Re: [asterisk-users] Asterisk in the call center - how do you do it?

[Kev S]

I was going to ask the same thing today as i am looking for better and more
efficient ways to run a call centre using asterisk!

Look forward to some responses. 

Kev

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ade Vickers
Sent: Thursday, 6 March 2008 8:27 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Asterisk in the call center - how do you do it?

Hi folks,

If you are running a call centre (large or small) using Asterisk, I'd be
interested to know how you log your agents in & out:

E.g.

 - Do you use AgentLogin (to force calls onto the agents, perhaps)?
 - Do you still use AgentCallbackLogin?
 - If you use AddQueueMember, are you 
- running it through the agent's phones (i.e. in the dialplan)?
- through a manager interface & some software (homebrew or otherwise)?
 - Do you allow agent hot-desking?
- if so, how do you determine which agent is logged in at which desk at
what time?
- how do you deal with authentication, or don't you bother?

It'd also be useful if you could tell me what version of Asterisk you're
using.

And, finally, a pure fishing expedition:

 - What kind of reporting (if any) do you currently get out of the Asterisk,
and are you happy with it?

The reason I'm asking this stuff is because since 2003 I've been working on
an ACD reporting product for Nortel Meridians (and, more recently, Avaya and
Cisco systems, although that's all early days); and I'm thinking that as
Asterisk gains a toe-hold in the call centre market, there maybe a market
for this reporting tool for Asterisk users too. The only downside is I just
know that my client (who owns the IPR) will never allow the s/w to be
opensourced, or even available for free :( But I guess I shouldn't be too
unhappy, as it puts the bread & butter on my table too...

All the above said - I should add that I'm a complete convert to Asterisk, &
use it daily (albeit at a fairly low & simplistic level), e.g. I've only
just got around to using a queue on my main POTS line, so I can login at any
of the 4 Asterisk boxes I use around Europe, without having horridly
complicated dialplans...

Many thanks in advance for any responses,
Ade.

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Re: [asterisk-users] Asterisk Voicemail for iPhone

[Kev S]

This looks great, Cant wait to try it on my iphone 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Carey
Sent: Friday, 29 February 2008 9:18 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk Voicemail for iPhone

Heres a little teaser for those of you with iPhones

Asterisk Voicemail for iPhone allows you to check your voicemail
messages on your house or business line from your iPhone. You can
think of it as "Visual Voicemail", but for your Asterisk PBX numbers
instead of your AT&T cell number. The technology behind it is Asterisk
(The Open-Source PBX), with iUI, Joe Hewitt's UI interface for iPhone.
This software can be installed on any Asterisk server (though you will
want to use one that is available via the Internet) and will allow you
to check messages in multiple folders, listen to messages, delete
messages, move messages, and change voicemail settings - all from your
iPhone.

Contact me with any questions or comments.

This software is unreleased. Most of the features are fully
functional, but I need to clean up certain portions of the code before
releasing it in order to avoid public ridicule. This software will be
released under the GPL or some other free license.

http://chriscarey.com/projects/asterisk/iphone/

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Re: [asterisk-users] Wait in Queue for 120 seconds for agent A to become free, THEN ring next agent

[Kev S]

Sorry to be painful, But how do i set the queue timeout?

Regards
Kev

Lenz wrote:
> You create three queues: queue A has only agent A, queue B only agent B,  
> and queue C only agent C.
> You call the firts queue witha timeout of 120 seconds; if call timed out,  
> you call queue B with a timeout of 120 seconds and so on.
> One note: this does not sound great from a service-level point of view :-)
> l.
>
>
>
> On Mon, 04 Feb 2008 06:13:17 +0100, Kev S <[EMAIL PROTECTED]> wrote:
>
>   
>> Hi all
>>
>> Just trying to set up a queue and wondering if this is possible.
>>
>> We have 3 agents, One of them is sort of the first point of contact
>>
>> What i am looking to do is
>>
>> 1. Someone rings the queue.
>>
>> 2. It rings Agent A.. If Agent A is on the phone then put them on hold
>> for 120 seconds, and if Agent A gets off the phone within those 120
>> seconds, put the call to them.
>>
>> 3. If 120 seconds expires, then call agent B, if B rings out, Call agent  
>> C
>>
>> So all i need is for the call to wait 120 seconds for agent A to get off
>> the phone. Then progress the call if they dont.
>>
>> Is this possible? If so, any pointers?
>>
>> Cheers
>> - Kev
>>
>> 
>
>
>
>   


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[asterisk-users] Wait in Queue for 120 seconds for agent A to become free, THEN ring next agent

[Kev S]

Hi all

Just trying to set up a queue and wondering if this is possible.

We have 3 agents, One of them is sort of the first point of contact

What i am looking to do is

1. Someone rings the queue.

2. It rings Agent A.. If Agent A is on the phone then put them on hold 
for 120 seconds, and if Agent A gets off the phone within those 120 
seconds, put the call to them.

3. If 120 seconds expires, then call agent B, if B rings out, Call agent C

So all i need is for the call to wait 120 seconds for agent A to get off 
the phone. Then progress the call if they dont.

Is this possible? If so, any pointers?

Cheers
- Kev

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Re: [asterisk-users] CallerID shows wrong values in manager interface

[Kev S]

remove the brackets around

("Devraj Mukherjee" <101>)


Regards
Kev
Devraj Mukherjee wrote:
> Hi everyone,
>
> My manager interface seems to be producing wrong CallerIDs when
> internal extensions call each other. Can anyone see anything wrong in
> the configuration snippets pasted below? The following instance has
> extension 101 call 103. The phone does show the right caller ID, but
> notice that the manager interface has the CallerID as the target
> number (103).
>
> Thanks a lot for your time.
>
> Manager interface output:
>
> CallerIDName: 
> State: Ringing
> Event: Newstate
> Privilege: call,all
> Uniqueid: 1201748091.843
> Channel: SIP/103-098500d8
> CallerID: 103
>
> SIP.conf snippets:
>
> [101]
> type=friend
> callerid=("Devraj Mukherjee" <101>)
> username=101
> secret=password
> context=default
> host=dynamic
> allow=alaw
> [EMAIL PROTECTED]
>
> [103]
> type=friend
> callerid=("System admin Den" <103>)
> username=103
> secret=password
> context=default
> host=dynamic
> allow=all
> [EMAIL PROTECTED]
>
> Extension.conf looks like:
>
> ; Standard POTS line configuration to pickup calls
> exten => _s,1,Wait(2)
> exten => _s,2,Queue(wagga-office-phones,90)
> exten => _s,3,VoiceMail([EMAIL PROTECTED])
> exten => _s,4,Hangup
>
> exten => 101,1,Wait(1)
> exten => 101,2,SetCIDNum(101)
> exten => 101,3,Dial(SIP/101,30,trw)
> exten => 101,4,Voicemail(s101)
> exten => 101,5,Hangup
>
> exten => 103,1,Wait(1)
> exten => 103,2,Dial(SIP/103,30,trw)
> exten => 103,3,Voicemail(s103)
> exten => 103,4,Hangup
>
>
>   


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Re: [asterisk-users] SIP <> GSM

[Kev S]

With that sort of set up, If for example i get a 8 channel GSM gateway 
and the X100P can i make more than 1 concurrent call though the gateway 
with the X100P or does it only support 1 call at a time?

What im looking to do is get a multi channel GSM gateway, and have the 
ability to make more than 1 call at once through it.

Thanks

-Kev

Sam Tam wrote:
> Try cyber-telecom.net
> May be get a X100P with a CT-G1000 or G2000
>
> Sam
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves
> Sent: Sunday, January 20, 2008 11:40 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] SIP <> GSM
>
> I'd like to add a device to my Asterisk server to leverage my cellular
> account. Does anyone on-list have experience with hardware gateways vs
> using cah_bluetooth and an old cell phone?
>
> I'm considering something like http://www.mobigater.com/index.php?p=5
>
> Thanks,
>
> Michael
> --
> Michael Graves
> mgravesmstvp.com
> blog.mgraves.org
> o713-861-4005
> c713-201-1262
> sip:[EMAIL PROTECTED]
> skype mjgraves
> fwd 54245
>
>
>
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Re: [asterisk-users] Your "favorite" Asterisk application.

[Kev S]

So I'm not the only one!

Kev

Anthony Francis wrote:
> Paul Hales wrote:
>   
>> I love writing dialplan, using vi.
>>
>> Does that make me weird?
>>
>> PaulH
>>
>>
>> On Wed, 2008-01-23 at 23:57 -0500, Ken D'Ambrosio wrote:
>>   
>> 
>>> Hi, all.  I've done some Asterisk recelling, but recently got roped into a
>>> Sr. SysAdmin position.  Our PBX is c. 1823, and -- well, as pretty much
>>> all circuit-based systems do, it sucks.  It sucks to administer, moves
>>> suck... you know the drill.  So, I'd love change to an Asterisk system. 
>>> My boss, who loves to spend money for no particular reason, wants to go
>>> proprietary, though.  So I'm going to have to try to sell him.  I figured
>>> one place to start would be some of the really cool applications that
>>> Asterisk has that -- generally, at least -- don't require licensing.  Some
>>> of my favorites are follow-me, meetme, voicemail-to-e-mail and
>>> fax-to-e-mail.  What are some of your favorite features/applications, be
>>> ith native or third-party?
>>>
>>> Thanks,
>>>
>>> -Ken
>>>
>>>
>>> 
>>>   
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>> 
> I simply love vi, to the point which if an IDE doesn't have vi key 
> bindings I loath using it.
>
> Anthony
>
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Re: [asterisk-users] Your "favorite" Asterisk application.

[Kev S]

The fact that it is so amazingly configurable should be enough :)

-Kev
Ken D'Ambrosio wrote:
> Hi, all.  I've done some Asterisk recelling, but recently got roped into a
> Sr. SysAdmin position.  Our PBX is c. 1823, and -- well, as pretty much
> all circuit-based systems do, it sucks.  It sucks to administer, moves
> suck... you know the drill.  So, I'd love change to an Asterisk system. 
> My boss, who loves to spend money for no particular reason, wants to go
> proprietary, though.  So I'm going to have to try to sell him.  I figured
> one place to start would be some of the really cool applications that
> Asterisk has that -- generally, at least -- don't require licensing.  Some
> of my favorites are follow-me, meetme, voicemail-to-e-mail and
> fax-to-e-mail.  What are some of your favorite features/applications, be
> ith native or third-party?
>
> Thanks,
>
> -Ken
>
>
>   


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Re: [asterisk-users] I am having a problem connecting my X-Litetomy Asterix box

[Kev S]

root login is not permitted by default via ssh

Try the username "admin" and the password you set during the install

Andrew Ladanowski wrote:
> I can not exit out of my Asterisk set up it.  When I try to login to my 
> server using ssh in denies the username and password.  I assume the default 
> name was "root" when I set up the Asterisk.  I remember the password.
>
> Andrew Ladanowski
> AddInSolutions Inc.
> www.addinsol.com
> [EMAIL PROTECTED]
> Phone: 954-815-2402
> Fax: 954-414-8432
>  
>  
> CONFIDENTIAL : The information in this email (including any attachments) is 
> confidential and may be privileged. If you are not the intended recipient, 
> you may not and must not read, print, forward, use or disseminate the 
> information contained herein. Although this email (and any attachments) are 
> believed to be free of any virus or other defect that might affect any 
> computer system into which it is received and opened, it is the 
> responsibility of the recipient to ensure that it is free of viruses or 
> defects and no responsibility is accepted by the sender for any loss or 
> damage arising or resulting in any way from its receipt or use. If you are 
> not the intended recipient of this message, please reply to the sender and 
> include this message and then delete this message from your inbox and your 
> archive and/or discarded messages files. 
>
>
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erik Anderson
> Sent: Sunday, January 20, 2008 9:14 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] I am having a problem connecting my X-Litetomy 
> Asterix box
>
> On Jan 20, 2008 8:06 PM, Andrew Ladanowski <[EMAIL PROTECTED]> wrote:
>   
>> Windows XP.
>> 
>
> Andrew - you're going to need to get us your sip.conf before we can
> really assist you any further.
>
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Re: [asterisk-users] I am having a problem connecting my X-Lite tomy Asterix box

[Kev S]

1. SSH Into the server
2. cd /etc/asterisk/
3. cat sip.conf

and copy and paste the output here

Regards
Kev
Andrew Ladanowski wrote:
> Here are my log information.   
> [Jan 20 12:34:00] NOTICE[2637] chan_sip.c: Registration from 
> '"Andrew"' failed for '192.168.3.116' - Device does 
> not match ACL
> [Jan 20 12:35:33] NOTICE[2637] chan_sip.c: Registration from 
> '"Andrew"' failed for '192.168.3.116' - Device does 
> not match ACL
>
> I am not a Linux guy I am a Windows Programmer I can not get to the sip.conf?
>
> Andrew Ladanowski
> AddInSolutions Inc.
> www.addinsol.com
> [EMAIL PROTECTED]
> Phone: 954-815-2402
> Fax: 954-414-8432
>  
>  
> CONFIDENTIAL : The information in this email (including any attachments) is 
> confidential and may be privileged. If you are not the intended recipient, 
> you may not and must not read, print, forward, use or disseminate the 
> information contained herein. Although this email (and any attachments) are 
> believed to be free of any virus or other defect that might affect any 
> computer system into which it is received and opened, it is the 
> responsibility of the recipient to ensure that it is free of viruses or 
> defects and no responsibility is accepted by the sender for any loss or 
> damage arising or resulting in any way from its receipt or use. If you are 
> not the intended recipient of this message, please reply to the sender and 
> include this message and then delete this message from your inbox and your 
> archive and/or discarded messages files. 
>
>
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erik Anderson
> Sent: Sunday, January 20, 2008 8:35 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] I am having a problem connecting my X-Lite tomy 
> Asterix box
>
> On Jan 20, 2008 7:14 PM, Andrew Ladanowski <[EMAIL PROTECTED]> wrote:
>   
>> I have added two extentsions.  I am try to test connecting X-lite to the
>> server.
>>
>> I have two extension one 1000 with password 1234 and one 2000 with password
>> 2000.
>> 
>
> Andrew - could you send us the relevent sections of your sip.conf?
> That would be quite helpful in helping you troubleshoot this problem.
> Also, please post any messages that appear on the asterisk console
> when you try and register your x-lite phone.
>
> -Erik
>
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Re: [asterisk-users] Voicemail systems- flow charts, digit/key cards, etc

[Kev S]

I too would like this, Please feel free to post a link on the list :)

Regards
Kevin


Justin Newman wrote:
> Does anyone have flow charts or digit/key cards for some of the more popular 
> voicemail systems out there?
> (shows which digits/keys to press, where it takes you, etc.)
>
> I need to create some of the new voicemail system.
>
> Send 'em my way if you have them.
>
> nt_jnewman at yahoo.com
>
> Justin
>
>
>   
> 
> Looking for last minute shopping deals?  
> Find them fast with Yahoo! Search.  
> http://tools.search.yahoo.com/newsearch/category.php?category=shopping
>
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[asterisk-users] Congestion/Forbidden issue with new carrier

[Kev S]

Hi everyone,

having a issue with asterisk and my new Voip providers service.
Iv set up many asterisk systems before but never seen this and have 
tried to fix this with no luck..
I have used this exact same sort of setup for 5 other providers and 
never had this issue, If i replace the trunk login details with my works 
voip account and set it to IAX then it works perfect, Just not the new 
provider,
I have also tried this on our work asterisk server, our development box 
and my home box all with the same bad result .
When i make a call, Straight away it just says Congested and i get a 
forbidden error.. Although Incoming calls work fine, and my provider 
confirms that i am authenticated.
Here is what happens when i make a call, i have put xx on the numbers 
and passwords. The dialplan strips the 0 in front of the number.


-- Executing [EMAIL PROTECTED]:1­ ] 
Macro("SIP/400-08280ae0", "trunkdial|SIP/trunk_1/043401"­ ) in new stack

-- Executing [EMAIL PROTECTED]:1] Dial("SIP/400-08280ae0", 
"SIP/trunk_1/043401") in new stack

-- Called trunk_1/043401

[Jan 11 14:33:16] WARNING[2439]: chan_sip.c:11918 
handle_response_invite: Received response: "Forbidden" from '"400" 
mailto:[EMAIL PROTECTED]>>;tag=as0767eb78'

-- SIP/trunk_1-08284e38 is circuit-busy

== Everyone is busy/congested at this time (1:0/1/0)

-- Executing [EMAIL PROTECTED]:2] Goto("SIP/400-08280ae0", 
"s-CONGESTION|1") in new stack
-- Goto (macro-trunkdial,s-CONGESTION,1)
-- Executing [EMAIL PROTECTED]:1] NoOp("SIP/400-08280ae0", 
"") in new stack

== Auto fallthrough, channel 'SIP/400-08280ae0' status is 'CONGESTION'



I cant work out what in the world this is, Why is the phone saying 
forbidden?

[Jan 11 14:33:16] WARNING[2439]: chan_sip.c:11918 
handle_response_invite: Received response: "Forbidden" from '"400" 
mailto:[EMAIL PROTECTED]>>;tag=as0767eb78



I can receive incoming call fine, Here is a copy of relevant parts of 
the configs and other info

Trunk Info

[trunk_1]
disallow =
allow = all
callerid = 028012
contact =
context = DID_trunk_1
dialformat = ${EXTEN:1}
fromdomain = iinetphone.iinet.net.au
fromuser = 028012
group =
hasexten = no
hasiax = no
hassip = yes
host = sip.nsw.iinet.net.au
insecure = very
port = 5060
provider =
registeriax = no
registersip = yes
secret = 
trunkname = Custom - iinet
trunkstyle = customvoip
username = 028012


The dialplan, Just dial 0, then number, then strip the first 0 and dial

[numberplan-custom-2]
include = default
plancomment = home
exten = _0X!,1,Macro(trunkdial,${trunk_1}/­ ${EXTEN:1})
comment = _0X!,1,All Numbers,standard


The trunks context, Wich is all incoming calls go to exten 400 (office)

[DID_trunk_1]
include = default
exten = _X.,1,Goto(default|400|1)
exten = s,1,Goto(default|400|1)


sip show registry

asdev*CLI> sip show registry
Host Username Refresh State Reg.Time
sip.nsw.iinet.net.au:5060 028012 105 Registered Fri, 11 Jan 2008 
14:43:01
asdev*CLI>


sip show peers

asdev*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
400/400 172.17.16.66 D 5060 Unmonitored
trunk_1/0280125553 203.59.xx.xx 5060 Unmonitored
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 
offline]
asdev*CLI>


Any help would be appreciated, Thanks in advance

Kevin S

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Re: [asterisk-users] IEEE 802.1x capable sip phones

[Kev S]

Im pretty sure the Cisco Unified IP Phones 7900 Series phones support 
this, Dont quote me on it but its worth checking out

Kev


Jeronimo Romero wrote:
>
> Does anyone know if sip phones from any of the major IP phone vendors 
> support 802.1x authentication? Any feedback would be greatly appreciated.
>
>  
>
> Thanks in advance.
>
>  
>
> ==
> Jeronimo Romero
> EUS Networks
> Email: [EMAIL PROTECTED] 
> Cell: 917-332-7238
> Office: 212-624-5943
> Web: www.euscorp.com 
> ==
>
>  
>
>  
>
>  
>
> 
>
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Re: [asterisk-users] Linksys SPA-9xx Audio Issues

[Kev S]

No, I haven't experienced this.

I think were lucky because most voip phones are in there own offices, I 
will check with our sales manager this afternoon who sits in the call 
center and see what the background noise is like on her phone.

I guess i'm just lucky that its a quiet environment, But there are a few 
people who *may* be affected and i will check this out and let you know.

Regards,
Kevin

Daniel Cole wrote:
> I have found with a number of clients to who we have installed the LinkSys 
> phones, that when you get the input gains to 6, that the phones have a 
> tendency to pick up too much background noise. Have you experienced this at 
> all?
>
> Cheers,
>
> Daniel Cole
>
>
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kev S
> Sent: Wednesday, 9 January 2008 12:35 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Linksys SPA-9xx Audio Issues
>
> The issues i have been having are probably similar to the original message, I 
> use the Linksys 9XX Series phones and we used to always receive complaints 
> from the person we were calling that they could hardly hear us.
>
> I fixed this by:
>
> Going into the Phone section of the config and setting the Handset, 
> Speakerphone and Headset input gain to 6.
>
> And i also went into SIP and changed the RTP Packet Size to 0.020
>
> This resolved the low volume issue, Sorry if you have a no sound issue, but 
> thats how i resolved very low volume.
>
> Phones sound great now!
>
> Regards,
> Kevin Sandalin
>
> Daniel Cole wrote:
>   
>> Can you describe the issue more please? Can the remote person not hear you 
>> at all? Or is there distorted/broken voice?
>>
>>
>> Cheers,
>>
>> Daniel Cole
>>
>>
>> -Original Message-
>> From: [EMAIL PROTECTED]
>> [mailto:[EMAIL PROTECTED] On Behalf Of Andrew
>> Joakimsen
>> Sent: Wednesday, 9 January 2008 9:26 AM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: [asterisk-users] Linksys SPA-9xx Audio Issues
>>
>> Anyone else have problems with phones like SPA-922, SPA-921, etc?
>> Inbound audio is perfect but the remote end reports audio quality issues on 
>> the audio the handset is sending out. It's not the network I've tried 
>> asterisk 1.2, 1.4. I've used ulaw, G726, G793 & G729. Ulaw seems to be the 
>> least problematic but its still an issue.
>> Doesn't matter if we send the calls to the PSTN via IAX or SIP or PRI.
>> I don't know it if happens all the time but about 40% of the time the remote 
>> caller reports they cannot hear me.
>>
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>> 
>
>
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Re: [asterisk-users] Linksys SPA-9xx Audio Issues

[Kev S]

The issues i have been having are probably similar to the original 
message, I use the Linksys 9XX Series phones and we used to always 
receive complaints from the person we were calling that they could 
hardly hear us.

I fixed this by:

Going into the Phone section of the config and setting the Handset, 
Speakerphone and Headset input gain to 6.

And i also went into SIP and changed the RTP Packet Size to 0.020

This resolved the low volume issue, Sorry if you have a no sound issue, 
but thats how i resolved very low volume.

Phones sound great now!

Regards,
Kevin Sandalin

Daniel Cole wrote:
> Can you describe the issue more please? Can the remote person not hear you at 
> all? Or is there distorted/broken voice?
>
>
> Cheers,
>
> Daniel Cole
>
>
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew 
> Joakimsen
> Sent: Wednesday, 9 January 2008 9:26 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Linksys SPA-9xx Audio Issues
>
> Anyone else have problems with phones like SPA-922, SPA-921, etc?
> Inbound audio is perfect but the remote end reports audio quality issues on 
> the audio the handset is sending out. It's not the network I've tried 
> asterisk 1.2, 1.4. I've used ulaw, G726, G793 & G729. Ulaw seems to be the 
> least problematic but its still an issue.
> Doesn't matter if we send the calls to the PSTN via IAX or SIP or PRI.
> I don't know it if happens all the time but about 40% of the time the remote 
> caller reports they cannot hear me.
>
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Re: [asterisk-users] Which IP Phone is really the best?

[Kev S]

But With all that experience you shouldn't have a issue working out what 
IP phone to get?

I have only been in the Voip industry 3 months now and i personally know 
what phone to supply to what client, Just through testing and playing 
around with different phones.

Anyway, To answer your question I like Most Cisco phones,

Cisco would be first preference and then a Linksys SPA942 which are 
quite nice.

Regards
Kevin

William Herrera wrote:
> Alright, enough.
> At first I was to ignore to you all making statements like this one but I
> feel at this point that if I do not stop this it seems it will never stop.
> First thing first. I have a Bach. in Network Engineering. I did work for the
> Telefónica of Puerto Rico installing Asterisk (and working with Polycom,
> Cisco, Astra and Grandstream) for a bit over 2 years. I have been doing this
> now on my own business since October 2003 (www.lan-solutions.net), so I am
> not as you might think I am.
> I asked a "simple" question just to hear your opinion. It was not intended
> for so many of you waste your time (and mine) writing all this useless notes
> 
> If you would have taken the same (or less) time just to answer the question
> (or to ignore it) we al would have been able to keep it "simple", as
> intended...
> Case closed.
>
> WH
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of David Cook
> Sent: Sunday, January 06, 2008 11:42 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Which IP Phone is really the best?
>
> Seriously, if you intend on proposing this to a customer it means you are
> selling your professional services. If you are asking questions like this,
> how successful do you expect your customer engagement to be?
>
> Even if someone recommends the "best" phone for your particular application,
> you will still have zero competency with it and spend inordinate amounts of
> learning time and re-work on the customer's time. Your inexperience will
> show. Customers are demanding and you will get thrown out on your a**.
> People expect IT to fail from time to time (unfortunately), but they expect
> 100% availability from their phones. Anything less and you will find
> yourself with a priority meeting at the client that includes your manager,
> CEO and their lawyer.
>
> Nothing travels faster than a bad reputation. Walk away. Research. Build a
> lab. Learn.
>
> - dbc.
>
> From: "William Herrera" <[EMAIL PROTECTED]>
> Subject: 
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
>   
>
>   
>> I need to quote a client for a job and I was just wondering.
>>
>> Out of all the IP Phones out there, which one is the best and why?
>>
>> Thank you all, all opinions will be accepted.
>>
>> William Herrera
>> LAN/WAN Technical Consultant
>> 
>
>
>
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