Re: [asterisk-users] Wi-Fi sip phones with auto provisioning

2009-12-03 Thread Mark Best 
The Ascom i75 isn't really an 'auto-provision' out of the box WiFi
phone, but it has a fairly painless USB cradle for programming. Works
well with Asterisk  DD-WRT.


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Howes
Sent: Thursday, December 03, 2009 06:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Wi-Fi sip phones with auto provisioning


On 3 Dec 2009, at 13:55, Fred Posner wrote:


 On Dec 3, 2009, at 8:49 AM, Lefteris Zafiris wrote:

 Im looking for wifi sip phones that support auto provisioning and  
 work
 flawlessly with atserisk. Can anyone suggest me some models?


 Don't know of any wifi phone that works flawlessly whatsoever. Best  
 to consider a DECT style phone.

If you do go Wifi, don't get a WIP310...

S

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Re: [asterisk-users] Asterisk Legacy PBX

2008-10-30 Thread Mark Best 
If I were to guess (with no config files it's really just a guess). I
would think your Dial-plan logic isn't using the right 'trunk group' for
calls.

 

 

context=from-pstn

group=0

 

context=from-legacy

group=4

 

 

[from-pstn]

exten = _.,1,Dial(Zap/g4/${EXTEN},190,r)

exten = _.,n,Hangup()

 

[from-legacy]

exten = _.,1,Dial(Zap/g0/${EXTEN},190,r)

exten = _.,n,Hangup()

 

 

Please post your configs.

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sriram
Sent: Thursday, October 30, 2008 10:23 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk Legacy PBX

 

Hi All

 

I am trying to setup :

 

PSTN E1 --- Asterisk--Legacy PBX---Legacy Analog extensions.

 

I've followed steps using  :
http://www.voipinfo.org/wiki/view/Asterisk-Panasonic

 

i get the green light (sync) on both the 2nd span of digium TE420P (that
is cnnected to the legacy pbx pri card) and the pri card of the legacy
pbx. but when i try to make a call to asterisk so that it can send the
call to the legacy pbx using Dial command - it exits saying -
CHANUNAVAIL , but if i try to dial an external PSTN number the call gets
thru..

 

Any help apprecriated.

 

Thnks

Sriram

 

 
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[asterisk-users] All calls want to go out only on interface ZAP/g0

2008-09-03 Thread Mark Best 
I have a legacy PBX that I want to slowly move off of. Below is a
diagram of what I want my setup to look-like after testing.

Old Mitel---24 Channels---Asterisk---PSTN
|  |   |
Ext. 3060  SIP. 2054  Cellular
 

No matter my dial-plan logic; all calls seem to default to ZAP/g0. I
can't seem to get any calls to go directly to ZAP/g2.

NOTE: For testing 11# is added to the front of all calls coming from the
PSTN.

PSTN to Asterisk (g0) from-pstn

Asterisk to LegacyPBX (g2) from-internal

-
-Deleted all Outbound routes.
-Re-writing Zaptel to only include Port 1  Port 3 (No 'red alarms' in
zttool)
AMI, D4, E  M and Wink - Master Timing on Port 3 (source from Port 1).
-Added 'To_PSTN' on port g0.
-Added 'To_LegacyPBX' on port g2.
-Added New 'Catch all Route' to PSTN and to LegacyPBX (.)

Test Performed: SIP to Cellular = Worked

- Executing [EMAIL PROTECTED]:20] Dial(SIP/2054-b7801d70,
ZAP/g0/2085553870|300|) in new stack
-- Called g0/2085553870
-- Zap/1-1 answered SIP/2054-b7801d70

Test Performed: SIP to 3060 = Failed
SIP to 3060 seems to go out g0 then came back in from g0 

-- Goto (macro-dialout-trunk,s,17)
-- Executing [EMAIL PROTECTED]:17] Macro(SIP/2054-b7801d70,
dialout-trunk-predial-hook|) in new stack
-- Executing [EMAIL PROTECTED]:18] GotoIf(SIP/2054-b7801d70,
0?bypass|1) in new stack
-- Executing [EMAIL PROTECTED]:19] GotoIf(SIP/2054-b7801d70,
0?customtrunk) in new stack
-- Executing [EMAIL PROTECTED]:20] Dial(SIP/2054-b7801d70,
ZAP/g0/3060|300|) in new stack
-- Called g0/3060
-- Starting simple switch on 'Zap/24-1'
-- Zap/1-1 answered SIP/2054-b7801d70
== Unknown extension '11#3060' in context 'from-pstn' requested
-- Zap/24-1 Playing 'ss-noservice' (language 'en')

Added 11#3060 to both PSTN and LegacyPBX dialplan
Test Performed: SIP to 3060 = Failed
-Goes out g0 and comes back unknown.

-- Executing [EMAIL PROTECTED]:13] Set(SIP/2054-b7802098,
OUTNUM=3060) in new stack
-- Executing [EMAIL PROTECTED]:14] Set(SIP/2054-b7802098,
custom=ZAP/g0) in new stack
-- Executing [EMAIL PROTECTED]:15] GotoIf(SIP/2054-b7802098,
1?gocall) in new stack
-- Goto (macro-dialout-trunk,s,17)
-- Executing [EMAIL PROTECTED]:17] Macro(SIP/2054-b7802098,
dialout-trunk-predial-hook|) in new stack
-- Executing [EMAIL PROTECTED]:18] GotoIf(SIP/2054-b7802098,
0?bypass|1) in new stack
-- Executing [EMAIL PROTECTED]:19] GotoIf(SIP/2054-b7802098,
0?customtrunk) in new stack
-- Executing [EMAIL PROTECTED]:20] Dial(SIP/2054-b7802098,
ZAP/g0/3060|300|) in new stack
-- Called g0/3060
-- Starting simple switch on 'Zap/24-1'
-- Zap/1-1 answered SIP/2054-b7802098
== Unknown extension '11#3060' in context 'from-pstn' requested
-- Zap/24-1 Playing 'ss-noservice' (language 'en')
-- Hungup 'Zap/1-1'

NOTE: For testing 11# is added to the front of all calls comming from
the PSTN.

Trying a Misc. Destination  Inbound route combination:
Added Misc Destination 811#3060
Changed DialPLan on LegacyPBX

.
11#3060
8|11#3060
8|11.
8|.
8|1NXXNXX
8|NXX

Added 'inbound route' of 11#3060 - to go to 'Misc dest 811#3060'
Test Performed: SIP to 3060 = Failed

-- Zap/1-1 answered SIP/2054-b7801bf0
== Unknown extension '11#30603060' in context 'from-pstn' requested
-- Zap/24-1 Playing 'ss-noservice' (language 'en')
-- Hungup 'Zap/24-1'

Added only 8|. to dial plan
Test Performed: SIP to 3060 = Failed
-Fast Busy

-- Executing [EMAIL PROTECTED]:20] Dial(Zap/24-1,
ZAP/g0/811#|300|) in new stack
-- Called g0/811#

What a mess! What else can I try?

 

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[asterisk-users] e911/CAMA/MF

2008-07-09 Thread Mark Best 
Does anyone have any experience getting inbound ANI information from a
CAMA/MF/EM Wink trunk on Asterisk?

Is this only do-able with a PRI interface? Any information would be
helpful.


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Re: [asterisk-users] 2 AVM ISDN Fritzcards

2008-07-07 Thread Mark Best 
Perhaps you could try the OpenSUSE LiveCD and find out.

-Mark Best
-Network Administrator
[EMAIL PROTECTED]
-(208) 750-2054

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-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Simon
Sent: Friday, July 04, 2008 7:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 2 AVM ISDN Fritzcards

On Thu, Jul 3, 2008 at 5:07 PM, Dave Cotton [EMAIL PROTECTED]
wrote:

 Yes, with Suse 10.2/10.3 and chan_misdn.

Just to follow up on this. SLES 10.2 SP2 worked bang on. The two cards
are configured and working correctly and recognised by Asterisk.

Question: I guess you were meaning openSUSE 10.2/10.3... will openSUSE
11 work here?

Simon

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[asterisk-users] Avaya IP Phones with *

2008-06-04 Thread Mark Best 
Does anyone have any experience getting Avaya phones working with
Asterisk? (I.E. 9650) BLF etc?

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Re: [asterisk-users] Avaya IP Phones with *

2008-06-04 Thread Mark Best 
Busy Lamp features? How is the sound quality compared to
Polycom/Cisco/Snom etc? Recommend this kind of phone? 

(FYI: Doing phone research - while trying to be 'backwards'-compatible
with an Avaya IP G450/S8700 system.)

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bob G
Sent: Wednesday, June 04, 2008 3:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Avaya IP Phones with *

 

Yes we do everyday here at Google

- Original Message -
From: Mark Best 
To: Asterisk Users Mailing List - Non-Commercial Discussion 
Subject: [asterisk-users] Avaya IP Phones with *
Date: Wed, 4 Jun 2008 15:24:16 -0700


Does anyone have any experience getting Avaya phones working with
Asterisk? (I.E. 9650) BLF etc?

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-- 

Mail.com Autos http://www.oncars.com - Powered by Oncars.com: Drive By
Today!

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[asterisk-users] Mitel SX-200 + *

2008-03-04 Thread Mark Best 
Does anyone have any experience getting a Mitel SX-200 EL/ML PBX system
to work with *?

I'm not getting inbound or outbound calls to work. (Inbound caller gets
line busy tone.)

 

SETTINGS FROM MITEL:

I built a Crossover cable and connected it like this:

PSTN--T1--ASTRISK--T1--OLD MITEL

-Crossover Cable Pin-out:

1 - 4

2 - 5

4 - 1

5 - 2

Mitel Settings:

Line Coding = B8ZS Framing = D4 (T1 Card 9109-021-001-NA)

(My config: span=1,1,0,d4,b8zs  em=1-24)

 

The card is a 24 channel T1 - not a PRI right? (T1 EM Emulation Trunk:
1)

So does that mean I use EM with Winkstart?

(signalling=em_w) (is this spelt right?)

(Wink setting is an assumption based upon a setting I saw in
the PBX called wink timer)

 

Using a Sangoma A104D:

zttool

-GAVE OK on PSTN

-Gave RED on MITEL

Timing

-Receiving Time from PSTN  Giving to the Mitel

 

What debugging can I do to get more information?

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Re: [asterisk-users] Mitel SX-200 + *

2008-03-04 Thread Mark Best 
Eric,
Looks like my Mitel does support ESF; I'll try changing it. One question
before I do. If I change it right now, do you suppose my Telco
automatically adapt?
TELCO---T1---MITEL


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
Wieling
Sent: Tuesday, March 04, 2008 12:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Mitel SX-200 + *

I would strongly recommend ESF/B8ZS. If you have a RED alarm that means 
the device does not see a line connected to it -- check cabling.

Mark Best wrote:
 Does anyone have any experience getting a Mitel SX-200 EL/ML PBX
system
 to work with *?
 
 I'm not getting inbound or outbound calls to work. (Inbound caller
gets
 line busy tone.)
 
  
 
 SETTINGS FROM MITEL:
 
 I built a Crossover cable and connected it like this:
 
 PSTN--T1--ASTRISK--T1--OLD MITEL
 
 -Crossover Cable Pin-out:
 
 1 - 4
 
 2 - 5
 
 4 - 1
 
 5 - 2
 
 Mitel Settings:
 
 Line Coding = B8ZS Framing = D4 (T1 Card 9109-021-001-NA)
 
 (My config: span=1,1,0,d4,b8zs  em=1-24)
 
  
 
 The card is a 24 channel T1 - not a PRI right? (T1 EM Emulation
Trunk:
 1)
 
 So does that mean I use EM with Winkstart?
 
 (signalling=em_w) (is this spelt right?)
 
 (Wink setting is an assumption based upon a setting I saw
in
 the PBX called wink timer)
 
  
 
 Using a Sangoma A104D:
 
 zttool
 
 -GAVE OK on PSTN
 
 -Gave RED on MITEL
 
 Timing
 
 -Receiving Time from PSTN  Giving to the
Mitel
 
  
 
 What debugging can I do to get more information?
 
 
 
 


 
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Re: [asterisk-users] Mitel SX-200 + *

2008-03-04 Thread Mark Best 
Jared,
Thanks for the help. 

There are other T1 trunks (using AMI,D4) going to different departments;
however the trunk line talking to the Telco is what it is. 

I do have the ability to change it to ESF, so I will try changing it -
do you suppose my Telco will automaticly adapt if I do? (I'm fairly
confident they support ESF.)


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jared
Smith
Sent: Tuesday, March 04, 2008 1:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Mitel SX-200 + *

On Tue, 2008-03-04 at 12:34 -0800, Mark Best wrote:
 Does anyone have any experience getting a Mitel SX-200 EL/ML PBX
 system to work with *?

No, but I'll make a few comments anyway.  I'm not sure if they'll help
you at all, but maybe it'll help explain a few things to someone else on
the list.

 I'm not getting inbound or outbound calls to work. (Inbound caller
 gets line busy tone.)

Is the T1 in alarm?  You've first gotta get it out of alarm before you
can make calls on it.  Alarms are usually caused by physical
connectivity problems, or problems with framing and/or linecoding.

 Mitel Settings:
 
 Line Coding = B8ZS Framing = D4 (T1 Card 9109-021-001-NA)

Hmmmn... D4 and B8ZS together?  That's bizarre.  Usually people will use
D4 framing and AMI linecoding together, or ESF framing and B8ZS
linecoding.  I've never seen anyone uuse B8ZS linecoding with D4
framing.


-- 
Jared Smith
Community Relations Manager
Digium, Inc.


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