=friend
callerid="SoftPhone de Nombre Apellido" <651>
host=dynamic ; This device needs to
register
username=sip651
secret=ext651
nat=yes ; X-Lite is behind a
NAT router
canreinvite=no ; Typically set to NO
if behind NAT
disallow=all
allow=gsm ; GSM consumes far less
bandwidth than ulaw
allow=ulaw




[sip652]
type=friend
callerid="SoftPhone de Nombre correo Cabrera" <652>
host=dynamic ; This device needs to
register
username=sip652
secret=ext652
nat=yes ; X-Lite is behind a
NAT router
canreinvite=no ; Typically set to NO
if behind NAT
disallow=all
allow=gsm ; GSM consumes far less
bandwidth than ulaw
allow=ulaw
allow=alaw


[sip653]
type=friend
callerid="xlite de Nombre correo Cabrera" <653>
host=dynamic ; This device needs to
register
username=sip653
secret=ext653
nat=yes ; X-Lite is behind a
NAT router
canreinvite=no ; Typically set to NO
if behind NAT
disallow=all
allow=gsm ; GSM consumes far less
bandwidth than ulaw
allow=ulaw
allow=alaw

;
;
;
;
; begin voicemail
;
;
;
;
;
; Voicemail Configuration
;
[general]
; Default formats for writing Voicemail

format=wav ;format=g723sf|wav49|wav
serveremail=***@mailinator.net ; Who the e-mail
notification should appear to come from
attach=no ; Should the email contain the voicemail
as an attachment
;maxmessage=180 ; Maximum length of a voicemail
message in seconds
;minmessage=3 ; Minimum length of a voicemail
message in seconds
;maxgreet=60 ; Maximum length of greetings in
seconds
skipms=3000 ; How many miliseconds to skip
forward/back when rew/ff in message playback
maxsilence=2 ; How many seconds of silence before we
end the recording
silencethreshold=30 ; Silence threshold (what we
consider silence, the lower, the more sensitive)
maxlogins=3 ; Max number of failed login attempts
languaje=es

; If you need to have an external program, i.e.
/usr/bin/myapp
; called when a voicemail is left, delivered, or your
voicemailbox
; is checked, uncomment this:
;externnotify=/usr/bin/myapp
; If you need to have an external program, i.e.
/usr/bin/myapp
; called when a voicemail password is changed,
; uncomment this:
;externpass=/usr/bin/myapp

;directoryintro=dir-intro ; For the directory, you can
override the intro file if you want
;charset=ISO-8859-1 ; The character set for voicemail
messages can be specified here
;adsifdn=0000000F ; The ADSI feature descriptor
number to download to
;adsisec=9BDBF7AC ; The ADSI security lock code
;adsiver=1 ; The ADSI voicemail application version
number.
;pbxskip=yes ; Skip the "[PBX]:" string from the
message title
;fromstring=The Asterisk PBX ; Change the From: string
;usedirectory=yes ; Permit finding entries for
forward/compose from the directory

;pagerfromstring=The Asterisk PBX ;Change the From:
string for pager messages

; Change the email body and/or subject, variables:
; VM_NAME, VM_DUR, VM_MSGNUM, VM_MAILBOX,
VM_CALLERID, VM_CIDNUM, VM_CIDNAME, VM_DATE
;
; Note: The emailbody config row can be up to 512
characters due to a limitation in
; asterisk config files.
;emailsubject=[PBX]: New message ${VM_MSGNUM} in
mailbox ${VM_MAILBOX}
; The following definition is very close to the
default, but the default shows just
; the CIDNAME, if it is not null, else just the
CIDNUM, or "an unknown caller" if they are both null.
;emailbody=Dear ${VM_NAME}:\n\n\tjust wanted to let
you know you were just left a ${VM_DUR} long message
(number ${VM_MSGNUM})\nin mailbox ${VM_MAILBOX} from
${VM_CALLERID}, on ${VM_DATE}, so you might\nwant to
check it when you get a chance.
Thanks!\n\n\t\t\t\t--Asterisk\n


; You can override the default program to send e-mail
if you wish, too
;mailcmd=/usr/sbin/sendmail -t


; Advanced options example is extension 4069
; NOTE: All options can be expressed globally in the
general section, and overriden in the per-mailbox
; settings, unless listed otherwise.
;
; tz=central ; Timezone from zonemessages
above. Irrelevant if envelope=no.
attach=yes ; Attach the voicemail to the
notification email *NOT* the pager email
;saycid=yes ; Say the caller id information
before the message. If not described,
; or set to no, it will be
in the envelope
; cidinternalcontexts=intern ; Internal Context for
Name Playback instead of extension digits when saying
caller id.
; sayduration=no ; Turn on/off the duration
information before the message. [ON by default]
; saydurationm=2 ; Specify the minimum duration
to say. Default is 2 minutes
; dialout=fromvm ; Context to dial out from
[option 4 from the advanced menu]
; if not listed, dialing
out will not be permitted
sendvoicemail=yes ; Context to Send voicemail
from [option 5 from the advanced menu]
; if not listed, sending
messages from inside voicemail will not be
; permitted
; callback=fromvm ; Context to call back from
; if not listed, calling
the sender back will not be permitted
; review=yes ; Allow sender to
review/rerecord their message before saving it [OFF by
default
operator=yes ; Allow sender to hit 0
before/after/during leaving a voicemail to
; reach an operator [OFF
by default]
; envelope=no ; Turn on/off envelope
playback before message playback. [ON by default]
; This does NOT affect
option 3,3 from the advanced options menu
; delete=yes ; After notification, the
voicemail is deleted from the server. [per-mailbox
only]
; This is intended for use
with users who wish to receive their voicemail ONLY by
email.
; nextaftercmd=yes ; Skips to the next message
after hitting 7 or 9 to delete/save current message.
; [global option only at
this time]
; forcename=yes ; Forces a new user to record
their name. A new user is
; determined by the
password being the same as
; the mailbox number. The
default is "no".
; forcegreetings=no ; This is the same as
forcename, except for recording
; greetings. The default
is "no".
; hidefromdir=yes ; Hide this mailbox from the
directory produced by app_directory
; The default is "no".

[zonemessages]
eastern=America/New_York|'vm-received' Q 'digits/at'
IMp
central=America/Chicago|'vm-received' Q 'digits/at'
IMp
central24=America/Chicago|'vm-received' q 'digits/at'
H 'digits/hundred' M 'hours'

[default]
600 => 600,Nombre
Apellido,***@mailinator.com,,operator=yes|attach=yes
601 => 601,Nombre
Apellido,***@mailinator.com,,operator=yes|attach=yes
602 => 602,Nombre
correo,***@mailinator.net,,operator=yes|attach=yes
603 => 603,Nombre
Apellido,***@mailinator.com,,operator=yes|attach=yes
604 => 604,cuarto
sip,***@sip.com,,operator=yes|attach=yes
605 => 605,quinto
sip,***@sip.com,,operator=yes|attach=yes



610 => 610,Nombre
Apellido,***@mailinator.com,,operator=yes|attach=yes
650 => 650,Nombre Apellido
remote,***@mailinator.com,,operator=yes|attach=yes

650 => 650,Nombre correo
remote,***@mailinator.net,,operator=yes|attach=yes




;4200 => 9855,Mark
Spencer,***@linux-support.net,***@digium.com,attach=no|serveremail=***@digium.com|tz=central
;4300 => 3456,Ben Rigas,***@american-computer.net
;4310 => -5432,Sales,***@marko.net
;4069 => 6522,Matt
Brooks,***@marko.net,,|tz=central|attach=yes|saycid=yes|dialout=fromvm|callback=fromvm|review=yes|operator=yes|envelope=yes|sayduration=yes|saydurationm=1
;4073 => 1099,Bianca
Paige,***@biancapaige.com,,delete=1
;4110 => 3443,Rob Flynn,***@blueridge.net

;
;
;
;
; begin zapata
;
;
;
;

[channels]

busydetect=no
busycount=5
relaxdtmf=yes
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
usecallerid=yes
hidecallerid=no
echotraining=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
immediate=no
context=default
signalling=fxs_ks
callerid=asreceived
callprogress=no
musiconhold=yes
channel => 1-2




____________________________________________________________________________________
Food fight? Enjoy some healthy debate
in the Yahoo! Answers Food & Drink Q&A.
http://answers.yahoo.com/dir/?link=list&sid=396545367

[Ken Morley]

I've had experience with quite a few different phones, so I think I'm
qualified to drop my two cents:

Alex is quite right that the Cisco phones are only designed to be used
with Cisco Call Manager. They are capable of being decent SIP
telephones, but Cisco won't provide the documentation so that you can
use them effectively with anything other than Cisco Call Manager, so
that's the deal killer. Like everything else Cisco, they're also
ridiculously expensive.

Despite what Alex says, the Cisco SIP phones have plenty of fundamental
flaws. I have a number of expensive 7970G phones with a beautiful color
display. Each of the various SIP firmware versions available for that
product has a serious flaw. The most acceptable version is about a year
old. It's biggest flaw is that the Message Waiting Indicator doesn't
work. Most of the other SIP firmware versions won't register with
Asterisk. If you are planning to usee Asterisk, save your money and
your sanity and buy something else.

In my last project, I used the Aastra 480i phones. Yes, the
documentation is lacking, but that's largely because the platform was
evolving quickly. Aastra has excellent and responsive technical support
via e-mail. Finally, the customer was very satisfied with the quality
and the price of the 480i phones.

In my latest project, I used the newer Aastra 57i and 57i CT phones. It
is obvious that these phones derived from the 480i software, but they
are much faster and more full-featured with great displays, etc. The
initial documentation with these is fairly good and complete. I have
them doing all kinds of things, including using the XML capabilities to
push server applications to the display, update the softkeys in
real-time, etc.

As contrasted against Cisco, Aastra even provides PHP include files to
greatly simplify web development on whatever platform (Asterisk,
Sylantro, etc.) you are using. The 57i phones are a little expensive,
but they are a top-notch product that works very well with Asterisk
right out of the box. Plus, they look and sound great and have 12
softkeys that shift to 20.

One of the others that responded to your question mentioned something
about setting up a TFTP server and I want to elaborate on that a little.
If you are deploying more than a small handful of phones, you will want
to setup a TFTP server anyway. It would be muy loco to try deploying and
supporting a few dozen phones otherwise. Many of the phone's features
aren't even accessible through the web interface anyway - you have to
have a TFTP server and make use of the configuration files for full
functionality. And that applies to Aastra, Cisco, Polycom or whatever.

Finally, it can take a fair amount of labor to configure Asterisk and
your particular phone to work together as a system. Don't kill yourself
by attempting to mix and match various phones on the same system as that
seriously increases the complexity. Keep it simple.

For what it's worth....

Ken

[Alex Balashov]

Ken,

You have certainly had experience with a broader range of phones, so I have
no doubt you can lend more insight on this count.

But for what it's worth, my experience is largely confined to the Cisco
7960s. I've never had any trouble getting any SIP firmware image to
register with Asterisk, nor configuring them by hand against Asterisk
in network situations that don't lend themselves to autoprovisioning
setups. And I've never had any issues with features like MWI or various
other notifications.

About the only thing I've run into is that some of the older default
dialplan.xml's tend to be hostile to numbers that start with 8xx, such as
(but not exclusively) toll-free numbers. TFTP provisioning is, of course,
the best way to blank those.

Other than that, I've got no complaints. Awesome speakerphone, nice
configuration interface, conferencing features, etc. And while they
are obviously Call Manager-centric, I wouldn't go so far as to say
that Cisco provides no documentation on how to get them to work
otherwise; I've needed -- and found -- it.

Thanks,

-- Alex


--
Alex Balashov <***@presidium.org>

[Yuan LIU]

Date: Thu, 12 Apr 2007 12:02:52 -0700 (PDT)
Hi there list!
I want to catch all numbers that don't exist, play a
nice message and restart operator, this is different
from dial i because that is for incorrect extensions,
an undefined number will give a busy signal, something
I don't like

May be you can explain what is the difference between an undefined extention
(number) and an extention (number) that doesn't exist. You can certainly
use extension i to catch wrong numbers and play nice messages instead of
busy.

If you only want to catch numbers that matches a certain pattern but don't
exist in your system, and want to give busy signal to all other dialed
numbers, you can match the pattern and transfer to another context, then use
i in that context. For example, suppose your extensions should start with
2,3,4 and must be 3 digits, but you have only defined 200-242, 320-350, and
400-420, you can do (untested)

[incoming]
exten => _[2-4]XX,1,Goto(valid,${EXTEN},1)
exten => i,1,Congestion; give busy to any other dialed number

[valid]
exten => _2[0-4]X,1,Dial(SIP/${EXTEN})
exten => _24[12],1,Dial(SIP/${EXTEN})
exten => _3[2-4]X,1,Dial(SIP/${EXTEN})
exten => _4[0-1]X,1,Dial(SIP/${EXTEN})
exten => 420,1,Dial(SIP/${EXTEN})
exten => 350,1,Dial(SIP/${EXTEN})
exten => i,1,Answer(); if
exten => i,n,Playback(nice-message)
exten => i,n,DISA(nopassword,incoming)

Hope this helps.

Yuan Liu

You can search for the word irc to see my comments,
the line above is my latest unsuccessful test, thanks!

[Salvatore Giudice]

Product selection is not cut and dry. What are your business requirements?

So you need encryption? If so, what kind?
Do they need support for outbound proxies?
Are you going to use the same model for remote deployments?
Do you need WAP capabilities?
Do you need programmable speed dials?
Do you need modular admin sidecars?
Do you need IPSEC capabilities built into the handset?
Do you need advanced/specific codec support?
Do you need guaranteed interoperability with specific vendor supplied
components?
Are you looking for a phone for 10 people, 100 people, or 10000 people? If
you are scaling, what does your provisioning system look like?
Do you need phone features like video or quality speaker phone?
What is your budget for phones?
Do you need an RTCP capable handset?
Do you need a handset that support 802.11p for QoS?

The more specific you can get about your business requirements, the better
targeted your product selection will be.

--------------------------------------------------
Salvatore Giudice
***@VoIPSecurityTraining.com

VoIP Security Training, LLC
http://VoIPSecurityTraining.com

848 N. Rainbow Blvd. #1676
Las Vegas, NV 89107
Phone: [phone]
Fax: [phone]


-----Original Message-----
From: asterisk-users-***@lists.digium.com
[mailto:asterisk-users-***@lists.digium.com] On Behalf Of Stephen Bosch
Sent: Wednesday, April 11, 2007 5:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Which SIP phones to buy?

I need to buy some new phones for our own offices.

I've used only Polycom phones until now, but I'd like to broaden my
experience.

I'm trying to decide which phones to experiment with. I have these options:

- A combination of Polycom, Aastra and Snom

- Just Polycom

One the one hand, I'd like to keep things uniform, since it greatly
simplifies provisioning. On the other hand, I don't want to broaden my
knowledge.

Advice, anyone?

-Stephen-
_______________________________________________
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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[Stephen Bosch]

Post by Salvatore Giudice
Product selection is not cut and dry. What are your business requirements?
So you need encryption? If so, what kind?

No.

Post by Salvatore Giudice
Do they need support for outbound proxies?

No.

Post by Salvatore Giudice
Are you going to use the same model for remote deployments?

Yes.

Post by Salvatore Giudice
Do you need WAP capabilities?

No.

Post by Salvatore Giudice
Do you need programmable speed dials?

Yes.

Post by Salvatore Giudice
Do you need modular admin sidecars?

Maybe.

Post by Salvatore Giudice
Do you need IPSEC capabilities built into the handset?

No.

Post by Salvatore Giudice
Do you need advanced/specific codec support?

Wideband (I think that's G.729) is a "nice-to-have".

Post by Salvatore Giudice
Do you need guaranteed interoperability with specific vendor supplied
components?

Not at the moment. (No)

Post by Salvatore Giudice
Are you looking for a phone for 10 people, 100 people, or 10000 people? If
you are scaling, what does your provisioning system look like?

10 - 250; TFTP or FTP-based provisioning.

Post by Salvatore Giudice
Do you need phone features like video or quality speaker phone?

Quality speaker phone. No demand for video.

Post by Salvatore Giudice
What is your budget for phones?

up to 300 CAD per unit, preferably around 200 CAD

Post by Salvatore Giudice
Do you need an RTCP capable handset?

If I knew what that was... :)

Post by Salvatore Giudice
Do you need a handset that support 802.11p for QoS?

No.

Will that help narrow things down?

-Stephen-

[Steve Totaro]

Post by Stephen Bosch

Post by Salvatore Giudice
Product selection is not cut and dry. What are your business requirements?
So you need encryption? If so, what kind?

No.

Post by Salvatore Giudice
Do they need support for outbound proxies?

No.

Post by Salvatore Giudice
Are you going to use the same model for remote deployments?

Yes.

Post by Salvatore Giudice
Do you need WAP capabilities?

No.

Post by Salvatore Giudice
Do you need programmable speed dials?

Yes.

Post by Salvatore Giudice
Do you need modular admin sidecars?

Maybe.

Post by Salvatore Giudice
Do you need IPSEC capabilities built into the handset?

No.

Post by Salvatore Giudice
Do you need advanced/specific codec support?

Wideband (I think that's G.729) is a "nice-to-have".

Post by Salvatore Giudice
Do you need guaranteed interoperability with specific vendor supplied
components?

Not at the moment. (No)

Post by Salvatore Giudice
Are you looking for a phone for 10 people, 100 people, or 10000 people? If
you are scaling, what does your provisioning system look like?

10 - 250; TFTP or FTP-based provisioning.

Post by Salvatore Giudice
Do you need phone features like video or quality speaker phone?

Quality speaker phone. No demand for video.

Post by Salvatore Giudice
What is your budget for phones?

up to 300 CAD per unit, preferably around 200 CAD

Post by Salvatore Giudice
Do you need an RTCP capable handset?

If I knew what that was... :)

Post by Salvatore Giudice
Do you need a handset that support 802.11p for QoS?

No.
Will that help narrow things down?
-Stephen-

Polycom 301 or 501 (probably 501 since you need speakerphone and the 301
has a great speaker but no mic)

Thanks,
Steve