RE: [asterisk-users] My Phone Review- Large Scale Corp Deployment.

2006-11-02 Thread Vitalie Apostu 
 Matt,

Thanks for posting this message. What version of Asterisk do you use? What
kind of T1 card do you use on asterisk?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew Mackes
(Webmail)
Sent: Wednesday, November 01, 2006 11:27 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] My Phone Review- Large Scale Corp Deployment.


I have had the opportunity to test many IP phones in the last 6 months and I
thought you might enjoy a quick review of what I have found.



Grandstream Budgtone 200 - Poor Quality for business use- Looks good, 
and the handset feels nice, buttons have a decent feel, but the disply 
is difficult to read when you are not directly over head of the phone, 
plus the sound quality of the handset, and speaker phone is very poor--- 
Its has a cave, tunnel sound.

Cisco 7960-   Great Units, Sound Great, Look Great, very professional. 
They are abit more difficult to program then others. You must find the 
SIP firmware, and be very familiar with TFTP to deploy these phones- 
However once you get a hand of them, they are a rock.

Zulty WIP 2-   THESE PHONES ARE AWESOME!!! AWESOME!!! WiFi SIP phones- 
They look like an early 90's wireless phone, but they are VERY well 
built, and work extremely well- If you are looking for a serious WIFI 
SIP phone, this is the unit for you. Trust me, I tested 3 of the other 
popular WIFI sip phones, and they are cheaply made- even the Linksys- 
other WiFI phones are built like cheap plastic cell phones.
We have purchaced over 125 Zulty WIP 2's and they are great. A company 
named Neobits has them for sale.

And the best hard phone- The Aastra 480i.
This phone  is a class act. I would even pick this over a Cisco 7960. It 
has many features, feels great, has a very cool look, and the sound 
quality is OUTSTANDING. They have great documentation, and can do more 
then the Cisco, like Busy Lamp, which is very important in a large corp 
environment when you have receptionists.

We have chosen to go with 350 Aastra 480i, 125 Zulty WIP2, and Asterisk 
to replace our 15 year old monster Meridian Nortel switch and phones.

Asterisk will be  running on Pound Key Linux, on three HP Servers- All 
DualCore Xeons, Dual Processor machines, (so four cores per machine) 
with 4 GB of RAM per. We will also be connecting the machines with Gbit 
Ethernet to one another on a private Switch, separated from the rest of 
the  LAN/WAN.


We will be handling Voice PRI's and 2 T1s for outbound PSTN, and long 
distance, and I have planned for about 100 extensions to be active at 
any one time (1/3)

If anyone would like to discuss large scale deployments- I would love to 
hear your thoughts.

Thanks Everyone,

Matt Mackes
[EMAIL PROTECTED]
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RE: [asterisk-users] Re: Newbie Questions - Grandstorm phones?

2006-11-02 Thread Vitalie Apostu 
I am agree with you. Do you use the latest version of firmware? 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Henry.L.Coleman
Sent: Wednesday, November 01, 2006 7:09 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Re: Newbie Questions - Grandstorm phones?

I came to the same conclusion.
There is one thing however that the GXP2000 needs in my opinion.
There is no dial plan avaiable in the configuration, this means that when
dialing a number there is a slight delay before it actually dials.
With a dial plan the dialed number is sent immeadiately the pattern is match
ed so it saves a second or two. Maybe they will fix this?



Henry L.Coleman CEO
*VoIP-PBX* 1-866-415-5355
Toronto Ontario
Canada


 After doing some research on the Internet and studying all the major 
 IP phones, I have came to a conclusion that Grandstream GXP-2000 has 
 the most features of all the phones for the least price of all. I 
 don't know how they are managing to manufacture their product for such 
 a cheap price, but they're doing it well for sure. Each and every 
 other phone has something missing in it, but Grandstream GXP-2000 has 
 every necessary thing in it.
 Even if they sell their product at 2x the price, it'll still be a fair 
 price. So Grandstream GXP-2000 is the best phone to go with. I only 
 wish if they could make its face look a litter more like Polycom, that 
 would be better.

  Aastra 9133i is the second best option. Good price for the features 
 they have. A lot of lines, PoE, dual ethernet etc. Looks very 
 professional, same design as those of existing non-VoIP office phones, 
 which people are used to look at as office phones. This is becasue 
 Aastra once used to make phones for Nortel, so they have the same 
 designs for their IP phones as well. It gives more professional image. 
 The only drawback could be smaller LCD.
 They
 could make it a little bigger. I am testing it these days.

 Third best option is Linksys 942. They have two lines, you pay extra 
 for the adapter and pay extra for other two lines. This all make them 
 more than twice expensive than GXP-2000. But then they come at the 
 same level with GXP-2000. Good thing is the big display. I am also 
 testing this phone these days.

 Polycom are best looking, expensive, but configuration a little 
 difficult, and don't have backlit LCDs? And also they have limited 
 lines. Mostly no PoE.

 Snom are good, ok looking, expensive and limited lines, either no PoE 
 or no backlit LCD. But very configurable.

 And an important advice: Don't buy a phone which doesn't have backlit 
 and non-tiltable LCD, or you'll regret later.

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[asterisk-users] IVR

2006-10-30 Thread Vitalie Apostu 
Greetings,

If somebody knows how to concatenate several .gsm files in one  or create a
macro and use with background() please reply.

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RE: [asterisk-users] asterisk guru needed for job in Chicago area

2006-10-22 Thread Vitalie Apostu 
Call CompuNetWorld. +1 (704) 644-5528

-Original Message-
From: Elvar[EMAIL PROTECTED]
Sent: 10/23/06 12:03:07 AM
To: asterisk-users@lists.digium.comasterisk-users@lists.digium.com
Subject: [asterisk-users] asterisk guru needed for job in Chicago area

Hello, I run a small network consulting company in the Chicago area and 
I have a client who is interested in doing an asterisk based VOIP 
installation. My company does not have the necessary experience to carry 
out the project alone so I am looking for an asterisk guru to lead the 
project. I'm interested in someone from the Chicago or northwest Indiana 
area who is very experienced with Asterisk deployements in multi-site 
scenarios connected via VPN tunnels. The person must be very experienced 
with the following;

 -  Working with various telcos to order and troubleshoot circuits and 
phone lines 

- Analog based VOIP gateways

-  Asterisk PBX on Linux

- VOIP in general

- SIP and IAX VOIP protocols 

- Solid experience with IP networks, routers, switches, firewalls

 
The person must also be willing to come on site during deployement to 
ensure smooth integration but a good portion of the work may possibly be 
done remotely since we can handle some of it. This is for a one project 
job initially but if it goes well it could definitely open the door for 
other VOIP related projects.


For anyone who might be interested, please email me your resume.


Kind regards,
Elvar


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[Asterisk-Users] Asterisk-H323

2005-02-14 Thread Vitalie Apostu 
Greetings,

I have a problem making a call from Asterisk to Cisco H323 PSTN gateway
using H323 channel. I can call but there are no sound in both way. If I call
H323 gateway directly from SJPhone I have no problem with sound.

Any advice are welcome.

Thanks in advance.

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RE: [Asterisk-Users] Asterisk-H323

2005-02-14 Thread Vitalie Apostu 
Cisco and Asterisk are not behind firewall.

Where can I check for settings noH245Tuneling and noFastStart in Asterisk
H323?

-
-- Executing Dial(SIP/msn-069a, H323/[EMAIL PROTECTED]:1720) in new stack
-- Called [EMAIL PROTECTED]:1720
-- H323/peer:1720 is making progress passing it to SIP/msn-069a
-- H323/peer:1720 is ringing
-- H323/peer:1720 answered SIP/msn-069a
  == Spawn extension (messanger, 73952389506, 1) exited non-zero on
'SIP/msn-069a'
--

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 
Sent: Monday, February 14, 2005 11:29 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk-H323

Make sure settings for: 

noH245Tuneling and noFastStart parameters are correctly tuned both sides. 

Is Cisco or Asterisk behind NAT? 

Send more info



 
 Greetings,
 
 I have a problem making a call from Asterisk to Cisco H323 PSTN 
 gateway using H323 channel. I can call but there are no sound in both 
 way. If I call
 H323 gateway directly from SJPhone I have no problem with sound.
 
 Any advice are welcome.
 
 Thanks in advance.
 
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RE: [Asterisk-Users] Asterisk-H323

2005-02-14 Thread Vitalie Apostu 
noH245Tunneling instead of noH245Tuneling

 typedef struct call_options {

charcid_num[80];

charcid_name[80];

int noFastStart;

int noH245Tunneling;

int noSilenceSuppression;

unsigned intport;

int progress_setup;

int progress_alert;

int progress_audio;

int dtmfcodec;

} call_options_t;

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 
Sent: Monday, February 14, 2005 11:29 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk-H323

Make sure settings for: 

noH245Tuneling and noFastStart parameters are correctly tuned both sides. 

Is Cisco or Asterisk behind NAT? 

Send more info



 
 Greetings,
 
 I have a problem making a call from Asterisk to Cisco H323 PSTN 
 gateway using H323 channel. I can call but there are no sound in both 
 way. If I call
 H323 gateway directly from SJPhone I have no problem with sound.
 
 Any advice are welcome.
 
 Thanks in advance.
 
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RE: [Asterisk-Users] Asterisk-H323

2005-02-14 Thread Vitalie Apostu 
 No, I am using H323 driver

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David Liu
Sent: Monday, February 14, 2005 11:36 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk-H323

Hi there,

The settings are in oh323.conf

; Enable fast start (yes,no).
;
fastStart=yes
;
; Enable H.245 tunnelling (yes,no).
;
h245Tunnelling=yes
;
; Enable early H.245 messages in call SETUP message.
;
h245inSetup=yes
;
; Enable in-band-DTMF detection.
; (Note: Netmeeting uses in-band DTMFs)
;

I assume you are using the OH323 driver right?

Also if no audio, it could also be a codec issue. You need to set the codec
for the OH323 call in oh323.conf as well.

David
Hong Kong

On Mon, 14 Feb 2005 11:27:53 -0500, Vitalie Apostu wrote
 Cisco and Asterisk are not behind firewall.
 
 Where can I check for settings noH245Tuneling and noFastStart in 
 Asterisk H323?
 
 -
 -- Executing Dial(SIP/msn-069a, H323/[EMAIL PROTECTED]:1720) in 
 new stack-- Called [EMAIL PROTECTED]:1720-- H323/peer:1720 is 
 making progress passing it to SIP/msn-069a-- H323/peer:1720 is ringing
 -- H323/peer:1720 answered SIP/msn-069a
   == Spawn extension (messanger, 73952389506, 1) exited non-zero on 
 'SIP/msn-069a'
 --
 
 -Original Message-
 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
 Sent: Monday, February 14, 2005 11:29 AM
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] Asterisk-H323
 
 Make sure settings for:
 
 noH245Tuneling and noFastStart parameters are correctly tuned both 
 sides.
 
 Is Cisco or Asterisk behind NAT?
 
 Send more info
 

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[Asterisk-Users] H323 no sound

2005-02-14 Thread Vitalie Apostu 
Could you help me with this problem? When I call H323 gateway there is no
sound in both ways.

Here is h323 debug:
- begin 
-- Executing Dial(SIP/msn-6297, H323/[EMAIL PROTECTED]:1720) in new
stack
Allowed Codecs:
 Table:
   G.729A{sw} 1
   G.729{sw} 2
   G.711-uLaw-64k 3
   G.711-ALaw-64k 4
   UserInput/hookflash 5
   UserInput/RFC2833 6
 Set:
   0:
 0:
   G.729A{sw} 1
   G.729{sw} 2
   G.711-uLaw-64k 3
   G.711-ALaw-64k 4
 1:
   UserInput/hookflash 5
 2:
   UserInput/RFC2833 6

 -- Making call to [EMAIL PROTECTED]:1720 without gatekeeper.
== New H.323 Connection created.
-- root is calling host [EMAIL PROTECTED]:1720
--Call token is ip$localhost/31515
-- Call reference is 31515
-- DTMF Payload is 101
-- Called [EMAIL PROTECTED]:1720
-- Sending SETUP message
-- Transmitting RFC2833 on payload 101
-- Started logical channel: sending G.729A{sw}
-- channelsOpen = 1
External RTP Session Starting
RTP channel id 1 parameters:
-- remoteIpAddress: 81.17.12.22
-- remotePort: 26454
-- ExternalIpAddress: 0.0.0.0
-- ExternalPort: 14182
-- Started logical channel: receiving G.729A{sw}
-- channelsOpen = 2
External RTP Session Starting
RTP channel id 1 parameters:
-- remoteIpAddress: 81.17.12.22
-- remotePort: 26454
-- ExternalIpAddress: 0.0.0.0
-- ExternalPort: 14182
ExternalRTPChannel Destroyed
ExternalRTPChannel Destroyed
ExternalRTPChannel Destroyed
ExternalRTPChannel Destroyed
ExternalRTPChannel Destroyed
ExternalRTPChannel Destroyed
=-= In OnAlerting for call 31515: sessionId=0
-- Ringing phone for 73952389512
- Progress Indicator: 8
-- H323/peer:1720 is making progress passing it to SIP/msn-6297
-- H323/peer:1720 is ringing
-- Transmitting RFC2833 on payload 101
=-= In OnConnectionEstablished for call 31515
-- Connection Established with Unknown
-- H323/peer:1720 answered SIP/msn-6297
-- Received Facility message... 
-- Received Facility message... 
-- Received Facility message... 
-- Received Facility message... 
-- ClearCall: Request to clear call with token ip$localhost/31515,
cause 3
-- Sending RELEASE COMPLETE
channelsOpen = 1
channelsOpen = 0
ExternalRTPChannel Destroyed
ExternalRTPChannel Destroyed
  == Spawn extension (messanger, 73952389512, 1) exited non-zero on
'SIP/msn-6297'
-- ClearCall: Request to clear call with token ip$localhost/31515,
cause 7
-- Unknown has cleared the call
== H.323 Connection deleted.
 end
 

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RE: [Asterisk-Users] free pocketPC softphone (toshiba e750)

2005-02-07 Thread Vitalie Apostu 
Use SJphone. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joao Pereira
Sent: Thursday, February 03, 2005 10:47 AM
To: [EMAIL PROTECTED]; asterisk-users@lists.digium.com
Subject: [Asterisk-Users] free pocketPC softphone (toshiba e750)

Hi all
I have a pocketPC Toshiba e750 and I want to make SIP calls from it, but I
didnt found any free softphones for my Toshiba.
 X lite's versions for pocketPC isnt free :( Did someone used before a free
softphone for pocketPC? witch one?

Thanks
Joao Pereira
www.fccn.pt

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[Asterisk-Users] Looking for Asterisk termination in Russia

2005-01-17 Thread Vitalie Apostu 
I would like to make inlimited call to russia in exchange to USA.

Any idea are welcome.

Thanks.

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RE: [Asterisk-Users] Unofficial Broadvoice-users query/offer and DIDrouting question

2005-01-12 Thread Vitalie Apostu 
Let me know how we can post message in your mailing list

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike Cathey
Sent: Wednesday, January 12, 2005 9:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Unofficial Broadvoice-users query/offer and
DIDrouting question

Let me start by stating that I've been thinking about setting up an
unofficial broadvoice-users mailing list for a while.  I've asked Broadvoice
techs about it a couple times by phone and email and it looks like they
haven't passed it up to the people who would make that decision or said
people aren't interested in creating/maintaining one.
If there's enough interest (read 10 or more existing BV customers) in one,
I'd be willing to host it.  Interested parties please contact me _offlist_.
Please let me know if you think a forum/BB or ML would be preferred.

My main interest in setting up the list is to be able to troubleshoot issues
without bugging (a) BV (since they don't officially support * and don't lock
us out of their network) and (b) asterisk-users.

--Topic change--

If any BV customer with a 312/625 DID reads this and has had incoming call
issues, please let me know.  I've had incoming call routing stop working no
less than 3 times since 07/2004.  By stop working, I mean BV support (and
other customers) can call me through their softswitch(es), but calls through
the PSTN won't go through.  I've tested from 3 different NPA/NXXs and 3
separate PSTN providers in the past when it 'broke' and all were unable to
complete the call.  This seems like it could be an issue that's isolated to
the provider they're getting the DID from.  I acquired another did from them
that they're getting from a different provider (Global Crossing) so that I
can test this theory without bugging them.  312/625 is utilized by Global
Naps according to nanpa.com.

Cheers,

Mike

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[Asterisk-Users] BroadVoice

2005-01-11 Thread Vitalie Apostu 
Did somebody connect Asterisk to BroadVoice provider? If so, can you share
instruction with me?

Thanks.

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RE: [Asterisk-Users] BroadVoice

2005-01-11 Thread Vitalie Apostu 
Can you give me example of sip.conf and extention.conf which work with
broadvoice? I want users who registered with Messenger through sip to be
able to make a call thought broadvoice.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of skamp
Sent: Tuesday, January 11, 2005 10:55 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [Asterisk-Users] BroadVoice

Broadvoice has instructions on their site on how to configure asterisk with
their service, and it works i use broadvoice with asterisk 

On Tue, 2005-01-11 at 10:43 -0500, Vitalie Apostu wrote:
 Did somebody connect Asterisk to BroadVoice provider? If so, can you 
 share instruction with me?
 
 Thanks.
 
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--
skamp [EMAIL PROTECTED]

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RE: [Asterisk-Users] BroadVoice

2005-01-11 Thread Vitalie Apostu 
Following links says: HTTP 404 - File not found . Is it a right link
http://lists.digium.com/pipermail/asterisk-users/2005-January/081534.htm ?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nabeel
Jafferali
Sent: Tuesday, January 11, 2005 11:09 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE: [Asterisk-Users] BroadVoice

 Can you give me example of sip.conf and extention.conf which work with 
 broadvoice? I want users who registered with Messenger through sip to 
 be able to make a call thought broadvoice.

I posted this just a few days ago:
http://lists.digium.com/pipermail/asterisk-users/2005-January/081534.htm
l

--
Nabeel Jafferali
Tel: +1 (416) 628-9342  Toronto
 +1 (646) 225-7426  New York
FWD: 46990
Email/MSN: nabeelatjafferali.net
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RE: [Asterisk-Users] BroadVoice

2005-01-11 Thread Vitalie Apostu 
What about extention.conf? Can you share with us?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of skamp
Sent: Tuesday, January 11, 2005 11:42 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: RE: [Asterisk-Users] BroadVoice


Guys
http://lists.digium.com/pipermail/asterisk-users/2005-January/081534.html


On Tue, 2005-01-11 at 10:21 -0600, David Ishmael wrote:
 I got the same error.
 
 -Dave
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Vitalie 
 Apostu
 Sent: Tuesday, January 11, 2005 10:12 AM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: RE: [Asterisk-Users] BroadVoice
 
 Following links says: HTTP 404 - File not found . Is it a right link 
 http://lists.digium.com/pipermail/asterisk-users/2005-January/081534.htm ?
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Nabeel 
 Jafferali
 Sent: Tuesday, January 11, 2005 11:09 AM
 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - 
 Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] BroadVoice
 
  Can you give me example of sip.conf and extention.conf which work 
  with broadvoice? I want users who registered with Messenger through 
  sip to be able to make a call thought broadvoice.
 
 I posted this just a few days ago:
 http://lists.digium.com/pipermail/asterisk-users/2005-January/081534.h
 tm
 l
 
 --
 Nabeel Jafferali
 Tel: +1 (416) 628-9342  Toronto
  +1 (646) 225-7426  New York
 FWD: 46990
 Email/MSN: nabeelatjafferali.net
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--
skamp [EMAIL PROTECTED]

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[Asterisk-Users] No sound for music on hold

2005-01-11 Thread Vitalie Apostu 
Greetings,

I try to set-up Music-on-hold. I use X100P.

[mainmenu]

exten = s,1,Answer

exten = s,2,mp3player(/var/lib/asterisk/mohmp3/fpm-calm-river.mp3)

File fpm-calm-river.mp3 exist but there is sound in line. If I play this
file using mp123 I can here sound in my sound-boxes but there is no sound in
telephone line.

Thanks.

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RE: [Asterisk-Users] No sound for music on hold

2005-01-11 Thread Vitalie Apostu 
How can I check it? 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C F
Sent: Tuesday, January 11, 2005 7:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] No sound for music on hold

You have to make sure you have a timer source.


On Tue, 11 Jan 2005 17:06:45 -0700, Mark [EMAIL PROTECTED] wrote:
 Did you build the symbolic link?
 
 ln -s /usr/local/bin/mpg123 /usr/bin/mpg123
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Vitalie 
 Apostu
 Sent: Tuesday, January 11, 2005 4:19 PM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] No sound for music on hold
 
 Greetings,
 
 I try to set-up Music-on-hold. I use X100P.
 
 [mainmenu]
 
 exten = s,1,Answer
 
 exten = 
 s,2,mp3player(/var/lib/asterisk/mohmp3/fpm-calm-river.mp3)
 
 File fpm-calm-river.mp3 exist but there is sound in line. If I play 
 this file using mp123 I can here sound in my sound-boxes but there is 
 no sound in telephone line.
 
 Thanks.
 
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