Re: [asterisk-users] monitoring Asterisk 1.8
Queuemetrics works well for this also, and can be installed on a separate machine/VM. www.queuemetrics.com Bruce From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of motty cruz Sent: Thursday, May 09, 2013 3:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] monitoring Asterisk 1.8 Thanks for your help; I just want to monitor the queue, calls on hold average time, incoming out going call, I only want to monitor Asterisk, not the server Asterisk in running on. thanks, -Motty On Thu, May 9, 2013 at 1:06 PM, Carlos Rojas crt.ro...@gmail.commailto:crt.ro...@gmail.com wrote: http://opennms.org/wiki/Installation:Yum On Thu, May 9, 2013 at 4:03 PM, Carlos Rojas crt.ro...@gmail.commailto:crt.ro...@gmail.com wrote: I'm using opennms and It's working fine. On Thu, May 9, 2013 at 3:23 PM, motty cruz motty.c...@gmail.commailto:motty.c...@gmail.com wrote: Hello, i'm looking for suggestions to monitor Asterisk Server? I installed Nagios but no success, I do prefer not to install any web server on the server running Asterisk. Thanks in advance. -Motty -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Does Asterisk support SIP Join Headers
I'm wondering if the SIP header join, RFC 3911, is supported in the asterisk stack? -- * Bruce Reeves, dCAp EUS Networks Office: 212-624-5943 Web: www.euscorp.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Does Asterisk support SIP Join Headers
I had seen that and figured as much. Thanks Alex.
On Tue, Oct 21, 2008 at 5:18 PM, Alex Balashov
[EMAIL PROTECTED] wrote:
chan_sip.c's sip_options[] array o' struct cfsip_options says:
/* RFC3911: SIP Join header support */
{ SIP_OPT_JOIN, NOT_SUPPORTED, join },
Bruce Reeves wrote:
I'm wondering if the SIP header join, RFC 3911, is supported in the
asterisk stack?
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Re: [asterisk-users] DUNDI Help
Sure, let me show you how I setup dundi on systems.
extensions.conf
exten = _1X,1,Goto(lookupdundi,${EXTEN},1)
[lookupdundi]
exten = _X,1,Goto(${ARG1},1)
switch = DUNDi/priv
exten = i,1,Playback(invalid)
You can have the i do whatever you want, and you can use the same
option in the macro you are using.
That is it, I leave out all the other context in the examples, from
time to time I add a dundi-static context and put in specific numbers
or patterns I want to accept, maybe for pstn calling or phones that
don't register, but in those cases I have multiple mappings in
dundi.conf for each context. For example:
priv = sipregistrations,0,SIP,[EMAIL PROTECTED],nopartial
priv = dundi-static,0,SIP,[EMAIL PROTECTED],nopartial
On Wed, Aug 27, 2008 at 3:56 AM, ronald ramos [EMAIL PROTECTED] wrote:
Hi Again,
Is there a way i can detect whether a user has been added into the
regcontext?
Currently i'm seeing this and just gives a fast busy.
[Aug 27 16:44:46] WARNING[17402]: pbx.c:2483 __ast_pbx_run: Channel
'SIP/10.10.10.10-b63101d0' sent into invalid extension '141100' in context
'lookupdundi', but no invalid handler
can i detect it somehow, so i can inform user that the extensions is not
available?
i have tried ChanIsAvail, but since i am using realtime ChanIsAvail thinks
it registered, since it really is registered on the other server. So it's
trying to call it, tries it for 30 secs (i set it to timeout at 30),
after 30 secs then it will go to DUNDI/priv. Is there a way that i can
detect it first so it does not try to dial it on the local before askng
dundi? thank you
regards,
Ron
--- On Tue, 8/26/08, Bruce Reeves [EMAIL PROTECTED] wrote:
From: Bruce Reeves [EMAIL PROTECTED]
Subject: Re: [asterisk-users] DUNDI Help
To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
Date: Tuesday, August 26, 2008, 8:16 PM
It is added when a phone registers, or re-registers. Depending on the
timing of the registrations and any restarts on the asterisk process
it may take some time for phones to re-register.
On Tue, Aug 26, 2008 at 2:10 PM, ronald ramos [EMAIL PROTECTED]
wrote:
Hi Bruce,
my apologies, but the error was because of the key.
i just run keys init on the CLI and it works,
question
on regcontext though, i set it to sipregistrations, how often
does
an extension be added to the context sipregistrations and for how long
will
it stay there? i'm looking at dialplan show sipregistration, sometimes
i
only see one extension there. even though i know i have 4 ip phones
registered to the asterisk.
TIA
Ron
--- On Tue, 8/26/08, Bruce Reeves [EMAIL PROTECTED]
wrote:
From: Bruce Reeves [EMAIL PROTECTED]
Subject: Re: [asterisk-users] DUNDI Help
To: [EMAIL PROTECTED], Asterisk Users Mailing List -
Non-Commercial
Discussion asterisk-users@lists.digium.com
Date: Tuesday, August 26, 2008, 6:23 PM
Ron,
What does the peers section in dundi.conf look like?
On Tue, Aug 26, 2008 at 3:00 AM, ronald
ramos
[EMAIL PROTECTED]
wrote:
Would like to try setting up dundi with 3-4 asterisk.
But for poc, i would like to try setting up dundi on between 2
asterisk.
I copied the config from DUNDI enterprise SIP with no password. Only
thing
i
changed is the part where i used regcontext.
on both boxes dundi.conf i have
[mapping]
priv = sipregistrations,0,SIP,[EMAIL PROTECTED],nopartial
i can see both peers on each server:
CLI dundi show peers
EID HostModel AvgTime Status
00:8e:8c:8e:cb:5310.10.10.XX (S) Symmetric Unavail OK (1 ms)
i can see my extension being added
on sipregistrations context
Added extension '136101' priority 1 to
sipregistrations
tried a dundi lookup but got no result
dundi lookup [EMAIL PROTECTED]
DUNDi lookup returned no results.
DUNDi lookup completed in 0 ms
here's what's on extensions.conf
; Private DUNDi network
[dundi-priv-canonical]
; Direct numbers
[dundi-priv-customers]
; If you are an ITSP or Reseller, list your customers here.
[dundi-priv-via-pstn]
; If you are freely delivering calls to the PSTN, list them here
[dundi-priv-local]
include = dundi-priv-canonical
include = dundi-priv-customers
include = dundi-priv-via-pstn
[dundi-priv-switch]
; Just a wrapper for the switch
switch = DUNDi/priv
[dundi-priv-lookup]
include =
dundi-priv-local
include = dundi-priv-switch
[macro-dundi-priv]
exten = s,1,Goto(${ARG1}|1)
include = dundi-priv-lookup
[diallocal]
exten = _1X,1,Macro(dundi-priv|${EXTEN})
i also tried dialing from my xlite:
[Aug 26 15:58:07] -- Executing [EMAIL PROTECTED]:1]
Macro(SIP/138100-08269548, dundi-priv|136101)
in
new stack
[Aug 26 15:58:07] -- Executing [EMAIL PROTECTED]:1]
Goto(SIP/138100-08269548, 136101|1) in new
stack
[Aug 26 15:58:07] -- Goto (macro-dundi-priv,136101,1)
[Aug 26 15:58:07] == Auto fallthrough, channel
'SIP
Re: [asterisk-users] DUNDI Help
Ron,
What does the peers section in dundi.conf look like?
On Tue, Aug 26, 2008 at 3:00 AM, ronald ramos [EMAIL PROTECTED] wrote:
Would like to try setting up dundi with 3-4 asterisk.
But for poc, i would like to try setting up dundi on between 2 asterisk.
I copied the config from DUNDI enterprise SIP with no password. Only thing i
changed is the part where i used regcontext.
on both boxes dundi.conf i have
[mapping]
priv = sipregistrations,0,SIP,[EMAIL PROTECTED],nopartial
i can see both peers on each server:
CLI dundi show peers
EID HostModel AvgTime Status
00:8e:8c:8e:cb:5310.10.10.XX (S) Symmetric Unavail OK (1 ms)
i can see my extension being added on sipregistrations context
Added extension '136101' priority 1 to sipregistrations
tried a dundi lookup but got no result
dundi lookup [EMAIL PROTECTED]
DUNDi lookup returned no results.
DUNDi lookup completed in 0 ms
here's what's on extensions.conf
; Private DUNDi network
[dundi-priv-canonical]
; Direct numbers
[dundi-priv-customers]
; If you are an ITSP or Reseller, list your customers here.
[dundi-priv-via-pstn]
; If you are freely delivering calls to the PSTN, list them here
[dundi-priv-local]
include = dundi-priv-canonical
include = dundi-priv-customers
include = dundi-priv-via-pstn
[dundi-priv-switch]
; Just a wrapper for the switch
switch = DUNDi/priv
[dundi-priv-lookup]
include = dundi-priv-local
include = dundi-priv-switch
[macro-dundi-priv]
exten = s,1,Goto(${ARG1}|1)
include = dundi-priv-lookup
[diallocal]
exten = _1X,1,Macro(dundi-priv|${EXTEN})
i also tried dialing from my xlite:
[Aug 26 15:58:07] -- Executing [EMAIL PROTECTED]:1]
Macro(SIP/138100-08269548, dundi-priv|136101) in new stack
[Aug 26 15:58:07] -- Executing [EMAIL PROTECTED]:1]
Goto(SIP/138100-08269548, 136101|1) in new stack
[Aug 26 15:58:07] -- Goto (macro-dundi-priv,136101,1)
[Aug 26 15:58:07] == Auto fallthrough, channel 'SIP/138100-08269548'
status is 'UNKNOWN'
any guess what's wrong? Thanks
ron
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Re: [asterisk-users] DUNDI Help
It is added when a phone registers, or re-registers. Depending on the
timing of the registrations and any restarts on the asterisk process
it may take some time for phones to re-register.
On Tue, Aug 26, 2008 at 2:10 PM, ronald ramos [EMAIL PROTECTED] wrote:
Hi Bruce,
my apologies, but the error was because of the key.
i just run keys init on the CLI and it works,
question on regcontext though, i set it to sipregistrations, how often does
an extension be added to the context sipregistrations and for how long will
it stay there? i'm looking at dialplan show sipregistration, sometimes i
only see one extension there. even though i know i have 4 ip phones
registered to the asterisk.
TIA
Ron
--- On Tue, 8/26/08, Bruce Reeves [EMAIL PROTECTED] wrote:
From: Bruce Reeves [EMAIL PROTECTED]
Subject: Re: [asterisk-users] DUNDI Help
To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
Date: Tuesday, August 26, 2008, 6:23 PM
Ron,
What does the peers section in dundi.conf look like?
On Tue, Aug 26, 2008 at 3:00 AM, ronald ramos [EMAIL PROTECTED]
wrote:
Would like to try setting up dundi with 3-4 asterisk.
But for poc, i would like to try setting up dundi on between 2 asterisk.
I copied the config from DUNDI enterprise SIP with no password. Only thing
i
changed is the part where i used regcontext.
on both boxes dundi.conf i have
[mapping]
priv = sipregistrations,0,SIP,[EMAIL PROTECTED],nopartial
i can see both peers on each server:
CLI dundi show peers
EID HostModel AvgTime Status
00:8e:8c:8e:cb:5310.10.10.XX (S) Symmetric Unavail OK (1 ms)
i can see my extension being added
on sipregistrations context
Added extension '136101' priority 1 to sipregistrations
tried a dundi lookup but got no result
dundi lookup [EMAIL PROTECTED]
DUNDi lookup returned no results.
DUNDi lookup completed in 0 ms
here's what's on extensions.conf
; Private DUNDi network
[dundi-priv-canonical]
; Direct numbers
[dundi-priv-customers]
; If you are an ITSP or Reseller, list your customers here.
[dundi-priv-via-pstn]
; If you are freely delivering calls to the PSTN, list them here
[dundi-priv-local]
include = dundi-priv-canonical
include = dundi-priv-customers
include = dundi-priv-via-pstn
[dundi-priv-switch]
; Just a wrapper for the switch
switch = DUNDi/priv
[dundi-priv-lookup]
include =
dundi-priv-local
include = dundi-priv-switch
[macro-dundi-priv]
exten = s,1,Goto(${ARG1}|1)
include = dundi-priv-lookup
[diallocal]
exten = _1X,1,Macro(dundi-priv|${EXTEN})
i also tried dialing from my xlite:
[Aug 26 15:58:07] -- Executing [EMAIL PROTECTED]:1]
Macro(SIP/138100-08269548, dundi-priv|136101) in
new stack
[Aug 26 15:58:07] -- Executing [EMAIL PROTECTED]:1]
Goto(SIP/138100-08269548, 136101|1) in new stack
[Aug 26 15:58:07] -- Goto (macro-dundi-priv,136101,1)
[Aug 26 15:58:07] == Auto fallthrough, channel
'SIP/138100-08269548'
status is 'UNKNOWN'
any guess what's wrong? Thanks
ron
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Re: [asterisk-users] implementing an intercom with asterisk
Some one already touched on this, but my guess is the Nortel system is sending the page signal out to an actual paging system and the speakers are in the remote building or the page port on the Nortel is running over cat 3 copper to the other building. in either case tie it in to the Asterisk system via SIP ATA or FXO port on the box. I have done a number of these setups with an extra FXO port connected to a bogen or viking system, even page pac. On Tue, Aug 26, 2008 at 3:02 PM, Jonathan Disher [EMAIL PROTECTED] wrote: On Aug 26, 2008, at 2:27 AM, Gordon Henderson wrote: Do you have some sort of IP connectivity between the sites? 400 yards is a too long for copper cat5, but can be done with fibre, wireless or free-space optics... (which I don't personally recommend!) The current plan is wireless bridge + directional antennae. That wasn't the problem I needed to solve. (And if you haven't IP how are you talking to the phones between sites?) So what's to stop you from putting a Cisco phone into auto-answer mode and calling it via ths Page() application? This is an industrial environment. I'm looking for a slightly less expensive (and hopefully more robust) device - whether an intercom unit + ATA or a magic black box that does everything I want and has a power plug and an ethernet jack. Dedicating a $175 cisco phone to this is overkill, IMO. I had given thought to this, it is a backup plan, but again, I'd like to get something perhaps less expensive to the function. -j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- * Bruce Reeves, dCAp EUS Networks Office: 212-624-5943 Web: www.euscorp.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need application, CID number match list to call cell phone
Hey JR, Is this a one VIP to one cell number match? Or is it on VIP to multiple cells? On Tue, Aug 26, 2008 at 7:28 PM, JR Richardson [EMAIL PROTECTED] wrote: Hi All, I received a request for a special application and need some guidance. Cust has there own Asterisk PBX with SIP phones, pretty standard setup. They want an after hours application that checks inbound caller ID numbers and matches them to a list, say 5 to 10 numbers of special VIP customers, if there is a match on the list, then forward the call straight to a cell phone, instead of ringing local extension and then to voicemail. The customer also wants to be able to manage this VIP list and the call forward cell phone number themselves, so it needs to be configured, numbers added and deleted, through a web page on the PBX. So I'm thinking I need a dialplan app that has to interface with a MySQL database that holds the list of numbers, so I can build a webpage to add/delete the numbers. Any ideas would be much appreciated. Thanks. JR - JR Richardson Engineering for the Masses ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- * Bruce Reeves, dCAp EUS Networks Office: 212-624-5943 Web: www.euscorp.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk for larg
You have the basic idea right, the dial plans are limited down to specific functions to be provided and then told how to connect to other features. For example you might have a box that only provide PSTN connectivity so all calls come in and the dial plan routes the calls to another box or boxes. I prefer to use Dundi in larger setups to avoid multiple IAX trunks having to be configured. On Mon, Apr 28, 2008 at 7:19 PM, gmail [EMAIL PROTECTED] wrote: Does anyone know how to off-load an Asterisk Box so that to distribute its functions like IVR and VoiceMail or its PTSN gateway function into different servers? in this case , will the installation of Asterisk on each server differe and how these different servers will interact as a single logical -vs physical- server? thx alot ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- * Bruce Reeves, dCAp EUS Networks Office: 212-624-5943 Web: www.euscorp.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DUNDi and SIP
Jeremy,
It is not the dip peer that is failing but the dial plan:
-- Goto (macro-dundi-lookup,400,1)
[Apr 23 12:46:44] WARNING[2269]: chan_sip.c:2898 create_addr: No such
host: 192.168.4.51/400
[Apr 23 12:46:44] WARNING[2269]: app_dial.c:1191 dial_exec_full:
Unable to create channel of type 'SIP' (cause 3 - No route to
destination)
== Everyone is busy/congested at this time (1:0/0/1)
What is in the context macro-dundi-lookup?
On Wed, Apr 23, 2008 at 12:47 PM, Jeremy Mann [EMAIL PROTECTED] wrote:
Nope..
asterisk*CLI dundi lookup [EMAIL PROTECTED]
1. 0 SIP/priv:[EMAIL PROTECTED]/400 (EXISTS)
from 00:1e:0b:dd:e9:99, expires in 5 s
DUNDi lookup completed in 104 ms
-- Executing [EMAIL PROTECTED]:1] Set(SIP/156-08274b60,
CDR(accountcode)=wth) in new stack
-- Executing [EMAIL PROTECTED]:2] Set(SIP/156-08274b60,
CALLERID(all)=Corporate 100) in new stack
-- Executing [EMAIL PROTECTED]:3] Macro(SIP/156-08274b60,
dundi-lookup|400) in new stack
-- Executing [EMAIL PROTECTED]:1] Goto(SIP/156-08274b60, 400|1) in
new stack
-- Goto (macro-dundi-lookup,400,1)
[Apr 23 12:46:44] WARNING[2269]: chan_sip.c:2898 create_addr: No such host:
192.168.4.51/400
[Apr 23 12:46:44] WARNING[2269]: app_dial.c:1191 dial_exec_full: Unable to
create channel of type 'SIP' (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [EMAIL PROTECTED]:4] Hangup(SIP/156-08274b60, ) in new
stack
== Spawn extension (from-sip, 400, 4) exited non-zero on 'SIP/156-08274b60'
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves
Sent: Tuesday, April 22, 2008 10:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and SIP
Try this,
[priv]
dbsecret=dundi/secret
disallow=all
allow=ulaw
canreinvite=no
nat=no
context=from-internal
type=friend
priv = dundi-priv-local,0,SIP,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
On Tue, Apr 22, 2008 at 8:23 AM, Jeremy Mann [EMAIL PROTECTED] wrote:
No.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce
Reeves
Sent: Tuesday, April 22, 2008 6:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and SIP
Jeremy,
Did you get this working?
On Thu, Apr 17, 2008 at 7:38 AM, Jeremy Mann [EMAIL PROTECTED] wrote:
I have it working via IAX, when I try changing everything to SIP I
can't specify a username and an extension, so it becomes useless.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce
Reeves
Sent: Thursday, April 17, 2008 6:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and SIP
Jeremy,
Here is a working sample to compare to. This is an IAX2 setup, but the
only difference is in the mapping change IAX2 to SIP. Notice the 4th
setting in the mapping? It defines to use the IAX2 peer priv with
the secret generated of the key defined in the peers section of
dundi.conf. When you look at the peer in iax.conf on the remote box,
there is no host entry and it uses dbsecret=dundi/secret, the
dundi.conf
priv = dundi-internal,0,IAX2,priv:[EMAIL
PROTECTED]/${NUMBER},nopartial
[00:19:66:1C:78:D5] ; Dev Box
model = symmetric
host = 192.168.99.252
inkey = eus
outkey = eus
include = priv
permit = priv
qualify = yes
From iax.conf
[priv]
type=friend
dbsecret=dundi/secret
context=longdistance
Hope this helps, in your case Dundi will save you a world of work on
configuring that many systems, in fact if you structure Dundi like
spokes around a small number of master servers, the config gets real
easy.Let me know how it goes.
On Wed, Apr 16, 2008 at 8:41 AM, Jeremy Mann [EMAIL PROTECTED] wrote:
I'm a little confused with DUNDi and SIP as the backend channel type:
Dundi.conf:
[mappings]
priv = dundi-priv-local,0,SIP,[EMAIL PROTECTED],nopartial
Using the above, the dial string passed to the person on the other
box is
SIP/[EMAIL PROTECTED]
How can you use authentication, along with SIP, along with specifying
extension?
My sip.conf has a friend defined:
[priv]
host=dynamic
secret=priv
disallow=all
allow=ulaw
canreinvite=no
nat=no
context=from-internal\
type=friend
Re: [asterisk-users] DUNDi and SIP
Take a look at this setup, it does not use passwords on the sip peers
or the mappings in Dundi. As long as you inside your network this
maybe the way to go.
http://www.voip-info.org/wiki/view/DUNDi+Enterprise+Configuration+SIP+with+no+passwords
You could also look at the incominglimit and outgoinglimit on IAX peers
On Wed, Apr 23, 2008 at 4:51 PM, Jeremy Mann [EMAIL PROTECTED] wrote:
I'm fairly sure SIP will never work unless I hard-code peers everywhere,
which isn't going to happen. The only reason I want to use it is for the
call-limit option.
Looking at sip channels there is no option to pass the extension after the
IP, it's always [EMAIL PROTECTED], or [EMAIL PROTECTED], not [EMAIL
PROTECTED]/extension or [EMAIL PROTECTED]/extension
Looks like IAX and ZAP are the only two channel types that do a /extension
type setup.
Extensions.conf:
[macro-dundi-lookup]
exten = s,1,Goto(${ARG1},1)
include = dundi-priv-local
include = dundi-priv-lookup
[dundi-priv-local]
include = internal
[dundi-priv-lookup]
switch = DUNDi/priv
Dundi.conf:
[mappings]
priv = dundi-priv-local,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves
Sent: Wednesday, April 23, 2008 4:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and SIP
Jeremy,
It is not the dip peer that is failing but the dial plan:
-- Goto (macro-dundi-lookup,400,1)
[Apr 23 12:46:44] WARNING[2269]: chan_sip.c:2898 create_addr: No such
host: 192.168.4.51/400
[Apr 23 12:46:44] WARNING[2269]: app_dial.c:1191 dial_exec_full:
Unable to create channel of type 'SIP' (cause 3 - No route to
destination)
== Everyone is busy/congested at this time (1:0/0/1)
What is in the context macro-dundi-lookup?
On Wed, Apr 23, 2008 at 12:47 PM, Jeremy Mann [EMAIL PROTECTED] wrote:
Nope..
asterisk*CLI dundi lookup [EMAIL PROTECTED]
1. 0 SIP/priv:[EMAIL PROTECTED]/400 (EXISTS)
from 00:1e:0b:dd:e9:99, expires in 5 s
DUNDi lookup completed in 104 ms
-- Executing [EMAIL PROTECTED]:1] Set(SIP/156-08274b60,
CDR(accountcode)=wth) in new stack
-- Executing [EMAIL PROTECTED]:2] Set(SIP/156-08274b60,
CALLERID(all)=Corporate 100) in new stack
-- Executing [EMAIL PROTECTED]:3] Macro(SIP/156-08274b60,
dundi-lookup|400) in new stack
-- Executing [EMAIL PROTECTED]:1] Goto(SIP/156-08274b60, 400|1) in
new stack
-- Goto (macro-dundi-lookup,400,1)
[Apr 23 12:46:44] WARNING[2269]: chan_sip.c:2898 create_addr: No such
host: 192.168.4.51/400
[Apr 23 12:46:44] WARNING[2269]: app_dial.c:1191 dial_exec_full: Unable
to create channel of type 'SIP' (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [EMAIL PROTECTED]:4] Hangup(SIP/156-08274b60, ) in
new stack
== Spawn extension (from-sip, 400, 4) exited non-zero on
'SIP/156-08274b60'
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce
Reeves
Sent: Tuesday, April 22, 2008 10:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and SIP
Try this,
[priv]
dbsecret=dundi/secret
disallow=all
allow=ulaw
canreinvite=no
nat=no
context=from-internal
type=friend
priv = dundi-priv-local,0,SIP,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
On Tue, Apr 22, 2008 at 8:23 AM, Jeremy Mann [EMAIL PROTECTED] wrote:
No.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce
Reeves
Sent: Tuesday, April 22, 2008 6:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and SIP
Jeremy,
Did you get this working?
On Thu, Apr 17, 2008 at 7:38 AM, Jeremy Mann [EMAIL PROTECTED] wrote:
I have it working via IAX, when I try changing everything to SIP I
can't specify a username and an extension, so it becomes useless.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of
Bruce Reeves
Sent: Thursday, April 17, 2008 6:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and SIP
Jeremy,
Here is a working sample to compare to. This is an IAX2 setup, but
the
only difference is in the mapping change IAX2 to SIP. Notice the 4th
setting in the mapping? It defines to use the IAX2 peer priv with
the secret generated of the key defined in the peers section of
dundi.conf. When you look at the peer in iax.conf on the remote box
Re: [asterisk-users] DUNDi and SIP
Jeremy,
Did you get this working?
On Thu, Apr 17, 2008 at 7:38 AM, Jeremy Mann [EMAIL PROTECTED] wrote:
I have it working via IAX, when I try changing everything to SIP I can't
specify a username and an extension, so it becomes useless.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves
Sent: Thursday, April 17, 2008 6:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and SIP
Jeremy,
Here is a working sample to compare to. This is an IAX2 setup, but the
only difference is in the mapping change IAX2 to SIP. Notice the 4th
setting in the mapping? It defines to use the IAX2 peer priv with
the secret generated of the key defined in the peers section of
dundi.conf. When you look at the peer in iax.conf on the remote box,
there is no host entry and it uses dbsecret=dundi/secret, the
dundi.conf
priv = dundi-internal,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
[00:19:66:1C:78:D5] ; Dev Box
model = symmetric
host = 192.168.99.252
inkey = eus
outkey = eus
include = priv
permit = priv
qualify = yes
From iax.conf
[priv]
type=friend
dbsecret=dundi/secret
context=longdistance
Hope this helps, in your case Dundi will save you a world of work on
configuring that many systems, in fact if you structure Dundi like
spokes around a small number of master servers, the config gets real
easy.Let me know how it goes.
On Wed, Apr 16, 2008 at 8:41 AM, Jeremy Mann [EMAIL PROTECTED] wrote:
I'm a little confused with DUNDi and SIP as the backend channel type:
Dundi.conf:
[mappings]
priv = dundi-priv-local,0,SIP,[EMAIL PROTECTED],nopartial
Using the above, the dial string passed to the person on the other box is
SIP/[EMAIL PROTECTED]
How can you use authentication, along with SIP, along with specifying
extension?
My sip.conf has a friend defined:
[priv]
host=dynamic
secret=priv
disallow=all
allow=ulaw
canreinvite=no
nat=no
context=from-internal\
type=friend
I need to specify the sip channel to use the priv peer, priv secret, and
pass the extension. I've tried defining my mapping as:
Priv = dundi-priv-local,0,SIP,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
But obviously the console on the far end complains that peer
a.b.c.d/${NUMBER} cannot be found.
Thanks for any insight into this. I'd prefer not having to define a sip
peer per box(I have 25 connected in my dundi cloud), nor would I like to
enable anonymous SIP calls, as I have the ports open to the world for
inbound sip from bandwidth.com
This e-mail, facsimile, or letter and any files or attachments transmitted
with it contains information that is confidential and privileged. This
information is intended only for the use of the individual(s) and
entity(ies) to whom it is addressed. If you are the intended recipient,
further disclosures are prohibited without proper authorization. If you are
not the intended recipient, any disclosure, copying, printing, or use of
this information is strictly prohibited and possibly a violation of federal
or state law and regulations. If you have received this information in
error, please notify Texas Health Management Group immediately at
1-817-310-4999. Texas Health Management Group, its subsidiaries, and
affiliates hereby claim all applicable privileges related to this
information.
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--
*
Bruce Reeves, dCAp
EUS Networks
Office: 212-624-5943
Web: www.euscorp.com
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This e-mail, facsimile, or letter and any files or attachments transmitted
with it contains information that is confidential and privileged. This
information is intended only for the use of the individual(s) and entity(ies)
to whom it is addressed. If you are the intended recipient, further
disclosures are prohibited without proper authorization. If you are not the
intended recipient, any disclosure, copying, printing, or use of this
information is strictly prohibited and possibly a violation of federal or
state law and regulations. If you have received this information in error,
please notify Texas Health
Re: [asterisk-users] DUNDi and SIP
Try this,
[priv]
dbsecret=dundi/secret
disallow=all
allow=ulaw
canreinvite=no
nat=no
context=from-internal
type=friend
priv = dundi-priv-local,0,SIP,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
On Tue, Apr 22, 2008 at 8:23 AM, Jeremy Mann [EMAIL PROTECTED] wrote:
No.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves
Sent: Tuesday, April 22, 2008 6:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and SIP
Jeremy,
Did you get this working?
On Thu, Apr 17, 2008 at 7:38 AM, Jeremy Mann [EMAIL PROTECTED] wrote:
I have it working via IAX, when I try changing everything to SIP I can't
specify a username and an extension, so it becomes useless.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce
Reeves
Sent: Thursday, April 17, 2008 6:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and SIP
Jeremy,
Here is a working sample to compare to. This is an IAX2 setup, but the
only difference is in the mapping change IAX2 to SIP. Notice the 4th
setting in the mapping? It defines to use the IAX2 peer priv with
the secret generated of the key defined in the peers section of
dundi.conf. When you look at the peer in iax.conf on the remote box,
there is no host entry and it uses dbsecret=dundi/secret, the
dundi.conf
priv = dundi-internal,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
[00:19:66:1C:78:D5] ; Dev Box
model = symmetric
host = 192.168.99.252
inkey = eus
outkey = eus
include = priv
permit = priv
qualify = yes
From iax.conf
[priv]
type=friend
dbsecret=dundi/secret
context=longdistance
Hope this helps, in your case Dundi will save you a world of work on
configuring that many systems, in fact if you structure Dundi like
spokes around a small number of master servers, the config gets real
easy.Let me know how it goes.
On Wed, Apr 16, 2008 at 8:41 AM, Jeremy Mann [EMAIL PROTECTED] wrote:
I'm a little confused with DUNDi and SIP as the backend channel type:
Dundi.conf:
[mappings]
priv = dundi-priv-local,0,SIP,[EMAIL PROTECTED],nopartial
Using the above, the dial string passed to the person on the other box
is
SIP/[EMAIL PROTECTED]
How can you use authentication, along with SIP, along with specifying
extension?
My sip.conf has a friend defined:
[priv]
host=dynamic
secret=priv
disallow=all
allow=ulaw
canreinvite=no
nat=no
context=from-internal\
type=friend
I need to specify the sip channel to use the priv peer, priv secret, and
pass the extension. I've tried defining my mapping as:
Priv = dundi-priv-local,0,SIP,priv:[EMAIL
PROTECTED]/${NUMBER},nopartial
But obviously the console on the far end complains that peer
a.b.c.d/${NUMBER} cannot be found.
Thanks for any insight into this. I'd prefer not having to define a sip
peer per box(I have 25 connected in my dundi cloud), nor would I like to
enable anonymous SIP calls, as I have the ports open to the world for
inbound sip from bandwidth.com
This e-mail, facsimile, or letter and any files or attachments
transmitted
with it contains information that is confidential and privileged. This
information is intended only for the use of the individual(s) and
entity(ies) to whom it is addressed. If you are the intended recipient,
further disclosures are prohibited without proper authorization. If you
are
not the intended recipient, any disclosure, copying, printing, or use of
this information is strictly prohibited and possibly a violation of
federal
or state law and regulations. If you have received this information in
error, please notify Texas Health Management Group immediately at
1-817-310-4999. Texas Health Management Group, its subsidiaries, and
affiliates hereby claim all applicable privileges related to this
information.
___
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
*
Bruce Reeves, dCAp
EUS Networks
Office: 212-624-5943
Web: www.euscorp.com
Re: [asterisk-users] DUNDi and SIP
Jeremy,
Here is a working sample to compare to. This is an IAX2 setup, but the
only difference is in the mapping change IAX2 to SIP. Notice the 4th
setting in the mapping? It defines to use the IAX2 peer priv with
the secret generated of the key defined in the peers section of
dundi.conf. When you look at the peer in iax.conf on the remote box,
there is no host entry and it uses dbsecret=dundi/secret, the
dundi.conf
priv = dundi-internal,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
[00:19:66:1C:78:D5] ; Dev Box
model = symmetric
host = 192.168.99.252
inkey = eus
outkey = eus
include = priv
permit = priv
qualify = yes
From iax.conf
[priv]
type=friend
dbsecret=dundi/secret
context=longdistance
Hope this helps, in your case Dundi will save you a world of work on
configuring that many systems, in fact if you structure Dundi like
spokes around a small number of master servers, the config gets real
easy.Let me know how it goes.
On Wed, Apr 16, 2008 at 8:41 AM, Jeremy Mann [EMAIL PROTECTED] wrote:
I'm a little confused with DUNDi and SIP as the backend channel type:
Dundi.conf:
[mappings]
priv = dundi-priv-local,0,SIP,[EMAIL PROTECTED],nopartial
Using the above, the dial string passed to the person on the other box is
SIP/[EMAIL PROTECTED]
How can you use authentication, along with SIP, along with specifying
extension?
My sip.conf has a friend defined:
[priv]
host=dynamic
secret=priv
disallow=all
allow=ulaw
canreinvite=no
nat=no
context=from-internal\
type=friend
I need to specify the sip channel to use the priv peer, priv secret, and
pass the extension. I've tried defining my mapping as:
Priv = dundi-priv-local,0,SIP,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
But obviously the console on the far end complains that peer
a.b.c.d/${NUMBER} cannot be found.
Thanks for any insight into this. I'd prefer not having to define a sip
peer per box(I have 25 connected in my dundi cloud), nor would I like to
enable anonymous SIP calls, as I have the ports open to the world for
inbound sip from bandwidth.com
This e-mail, facsimile, or letter and any files or attachments transmitted
with it contains information that is confidential and privileged. This
information is intended only for the use of the individual(s) and
entity(ies) to whom it is addressed. If you are the intended recipient,
further disclosures are prohibited without proper authorization. If you are
not the intended recipient, any disclosure, copying, printing, or use of
this information is strictly prohibited and possibly a violation of federal
or state law and regulations. If you have received this information in
error, please notify Texas Health Management Group immediately at
1-817-310-4999. Texas Health Management Group, its subsidiaries, and
affiliates hereby claim all applicable privileges related to this
information.
___
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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--
*
Bruce Reeves, dCAp
EUS Networks
Office: 212-624-5943
Web: www.euscorp.com
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Re: [asterisk-users] need * consultant in houston area
What kind of help are you needing? On Mon, Mar 10, 2008 at 8:40 PM, A_ Navone [EMAIL PROTECTED] wrote: pls kindly respond to this email thx ! _ Connect and share in new ways with Windows Live. http://www.windowslive.com/share.html?ocid=TXT_TAGHM_Wave2_sharelife_012008 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- * Bruce Reeves, dCAp EUS Networks Office: 212-624-5943 Web: www.euscorp.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] replace astdb with a cluster-capable sql database engine
Vieri, What values are you looking to move from astdb? I have used realtime to store values for call features and other functions in the dial plan. I'm curious what you are looking to do. On Sat, Mar 8, 2008 at 12:01 PM, Vieri [EMAIL PROTECTED] wrote: I've been searching the Internet for information regarding the replacement of astdb with a modern sql engine. There are several reasons one would like to do this. First of all, external applications have a hard time reading/writing to the now-old astdb format. Also (and this is what interests me most), the sql astdb could easily be clustered throughout several servers (I'm looking for a master-master MySQL 2-server cluster solution). Asterisk has brought up Realtime which is very powerful but, correct me if I'm wrong, it still requires astdb internally. In other words, if I call Set(DB) in the dialplan then it will always be using astdb regardless of realtime. Some projects like Callweaver have forked from Asterisk 1.2 and replaced astdb with sqlite. I'm wondering if Asterisk has plans to allow the user to choose the astdb backend: standard db1, sqlite, MySQL (which I would use with nbcluster for my clustering purposes), Postgresql with Slony-II, PGcluster, etc. Or is it already possible? There has been some talk on this before: http://lists.digium.com/pipermail/asterisk-dev/2004-December/007846.html Also, the func_odbc feature seems to be very powerful: http://www.asteriskpbx.org/func_odbc but: 1) would there be potential issues with db handles on a very busy asterisk system after a relatively long run time? 2) would there be a way to map the odbc function(s) to the DB functions (Set(DB), read and write, DBdel, etc) so that rewriting the whole dialplan would not be necessary? (that's the whole point of defining a different astdb backend) If there are known problems/issues/projects/alternatives then please let me know. Thanks Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- * Bruce Reeves, dCAp EUS Networks Office: 212-624-5943 Web: www.euscorp.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] when we try to add CURL code to file channel.c we get an error - undefined reference to curl_easy_init
I may be asking the wrong question, but if you want to capture the
input and pass it to another process why not use the read application
and store the input in a variable? Could you not pass that variable
and use the curl function or an AGI to post it?
On Mon, Mar 3, 2008 at 11:05 PM, Prashant Sharma
[EMAIL PROTECTED] wrote:
Hi,
Thanks but using the logger.c approach will allow the IVR to receive the
digits in case 's' extension answers the call. That might result in the dial
plan dialing an extension or going to the 'i' extension and hanging up.
Ssorry about the confusion.
Thanks Regards
Prashant Sharma
On Mon, Mar 3, 2008 at 10:43 PM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Monday 03 March 2008 07:18, Prashant Sharma wrote:
I'm trying to make asterisk detect some DTMF digits during a call and
post
them (can't use WaitExten or Features.conf).
I would suggest that you implement that in logger.c and configure a line
to
send logs to an HTTP POST (via logger.conf), with the
pbx_substitute_variables_helper function, using the ${CURL()} function
directly. You may need to preload = func_curl.so in modules.conf, but
that will work well.
--
Tilghman
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Re: [asterisk-users] Switchvox feedback
Having recently worked with the latest version here are my thoughts. 1-3 The only out of the box method to configure and manage switchvox is the GUI. They are using realtime but manually configuring the system is not available by default. SSH access is not available by default and the root account information is not known unless you set it like Steve mentioned. 4 The interface is great for what Switchvox designed it to do. In my opinion this is a small office product. Administering the system is easy, initial configuration is not. There are several great things about it, the problem comes in when you expect it to do everything Asterisk can do. It has a feature set and a lot of limits in comparison to running Asterisk. Before selling it to a customer I would urge you to download the free version and configure it, I ran into several things the customer ask for that are easy to do in Asterisk dial plan and even in sip.conf settings that could not be done. On Mon, Mar 3, 2008 at 3:35 PM, C F [EMAIL PROTECTED] wrote: I have a customer that wants to get switchvox, since I have never used it, I would like to hear some feedback from active users of switchvox. In specific: 1. Does it use realtime or conf files 2. Is it possible to change it manually? 3. Is SSH access to login to console/shell available? 4. Are you or your customers happy with the user interface? TIA ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie dialplan: dial 0 for outside line
John, Try changing the entry in extensions.conf to Dial(Zap/g1/0). you need to specify what the dial command should send on the ZAP channel. On Tue, Mar 4, 2008 at 10:51 PM, Lee, John (Sydney) [EMAIL PROTECTED] wrote: I just managed to put in a TE410 card in an Asterisk box to work with OnRamp 20(E1 downunder). I am able to dial in but was not able to dial out. Can anyone offer me some advice please? In my extensions.conf, I just put in: [default] ... exten = 0,1,Dial(Zap/g1) and I get this on the console when I dialled 0. -- Executing [EMAIL PROTECTED]:1] Dial(SIP/5166-b76004f8, Zap/g1) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1LI -- Channel 0/1, span 1 got hangup, cause 100 -- Hungup 'Zap/1-1' [Mar 5 15:37:01] NOTICE[10479]: cdr.c:434 ast_cdr_free: CDR on channel 'Zap/1-1' not posted == Everyone is busy/congested at this time (1:0/0/1) == Auto fallthrough, channel 'SIP/5166-b76004f8' status is 'CHANUNAVAIL' The following is my set up: --- - # vi /etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-15 bchan=17-21 unused=22-31 dchan=16 loadzone=au defaultzone=au --- - [channels] language=en context=default switchtype=euroisdn pridialplan=unknown prilocaldialplan=unknown overlapdial=yes priindication=outofband signalling=pri_cpe rxwink=300 usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes restrictcid=no usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=-3.0 txgain=-6.0 callgroup=1 pickupgroup=1 immediate=no group=1 channel = 1-15 channel = 17-21 --- - *CLI pri show spans PRI span 1/0: Provisioned, Up, Active --- - *CLI zap show channels Chan Extension Context Language MOH Interpret pseudo default en default 1 default en default 2 default en default ... 14 default en default 15 default en default 17 default en default ... 20 default en default 21 default en default --- - # cat /proc/zaptel/1 Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 HDB3/CCS/CRC4 Timing slips: 556 1 TE4/0/1/1 Clear (In use) 2 TE4/0/1/2 Clear (In use) ... 15 TE4/0/1/15 Clear (In use) 16 TE4/0/1/16 HDLCFCS (In use) 17 TE4/0/1/17 Clear (In use) ... 21 TE4/0/1/21 Clear (In use) 22 TE4/0/1/22 Clear ... 31 TE4/0/1/31 Clear ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Followme
In the dialplan you would just add a prompt and ask the caller to press 1 to locate or hold for voice mail. If they press 1 launch the followme app. On Jan 22, 2008 10:25 AM, Anciso, Roy [EMAIL PROTECTED] wrote: I've been reading up on followme app for asterisk 1.4 and I have it working but I was wondering if the following was possible: Based on followme.conf present the caller with the option to locate the person: Call comes in (external or internal) and rings extension with followme configured. Before the followme app is initiated the caller is prompted to locate the person (by pressing 1 which initiates followme) or to continue onto voicemail. Thanks Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690 [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX and NAT Transparency
Use the qualify to help, but yes the soft phone registers and the system sends calls to it. On Jan 21, 2008 4:25 PM, bilal ghayyad [EMAIL PROTECTED] wrote: Hi Gordon; They are able to receive calls? Origination is not a problem I know, but what about receiving calls from the Asterisk to them? For example, how I can call the extension 200 that is behind NAT? (Assuming that extension 200 is registered on the Asterisk). Regards Bilal --- Hi All; Did anyone try to use IAX IP Phone behind NAT, and let it receive calls from Asterisk without doing port mapping at the router existed at the site where the IAX IP Phone existed? Is the need just to let the IAX IP Phone that is NATed to register on the Asterisk and at asterisk I set nat=yes for the IAX client configuration? Yes is the easy answer. I do this all the time fromn my laptop at friends/colleagues/other locations where I get a broadband connection, and don't ever fiddle with their routers. Or it is impossible to let the NATed IAX to receive calls without doing a port mapping at the router? Not impossible at all. You *may* find that some routers don't like it, but the majority of them are just fine. What about SIP, any luck? Same again, it just works. Not router fiddling required. I visit client sites with a small number of differnet SIP phones - plug them into their network and let them make calls, and unless they have weird routers with broken SIP ALGs or strict firewalling, it just works. We've been through this with you recently. Are you not getting list emails? What about google or the VoIP WiKi? There is plenty of stuff there all about it. I can easilly make SIP calls from SIP phones behind a NAT router to an asterisk box behind a different NAT router. It just works and there is a good page on the VoIP WiKi all about it. Go read this: http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions (Although I don't think that page is quite correct as I can make their '3' scenario just work ...) Gordon Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [HELP] Problems with VOIP organization
Yes Asterisk can receive the calls and based either on the line the call is on or some other method route the call to a destination. That being said there are 2 things to keep in mind, the hardware cost to setup 2 incoming lines and a analog port for the fax as well as phones may be high for a 2 line setup. The other thing to keep in mind is that faxing and asterisk is one of the more complicated task. There are so many things that can break faxing that setting this up is not for the faint hearted. On Dec 5, 2007 2:49 AM, Григорий Никоноров [EMAIL PROTECTED] wrote: Hello! Please help me with decision problem. I need to organize voip telephony in office. I have 2 phone lines(2 physical number) for phone and fax.I need to recive call on 1 phone then redirect it to neccessary phone or fax. Can Asterisk do that ? Thanks in advance ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IP Trunk and increasing volume level on diguim card
In zapata.conf you can add rxgain and txgain settings and use ztmonitor to get it set. There are some more details on this on voip-info.org. On Nov 29, 2007 1:49 AM, bilal ghayyad [EMAIL PROTECTED] wrote: Hi All; I have an IP Trunk established between Asterisk and the VoIP service provider, when call from my mobile to the PBX and then enter the destination number to call via the VoIP, I got a connection but the voice level volume need to be increased, I am trying to find if zaptel (diguim card) can increase the volume (if there is any command can do that)? And if that volume level is possible to be applied only for that IP Trunk and not for others. Any Help? Regards Bilal Get easy, one-click access to your favorites. Make Yahoo! your homepage. http://www.yahoo.com/r/hs ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with Polycom 320
Is there anything in the CLI about the sip peer? Can you show the settings you have in sip.conf and the phone setting you entered? Bruce Reeves On Nov 16, 2007 9:00 AM, Jarga Jallow [EMAIL PROTECTED] wrote: I am having trouble configuring my Polycom 320 IP phone. When I dial an extension it seems like am calling from outside. Also on the phone menu it says not registered. Does anybody know how to fix this? Thanks in advance Jarga Jallow Technical Support Engineer 2985 S. Hwy. 360 Grand Praire, Texas 75052 Direct: 972-206-1212 ext# 29 Mobile: 214-669-9046 Fax:972-999-4113 Toll Free: 1-877-801-5511 ext 34 Toll Free: 1-877-926-2288 www.2mcctv.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wifi - Managed access points...
Luis, Like Ron, I have tested deploying several different handsets and have been disappointed. I am currently testing a deployment with a DECT system by Aastra that uses multiple access points the talk SIP to Asterisk and DECT to the handset. Being based on DECT they have good battery life and handover of live calls between points is a key feature. Pricing is along the same as what I would pay for High end access points and good handsets. There are systems like this coming out from Aastra, Snom, Polycom/Kirk and probably some others. If I were going to deploy the setup you are talking about I would check this option out before jumping solely on wifi. If you want contact me off list and I'd be happy to visit in more detail. On 10/10/07, Luis Antonio Prata Barbosa [EMAIL PROTECTED] wrote: Hi, I'm working on a Wifi VoIP project specification. It will have almost 8 APs and 20-30 wifi phones. And after some research, I still having some questions ... 1) Are Managed Access Points (and switch controllers) really important to implement good wifi woip (w/ low latency and acceptable handover time) ? 2) What is the difference between (3com WX1200 + 3com AP 3750) and (DES-1228P + DWL-3140AP) ??? 3) 3com says their AP implements WMM ... and DLink says they priorize VoIP traffic based on VLAN ... are those methods the same ? Thank you, Luis A P Barbosa ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wifi - Managed access points...
Wai, The IP address is really on the access points, since they are the SIP part of the solution. Let me see how well I can explain this, The access points register to a manager application, running on one AP, and the phones have a hard coded DECT id and register to the same manager app. The manager actually performs the connection to the Asterisk system and all the access points have IP's and each phone an account on Asterisk. In a handover the manager app routes the SIP traffic to the AP that the handset is on and as the caller moves the phone detects other AP's and picks a new AP. That AP coordinates with the manager app to re-route the sip/rtp. In testing so far, you cannot tell the hand off occurred, even while watching the signal meters on the phone, there is no noticeable audio loss. There has to be a fair overlap in coverage, they say around -60db to -70db in signal you should have another AP and the phone can see up to 4 APs at a time. Each AP can handle 8 voice channels so you have to keep that in mind also. So did that make sense. == Little Commercial blip == Aastra requires people be certified resellers on this solution to purchase\sale it. In that process they give a fantastic amount of attention to planning a wireless deployment. Nortex is a certified reseller of the Aastra SPI-DECT solution. ==End of blip == On 10/10/07, Wai Wu [EMAIL PROTECTED] wrote: Hope you don't mind I jump in here. I am interested in DECT's handover of live calls. My question is, does the IP address on the phone change when moving from on access point to another? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves Sent: Wednesday, October 10, 2007 9:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Wifi - Managed access points... Luis, Like Ron, I have tested deploying several different handsets and have been disappointed. I am currently testing a deployment with a DECT system by Aastra that uses multiple access points the talk SIP to Asterisk and DECT to the handset. Being based on DECT they have good battery life and handover of live calls between points is a key feature. Pricing is along the same as what I would pay for High end access points and good handsets. There are systems like this coming out from Aastra, Snom, Polycom/Kirk and probably some others. If I were going to deploy the setup you are talking about I would check this option out before jumping solely on wifi. If you want contact me off list and I'd be happy to visit in more detail. On 10/10/07, Luis Antonio Prata Barbosa [EMAIL PROTECTED] wrote: Hi, I'm working on a Wifi VoIP project specification. It will have almost 8 APs and 20-30 wifi phones. And after some research, I still having some questions ... 1) Are Managed Access Points (and switch controllers) really important to implement good wifi woip (w/ low latency and acceptable handover time) ? 2) What is the difference between (3com WX1200 + 3com AP 3750) and (DES-1228P + DWL-3140AP) ??? 3) 3com says their AP implements WMM ... and DLink says they priorize VoIP traffic based on VLAN ... are those methods the same ? Thank you, Luis A P Barbosa ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to DUNDi branch office with area code?
DUNDi can be used in branch office and I have a similar setup to what your are referring with 11 sites. One thing that I decided to do, but did not have to is define site extensions like you did, but I use the 4 digit extension locally and via dundi. Here is the details: In my case I check for the 4 digit extension in the current site then do a look up on dundi. Each site send the request to 2 core systems that keep up with all the peers and forward the request on.Once the extension is found in the dundi context on a server the call gets routed to the correct site.By doing it this way I don't have to remember to dial a specific number for an intra site call. The other thing is I do I have each site a set of numbers, like 3100 - 3199 is site B and 3200 - 3299 is C, but with DUNDi I do not have to do it that way, it will find the extensions since I use regexten=whatever extension in my sip.conf for each phone. I hope that makes sense, JR has done an excellent job explaining DUNDi in several white papers, and I have used something from all of them. On 9/6/07, d tbsky [EMAIL PROTECTED] wrote: hi: i am new to asterisk and dundi. we have some branch office which will use asterisk in the future. they will form a full-mesh structure so every site can contact each other directly. i want to try setup dundi, then we don't need to modify every pbx when a new site add in the cloud. thanks to the great dundi document caveman can do it and other resource in the voip-info.org. i learn the basic setup of dundi. but i want to a little advanced setup with area code. like this: site HQ: has extension 101,102,103, and site HQ has area code 99 site A: has extension 101,102,103, and site A has area code 01 site B: has extension 101,102,103 and site B has area code 02 site C: has extension 101,102,103 and site C has area code 03 we want to use 4 as prefix to call to the internal cloud. so user at site A can call 4-99-101 to contact extension 101 at HQ. site B can call 4-03-102 to contact extension 102 at site C. now i m confused about this structure with DUNDi. i don't know the best way to setup DUNDi for this structure. i think maybe i should do below when user call 4-99-101 at site A : 1. site A ask for dundi request 4-99-101 to site HQ 2. site HQ strip 4-99 and look up 101 at local context 3. site HQ return the destination to site A 4. site A use the destination to call extension 101 at site HQ i don't know if step 23 is possible in dundi.conf. the example in the internet didn't tell how to do this. or there are better/standard ways to do this? thanks a lot for any suggestion!! Regards, tbskyd ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Overhead paging over IP...
Most of the overhead paging systems I have worked with had an CO input, which worked with the FXO port on an ATA. A couple brands had multiple source options, it is worth checking, I had problems with poor audio quality using the sound card with asterisk. On 9/4/07, Carlos Chavez [EMAIL PROTECTED] wrote: I have a customer that has two buildings that are connected with a fiber link. We have a single Asterisk server to cover both buildings. Now the customer went and bought an overhead paging system for the remote building and they want to integrate it with Asterisk. Is there a device that can connect over IP or an ATA that has an audio output port? The buildings are about 500 meters apart so we cannot run a cable from one building to the other just for audio. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Distributed System
Realtime and DUNDi covers all the bases. On 8/28/07, Peder @ NetworkOblivion [EMAIL PROTECTED] wrote: The question I always have when someone mentions distributing the load across multiple machines is how do you handle contexts for phones on different machines? I want all of my phones to dial into [companyA-phones]. I have to define it in two different places (or more depending on the number of boxes). Also, say I have a single company and I want a single auto attendant with dial by name? If users go to two different boxes, then voicemail dial by name will break because voicemail won't check both boxes for the name. Also, what about dialing a peer. Say all of my phones are 2xx. If I am 201 and I dial 202, how is my dialplan setup so that it knows that 202 is on box 2, versus box 1 where I am registered? I think having several boxes works fine if you are doing home user type stuff where you don't have lots of users within one context, but if you have offices with several people, I just see lots of potential issues. I could be wrong, but I've never been able to figure out a way around it. Brian West wrote: On Aug 28, 2007, at 10:14 AM, Seysan wrote: Hi all, I'm kind a New to Asterisk.But I'm a Network Administrator with 5 years of experiance. I want to know for an installation with 90 clients, If I don't want to have just 1 server for it, then how is it possible to distribute it among about 3 servers. Should I do it in a cluster (kernel level) or something with SER? I would recommend SER plus Asterisk. I have had great success with using Asterisk with OpenSER. Best Regards, Seysan /b ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stereo Conferences?
The codec is G722 I believe. and Polycom has a conference speaker phone with a subwoofer option that has HD voice. On 8/27/07, Matthew Rubenstein [EMAIL PROTECTED] wrote: Do any softphones run the HD codec? What exactly is the HD codec technically called, and is there any info about its codec running inside Asterisk? On Mon, 2007-08-27 at 08:47 -0400, C F wrote: although not stereo i believe its the closest you will get if the codec is supported by asterisk. polycom has now HD codec On 8/27/07, Matthew Rubenstein [EMAIL PROTECTED] wrote: Are there any speakerphones or other conferencing HW phones that play the audio in stereo? Either their own speakers, or jacks for an amp with room speakers? Is there any way for Asterisk to deliver call legs with stereo channels in the RTP stream? If not, is it possible for Asterisk to keep 2 separate calls, or pairs of legs in a conference call, synced exactly enough (including traveling over the Net between the same 2 IP#s) for them to arrive as a stereo pair at the endpoint? -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: DELL Platforms
I have used both the powedge line for large deployments and the Optiplex N series for small offices. The only thing I have had to add to the pc's is 12 power extensions at times and here lately I have had a pc or 2 without the 4 pin molex connector so I had to find SATA to molex adapters. On 8/27/07, Arthur Miller [EMAIL PROTECTED] wrote: Hello list, I have a customer who is interested in standardizing on dell servers for asterisk deployments. Has anyone had success with a particular configuration? Anything specifically to watch out for? Thank you for your time, Art Arthur Miller Sr. Sales Associate VoIP Supply, LLC. 454 Sonwil Drive Buffalo, NY 14225 716-250-3871 OFFICE 716-630-1548 FAX [EMAIL PROTECTED] NOTICE: The information contained in this email and any document attached hereto is intended only for the named recipient(s). It is the property of the VoIP Supply, LLC and shall not be used, disclosed or reproduced without the express written consent of VoIP Supply, LLC. If you are not the intended recipient, nor the employee or agent responsible for delivering this message in confidence to the intended recipient(s), you are hereby notified that you have received this transmittal in error, and any review, dissemination, distribution or copying of this transmittal or its attachments is strictly prohibited. If you have received this transmittal and/or attachments in error, please notify me immediately by reply e-mail or telephone and then delete this message, including any attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY 14225 USA. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stable-Stable Asterisk
While it is not exactly running a huge system, I have had one 1.4 system running in a small office of 10 phones since June with no problems and another small system for about a month with no problems. I have also had a larger system (80+ phones, DUNDi and IAX trunking to 11 sites) running 1.4 for a over a month. That system has had stability issues from time to time with the IAX, I account most of the issues I have had to the changes being made and the fact that 90% of the systems it interacts with are 1.2 versions. I know there are bugs in 1.4, as are there bugs in 1.2 and likely even in 1.0. I did not move to 1.4 to avoid bugs or fix anything, but to use certain features to accomplish goals that the client had for the system. I think Tzafrir is right: --- Suppose you are a reader of a specific mailing list. Someone asked which is better: 1.2 or 1.4. Naturally the sample size you get is very small: only a handful of the large body of Asterisk users actually naswered it. So I am answering as someone using 1.4. Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Seeking opinions: Polycom IP330 phones?
I have deployed a couple 330's and they will use the same provisioning methods as the earlier models from Polycom, you just need to be sure and have the firmawre and configs for that version. The other thing I ran into, is the 330 did not ship with a power supply so you either go POE or buy the adapter in 5 packs. That may have changed, I order them the month they began shipping. I have not heard any complaints about them, but I deployed them in copy rooms and such not at anyone's desk. On 8/15/07, Michael Graves [EMAIL PROTECTED] wrote: Does anyone online have an opinion on these? I've used 500/510/6001/601 models before. Need to know if these apparently lesser models can be provisioned in the same way. Are end uers happy with them? Thanks, Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 c713-201-1262 skype mjgraves fwd 54245 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Allison Smith?
That happened last year. I remember getting emails about the site being discontinued and the prompts being added to digiums store. On 8/9/07, SIP [EMAIL PROTECTED] wrote: So I see: /*Note:* The site, TheVoice.digium.com and its credit system for purchasing voice prompts, has been discontinued. For customers who have outstanding credits through the site, please contact Customer Service http://www.digium.com/en/company/contact.php to receive a refund. / To me, that indicates less a cessation of contract with Digium/Allison and more a modification of the way things are handled. But who knows. N. Matt wrote: She's sort of on the website... click 'Purchase and Price', then 'Buy Online', You will see there is no place to purchase it. On 8/9/07, SIP [EMAIL PROTECTED] wrote: Steve Totaro wrote: Matt wrote: Did I miss something? I see Digium no longer contracts with Allison to record IVR prompts, was there a falling out? Where do you see that? Thanks, Steve http://www.digium.com/en/products/voice/ She's still on the website. N. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best and easiest soft phone for my Dad..
One I really like is the idefsk version that was a zip file, you could extract the file configure the softphone, zip it up and email it out. Saved the headache of walking someone through the process and even ran of thumb drives. On 7/21/07, WipeOut [EMAIL PROTECTED] wrote: Hi, Here is the situation.. My Dad is working on contract in overseas.. He has internet access in his hotel.. He wants to be able to talk to my Mum but the calls are expensive.. I have an asterisk box setup for my business and it has a public IP etc.. My Mum has access to a working phone extension on this box.. I got my Dad to install X-Lite but for some reason it won't register and trying to talk him through working out whats wrong is proving to be difficult.. Also I haven't used a softphone in years.. It could be the NAT in the hotel, it could be a firewall or any number of things that can cause these issues.. It could even be X-Lite or something running on his PC.. So I am looking for a softphone thats really simple to setup and as foolproof as possible.. If SIP is likely to be problematic to setup then I have no problem getting him to use IAX but will need suggestions of which IAX softphone to use and also how to configure it in the iax.conf (haven't done this before).. Any suggestions welcome and appreciated.. Thanks.. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] gui for conferencing
Check out this gui for meetme. http://sourceforge.net/projects/web-meetme/ On 7/14/07, Eric Smith [EMAIL PROTECTED] wrote: Is there something simple like gastman that provides functionality to establishing conferencing? -- Eric Smith ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] video calls - Windows / Linux interoperability ?
For windows you can't really go wrong with X-lite 3.0, the free version of eye-beam. http://www.counterpath.com/index.php?menu=Productssmenu=xlite On 7/10/07, Florin Andrei [EMAIL PROTECTED] wrote: I will install Asterisk on my home server, I want to be able to route video calls, but I need the Windows and Linux clients to be interoperable. On Linux, it looks like Ekiga is a good candidate. But how about Windows? Anyone using Kapanga in an Asterisk network that includes Ekiga? Are these two interoperable? I'm not necessarily looking for open source software, free for personal use is enough. -- Florin Andrei http://florin.myip.org/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Suing Dell||Dull Computers for CID abuse
Being Independence Day and all Oh, so anyway, who was guy Eng you named the country after? And who was America named after ? Steve The name was derived from the Latinized http://en.wikipedia.org/wiki/Latinversion of the explorer Amerigo Vespucci http://en.wikipedia.org/wiki/Amerigo_Vespucci's name, *Americus Vespucius*, in its feminine form, *America*, as the other continents all have Latin feminine names. Wikipedia and a memory for things historical. Have a good 4th. Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] v1.4.x ready yet?
While I have not jumped all my systems to 1.4, there were some that I have moved to 1.4 and I have found it to be as stable as 1.2 was on those machines.One of the systems is a 10 user office with Sangoma cards and another is a 70+ user pure voip system. In both cases I have warning messages about my dialplan usage of realtime and the fact that it will be depreciated in the next release, but everything works as it should and the upgrades.txt guided me through the changes to my dialplan. Hope that helps. On 6/29/07, shadowym [EMAIL PROTECTED] wrote: Hi All, Eagerly waiting for v1.4.x to mature a bit before getting serious about it. Is it ready for production yet? If that's too general, where is it in terms of stability compared to where 1.2.x is now. Anyone running it successfully in production environment and if so what sort of config do you have? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ring/Off-hook in strange state 6
I have seen this on cards waiting for the callerID and there being a problem with the callerid signal. Is callerid working on theses lines? On 6/26/07, Alex Mcdowell [EMAIL PROTECTED] wrote: Can anybody at least point me in a direction?? On 6/25/07, Alex Mcdowell [EMAIL PROTECTED] wrote: I don't think my cards are bad, but maybe there is a problem with the one. It has been two weeks since I put my ticket in with Digium...and still no word. I am starting to get frustrated. On 6/22/07, Daniel Hazelbaker [EMAIL PROTECTED] wrote: Alex, I had this problem with a new TDM2400 card that we purchased. Specifically I would get that message and then it would pick up the ringing line AND the line next to it. Basically, lines 1 2 had been cross-linked somehow. After a few weeks of trouble-shooting with Digium tech support they cross-shipped me a new card and the problem (and that message) went away. Daniel Hazelbaker High Desert Church On Jun 22, 2007, at 1:22 PM, Alex Mcdowell wrote: HI I have two servers both of which get this message on one of the lines. Ring/Off-hook in strange state 6. The one server seems to be ok with it, but the other one when an extension picks up there is no one there and the incoming call keeps ringing. I tried to adjust the levels in wcfxo.clike someone had suggested, but it didn't do anything. I also upgraded zaptel to the latest. 1.2.18 and asterisk is running at 1.2.14. callprogess is set to no, as well as busydetect=no. This is a major problem since this box only has 1 other line, but at least it works. I can't seem to find much info on this issue. I can't believe others haven't run into it. I started a ticket with digium, but I guess they are pretty backed up. Here is what I am getting in the CLI: Thanks for any help -Alex -- Starting simple switch on 'Zap/4-1' -- Executing Wait(Zap/4-1, 1) in new stack -- Executing Answer(Zap/4-1, ) in new stack -- Executing Dial(Zap/4-1, SIP/4125|20|tr) in new stack -- Called 4125 Jun 22 13:44:45 WARNING[24296]: chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 4 -- SIP/4125-09559118 is ringing Jun 22 13:44:46 WARNING[24296]: chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 4 -- SIP/4125-09559118 answered Zap/4-1 == Spawn extension (incoming, s, 3) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' -- Starting simple switch on 'Zap/4-1' -- Executing Wait(Zap/4-1, 1) in new stack -- Executing Answer(Zap/4-1, ) in new stack -- Executing Dial(Zap/4-1, SIP/4125|20|tr) in new stack -- Called 4125 Jun 22 13:44:57 WARNING[24303]: chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 4 -- SIP/4125-09559118 is ringing Jun 22 13:44:58 WARNING[24303]: chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 4 -- SIP/4125-09559118 answered Zap/4-1 == Spawn extension (incoming, s, 3) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' -- Starting simple switch on 'Zap/4-1' -- Executing Wait(Zap/4-1, 1) in new stack -- Executing Answer(Zap/4-1, ) in new stack -- Executing Dial(Zap/4-1, SIP/4125|20|tr) in new stack -- Called 4125 -- SIP/4125-09559118 is ringing ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4 Zaptel/Sangoma Issues on CentOS
Which CentOS version? You might try, if you have not already the beta wanpipe drivers, they have: Support for 2.6.20 kernels which include CentOS 5 On 6/6/07, Steve Totaro [EMAIL PROTECTED] wrote: Any ideas? Sangoma support is closed for the evening. I have the latest Sangoma drivers and Asterisk 1.4 everything installed. When I fire up asterisk, I keep getting Primary D-Channel on span 1 up repeated over and over. The B channels never come up. There are no errors in any of the logs, zttool, or the wanpipe tools. Intense pri debug output: Unnumbered frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 M3: 3 P/F: 1 M2: 3 11: 3 [ SABME (set asynchronous balanced mode extended ) ] 0 bytes of data -- Got SABME from network peer. Sending Unnumbered Acknowledgement [ 02 01 73 ] Unnumbered frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 M3: 3 P/F: 1 M2: 0 11: 3 [ UA (unnumbered acknowledgement) ] 0 bytes of data -- Restarting T203 counter q921.c:709 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED q921.c:664 q921_dchannel_up: q921_state now is Q921_LINK_CONNECTION_ESTABLISHED -- Restarting T203 counter == Primary D-Channel on span 1 up pri intense debug span 1 [ 02 01 7f ] Unnumbered frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 M3: 3 P/F: 1 M2: 3 11: 3 [ SABME (set asynchronous balanced mode extended ) ] 0 bytes of data -- Got SABME from network peer. Sending Unnumbered Acknowledgement [ 02 01 73 ] Unnumbered frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 M3: 3 P/F: 1 M2: 0 11: 3 [ UA (unnumbered acknowledgement) ] 0 bytes of data -- Restarting T203 counter q921.c:709 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED q921.c:664 q921_dchannel_up: q921_state now is Q921_LINK_CONNECTION_ESTABLISHED -- Restarting T203 counter == Primary D-Channel on span 1 up [ 02 01 7f ] Unnumbered frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 M3: 3 P/F: 1 M2: 3 11: 3 [ SABME (set asynchronous balanced mode extended ) ] 0 bytes of data -- Got SABME from network peer. Sending Unnumbered Acknowledgement [ 02 01 73 ] Unnumbered frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 M3: 3 P/F: 1 M2: 0 11: 3 [ UA (unnumbered acknowledgement) ] 0 bytes of data -- Restarting T203 counter q921.c:709 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED q921.c:664 q921_dchannel_up: q921_state now is Q921_LINK_CONNECTION_ESTABLISHED -- Restarting T203 counter == Primary D-Channel on span 1 up Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4 Zaptel/Sangoma Issues on CentOS
Which CentOS version? You might try, if you have not already the beta wanpipe drivers, they have: Support for 2.6.20 kernel s which include CentOS 5 On 6/6/07, Steve Totaro [EMAIL PROTECTED] wrote: Any ideas? Sangoma support is closed for the evening. I have the latest Sangoma drivers and Asterisk 1.4 everything installed. When I fire up asterisk, I keep getting Primary D-Channel on span 1 up repeated over and over. The B channels never come up. There are no errors in any of the logs, zttool, or the wanpipe tools. Intense pri debug output: Unnumbered frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 M3: 3 P/F: 1 M2: 3 11: 3 [ SABME (set asynchronous balanced mode extended ) ] 0 bytes of data -- Got SABME from network peer. Sending Unnumbered Acknowledgement [ 02 01 73 ] Unnumbered frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 M3: 3 P/F: 1 M2: 0 11: 3 [ UA (unnumbered acknowledgement) ] 0 bytes of data -- Restarting T203 counter q921.c:709 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED q921.c:664 q921_dchannel_up: q921_state now is Q921_LINK_CONNECTION_ESTABLISHED -- Restarting T203 counter == Primary D-Channel on span 1 up pri intense debug span 1 [ 02 01 7f ] Unnumbered frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 M3: 3 P/F: 1 M2: 3 11: 3 [ SABME (set asynchronous balanced mode extended ) ] 0 bytes of data -- Got SABME from network peer. Sending Unnumbered Acknowledgement [ 02 01 73 ] Unnumbered frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 M3: 3 P/F: 1 M2: 0 11: 3 [ UA (unnumbered acknowledgement) ] 0 bytes of data -- Restarting T203 counter q921.c:709 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED q921.c:664 q921_dchannel_up: q921_state now is Q921_LINK_CONNECTION_ESTABLISHED -- Restarting T203 counter == Primary D-Channel on span 1 up [ 02 01 7f ] Unnumbered frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 M3: 3 P/F: 1 M2: 3 11: 3 [ SABME (set asynchronous balanced mode extended ) ] 0 bytes of data -- Got SABME from network peer. Sending Unnumbered Acknowledgement [ 02 01 73 ] Unnumbered frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 M3: 3 P/F: 1 M2: 0 11: 3 [ UA (unnumbered acknowledgement) ] 0 bytes of data -- Restarting T203 counter q921.c:709 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED q921.c:664 q921_dchannel_up: q921_state now is Q921_LINK_CONNECTION_ESTABLISHED -- Restarting T203 counter == Primary D-Channel on span 1 up Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk with Multiple Network Interfaces
I have a box doing this, Asterisk listens on either IP unless you bind to a specific interface. On 5/25/07, Douglas Garstang [EMAIL PROTECTED] wrote: I have a scenario here with IP phones, on a private 192.168 network connecting to an Asterisk box, also on the same 192.168 private network. We'd like to have the Asterisk box also be able to send traffic to the public IP space. For this, we would need to multi-home the box, and put two network cards in it, with two IP addresses, one on each network. I know from past experience that Asterisk only listens on the first interface, or a single one if specified. I imagine this will cause all sorts of problems with a multi homed approach. Has anyone gotten around this? Thanks, Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Time Card
This can be accomplished by writing an IVR to prompt and then using AGI or dialplan commands the query strings can be executed. I have a setup like this for a inegrating a in house time keeping system with asterisk. On 5/24/07, Nitesh Divecha [EMAIL PROTECTED] wrote: Thanks for your reply, The basic system would work as follows: - Method 1 === An employee would call in to the system and a welcome message is prompted. After that a employee is asked to enter the employee ID and PIN number and once verified Employee ID, Caller ID, and time of day is stored into MySQL DB. By end of the day employee will call in again to logout from the system and same information is stored into the DB. Method 2 === This time employee is verified with Caller ID, so the employee ID and PIN number is skipped and time of day is logged into the DB. Is it possible? Thanks, Nitesh ram wrote: On 5/24/07, *Nitesh Divecha* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hello All, I have been looking for this solution for quite sometimes Asterisk Time Card System. I found some discussion from Digium forum but not quite helpful. Hi what is the mean of time card system ? is this kind of attendent system ? kindly give some more details ram ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: DUNDi configuration problem
that the century was going to end. -- Douglas Adams ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: DUNDi configuration problem
Tim,
When you are checking for 5000 are you typing dundi lookup [EMAIL PROTECTED]
On 5/18/07, Tim Verscheure [EMAIL PROTECTED] wrote:
Thank you for the quick response. Do I need to create a route to the
other machine? like a trunk?
greetz, Tim
2007/5/17, JR Richardson [EMAIL PROTECTED]:
[mappings]
priv = dundi-priv-canonical,0,SIP,${IPADDR}/${NUMBER},nopartial
priv = dundi-priv-customers,100,SIP,${IPADDR}/${NUMBER},nopartial
priv = dundi-priv-via-pstn,400,SIP,${IPADDR}/${NUMBER},nopartial
Your mappings are wrong, this is for IAX, for SIP to work, it should be:
priv = dundi-priv-canonical,0,SIP,${NUMBER}@the real IP
Address,nopartial
The rest looked ok I think.
Good luck.
JR
--
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Engineering for the Masses
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Re: [asterisk-users] Microsoft's Move Into IP PBX Market
How sad, cnet misspelled Polycom and Cisco didn't make the cut. On 5/16/07, George Pajari [EMAIL PROTECTED] wrote: From c|net News: On Monday,Microsoft and nine leading phone manufacturers--Asustek Computer, GN, LG-Nortel, NEC, Plantronics, Plycom, Samsung, Tatung, and Vitelix--announced the public beta program for Microsoft Office Communications Server 2007 and Microsoft Office Communicator 2007. http://news.com.com/8301-10784_3-9719931-7.html?part=rsssubj=newstag=2547-1_3-0-20 -- George Pajari (dCAP), netVOICE communications 604 484 VOIP(8647) x102 www.netvoice.ca www.ip-centrex.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom headset button blinking
There is a mode which sets the default mode to Headset when you press answer on incoming calls or dial o outgoing calls. The flashing icon indicates this mode is active. On 5/10/07, James Fromm [EMAIL PROTECTED] wrote: Does anyone know what it means when the headset button on Polycom phones is blinking? The blinking state is achieved by hitting the button twice while on-hook. First press activates the headset circuit and takes the phone off-hook. Second press deactivates the headset circuit, puts the phone on-hook, and starts the headset button blinking. Third press stops the headset button blinking. Any ideas? Thanks, James ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The purpose of DUNDi
I use DUNDi in this way, I have several remote sites and a MPLS network connecting the sites. I have each sites asterisk box looking at 2 DUNDi peers and those 2 central peers can query all sites. I don't have a lot of phones or people moving between sites, but I did not want to have to setup a IAX connection for every site on every server. I like the ability for DUNDi to determine which server to talk to and then configure the dial string for that call. This made my configuration easier to expand as I deployed new sites. I simply added the new peer to my central servers and configured the new site server and I could call between sites. While DUNDi's original intent was more for least cost routing or zero cost routing, I think it provides an excellent means to scale a network of asterisk systems. On 5/9/07, Ronaldo [EMAIL PROTECTED] wrote: Hi all, I'm planning to deploy many Asterisk servers for remote sites connected through IAX. Behind each server, there will be many sip clients connected. A sip client from one site must be able to make calls for the other sip clients connected to the other remote Asterisk servers. I've heard that DUNDi is a good option in order for each Asterisk server to locate the right (or the best) routes for the sip clients. Is DUNDi really used for that? Thanks in advance ... Ronaldo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The purpose of DUNDi
I use DUNDi in this way, I have several remote sites and a MPLS network connecting the sites. I have each sites asterisk box looking at 2 DUNDi peers and those 2 central peers can query all sites. I don't have a lot of phones or people moving between sites, but I did not want to have to setup a IAX connection for every site on every server. I like the ability for DUNDi to determine which server to talk to and then configure the dial string for that call. This made my configuration easier to expand as I deployed new sites. I simply added the new peer to my central servers and configured the new site server and I could call between sites. While DUNDi's original intent was more for least cost routing or zero cost routing, I think it provides an excellent means to scale a network of asterisk systems. On 5/9/07, Ronaldo [EMAIL PROTECTED] wrote: Hi all, I'm planning to deploy many Asterisk servers for remote sites connected through IAX. Behind each server, there will be many sip clients connected. A sip client from one site must be able to make calls for the other sip clients connected to the other remote Asterisk servers. I've heard that DUNDi is a good option in order for each Asterisk server to locate the right (or the best) routes for the sip clients. Is DUNDi really used for that? Thanks in advance ... Ronaldo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The purpose of DUNDi
There are nine sites, 10 servers. While it is not a huge deployment by some standards, it was simplified with DUNDi. On 5/9/07, Olivier [EMAIL PROTECTED] wrote: Just the sake of curiosity, how many sites (or user) did you interconnect using DUNDi ? Regards 2007/5/9, Bruce Reeves [EMAIL PROTECTED] : I use DUNDi in this way, I have several remote sites and a MPLS network connecting the sites. I have each sites asterisk box looking at 2 DUNDi peers and those 2 central peers can query all sites. I don't have a lot of phones or people moving between sites, but I did not want to have to setup a IAX connection for every site on every server. I like the ability for DUNDi to determine which server to talk to and then configure the dial string for that call. This made my configuration easier to expand as I deployed new sites. I simply added the new peer to my central servers and configured the new site server and I could call between sites. While DUNDi's original intent was more for least cost routing or zero cost routing, I think it provides an excellent means to scale a network of asterisk systems. On 5/9/07, Ronaldo [EMAIL PROTECTED] wrote: Hi all, I'm planning to deploy many Asterisk servers for remote sites connected through IAX. Behind each server, there will be many sip clients connected. A sip client from one site must be able to make calls for the other sip clients connected to the other remote Asterisk servers. I've heard that DUNDi is a good option in order for each Asterisk server to locate the right (or the best) routes for the sip clients. Is DUNDi really used for that? Thanks in advance ... Ronaldo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The purpose of DUNDi
Alex, Thanks for the linking to JR's article. That was my source for setting up DUNDi also. On 5/9/07, Alex Robar [EMAIL PROTECTED] wrote: Hi Ronaldo, Yes, you can use DUNDi for this. DUNDi simply advertises routes that a given server can terminate to its peers. As a very simple example, if ServerA houses extensions 500 through 599 and ServerB houses extensions 600 through 699, ServerA would advertise that it can terminate 5XX, and ServerB would advertise that it can terminate 6XX. When any peer in your DUNDi cloud requests how to terminate extension 502, ServerA will return a route to itself that will allow that call to be made. There's a nice article on the Texas AUG site about setting up DUNDi with dynamic extensions ( http://txaug.net/storage/users/3/3/images/17/Using%20DUNDi%20with%20a%20Cluster%20of%20Asterisk%20Servers.pdf ). Cheers, Alex Robar On 5/9/07, Ronaldo [EMAIL PROTECTED] wrote: Hi all, I'm planning to deploy many Asterisk servers for remote sites connected through IAX. Behind each server, there will be many sip clients connected. A sip client from one site must be able to make calls for the other sip clients connected to the other remote Asterisk servers. I've heard that DUNDi is a good option in order for each Asterisk server to locate the right (or the best) routes for the sip clients. Is DUNDi really used for that? Thanks in advance ... Ronaldo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX Trunk
Yes it is. On 5/3/07, Ronaldo [EMAIL PROTECTED] wrote: Hi all, Is it possible to have something like this: SoftPhone -(SIP)- Asterisk -(IAX trunk)- Asterisk -(SIP)- SoftPhone I want a IAX trunk between two asterisks and on each tip I have SIP clients that need to talk to each other. Thansk. Ronaldo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX Trunk
Yes it is On 5/3/07, Ronaldo [EMAIL PROTECTED] wrote: Hi all, Is it possible to have something like this: SoftPhone -(SIP)- Asterisk -(IAX trunk)- Asterisk -(SIP)- SoftPhone I want a IAX trunk between two asterisks and on each tip I have SIP clients that need to talk to each other. Thansk. Ronaldo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX Trunk
The wiki has a decent page about it. http://www.voip-info.org/wiki-IAX What you are trying to setup sounds simple enoug, you mainly will have an extension or pattern match that executes a dial command from box A to box B and passes the remote extension. On 5/3/07, Ronaldo [EMAIL PROTECTED] wrote: Hi Bruce, Can you suggest me any documentation about using IAX truking? Thank you. Ronaldo. Bruce Reeves wrote: Yes it is. On 5/3/07, *Ronaldo* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi all, Is it possible to have something like this: SoftPhone -(SIP)- Asterisk -(IAX trunk)- Asterisk -(SIP)- SoftPhone I want a IAX trunk between two asterisks and on each tip I have SIP clients that need to talk to each other. Thansk. Ronaldo ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk-Polycom HELLLLPPP!!!!
Jim, What happens in your first senario is an attended transfer, after User1 and 3 have initiadted their call, User1 should press transfer again to complete the transfer. At which point User1 will be disconnected and Users 2 3 will talk. The second issue is the limit of digits and is likely due to a very short timeout in features.conf, check the entry transferdigittimeout. On 5/3/07, Jim Suber [EMAIL PROTECTED] wrote: PBX: Asterisk 1.4 Phones: PSTN phone connected to TDM400 X-Ten Lite Polycom 430 Scenario Polycom 430 = User1 User2 calls User1(Polycom 430) asks to be transfered to User3 User1 does an attended transfer using the trnsfr button on the polycom User2 is placed in music-on-hold User3s phone rings. (So far so good Right?) User3 picks up the phone to answer User2 only to find that he is talking to User1 User2 is stuck in music-on-hold. FOREVER! The other two phones work exactly as they should using the # key Using the # key on the Polycom only allow the dialing of 1 number before Alice announces That there is no such extension. HELP Thanks in advance Jim ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] is dundi worth pursuing in this situation?
Erik, Your setup is very similar to one of my own, and I started of manually configuring it, creating IAX connections for each site and then using dial plan to route the call. When I looked at Dundi and finally got it working, I have one IAX connection for all sites and the connections are dynamically created. My dial plan also got simpler, as I add sites I add them to Dundi and the dial plan routes all unmatched extensions to Dundi for lookup. For me dundi has reduced the complexity of my dial plan and I have a pair of servers that query everybody and the that pair listed at my remote sites. I am not using it for least cost routing, yet, but so far it has made things a little easier. You might take a look at the article on txaug.net, under hubguru's articles, it is from JR's Astricon 2006 session. On 5/1/07, Erik Anderson [EMAIL PROTECTED] wrote: At work, I have 4 branch offices at which I've deployed asterisk. Call termination/origination at each branch office is handled either through a frac PRI or 3rd party SIP provider. Soon, I'll be replacing the legacy PBX at our HQ with asterisk. Each branch office has between 3 and 20 employees, each with their own extension and DID, and at headquarters, we have about 70 people, again each with their own extensions and DID. Handling local and LD calls from all the offices isn't a big deal - just normal call routing for that. My main question is what to do with calls between the offices. Each branch is connected back to HQ with a persistant VPN tunnel - I've tested IAX2 traffic over these tunnels before, and things work great. Since this works fairly well, I envision using IAX trunks for all intra-office calls. So - in this situation, would it be easier to just manage the office dialplan(s) and call routing manually, or would it be worth it to set up dundi for extension discovery? Thanks! -- Erik Anderson http://andersonfam.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Capture Asterisk traffic
The RTP traffic is not going to be on port 5060, that is the sip only. Check your rtp.conf file in asterisk for the port range used for RTP traffic. On 5/1/07, CSB [EMAIL PROTECTED] wrote: I want to capture all my Asterisk traffic (including RTP) and then analyse it. My plan was to use tcpdump and then analyse with Wireshark. The following works: tcpdump -i eth0 -s 0 -w /tmp/tcpdump.1 But I want to be a bit more selective: tcpdump -C 100 -W 10 -w /tmp/tcpdump -i eth1 -s 0 udp and dst port = 5060 This doesn't capture the RTP traffic. Could anyone advise what I'm doing wrong or suggest a better way? Thanks Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SugarCRM, NO!, Foxpro, SI?
I have heard of people rejecting Sugar for their existing CRM/ERP product based on VS Foxpro. I'm not a huge fan of Foxpro myself, but if the system already exist then a lot of people see little advantage in changing. On 4/30/07, Paul [EMAIL PROTECTED] wrote: Joe acquisto wrote: Paul [EMAIL PROTECTED] Wrote: 4/30/2007 8:53 AM: Joe acquisto wrote: I have dual posted this to the user and biz lists. Has anyone ever heard of someone running an Asterisk based system, yet Has abandoning SugarCRM, and opting to develop their own Visual FoxPro Has database/CRM? Please don't dump on me now, this is not my idea, I am just asking for Please comments, to see if my own initial thoughts are reasonably accurate. I'll answer it on the user list. I don't think the idea is developed enough to discuss on biz. First - vtiger is available for those who don't like the SugarCRM licensing. It's not a licensing complaint. At least that has not surfaced. It is more that the programmer does not seem to be comfortable with SugarCRM, MySQL and php. Biggest compliant about sugar is - hard to configure, does not work with latest php. Second - developing your own CRM is an ambitious undertaking. You need good reasons to go in that direction. Third - I have enough exposure to Visual FoxPro to quickly rule it out as a choice for anything new. The fact that somebody is proposing to use it might give you the idea that they don't know what they are talking about at all. BTW - my exposure to it did include things like access from linux apps using ODBC so I know enough to hate the product. It still makes me wonder why the programmer chooses Visual Foxpro. Sounds like he also rejects many other language and database options. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom 650
What version of the SIP firmware is on you boot server? You need to check with your reseller, because the 650 needs at least 2.1.0. On 4/29/07, Klaverstyn, David C [EMAIL PROTECTED] wrote: All, I have a Polycom 650 phone, when turned on displays Checking application. Can any give me some information as to what is wrong? I have copied the CFG files from a 601 phone to work with this 650. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] polycom boot server...
Jordan, After the phone powers up go into the setup menu, before the autoboot, and set the following: Under DHCP Menu set Boot Server to Static Under the Server Menu setup your boot server information. If you want to completely forgo setting up an FTP server for the files you might look at running a TFTP server of your workstation and pointing the phones to it. Good Luck! On 4/23/07, Jordan Novak [EMAIL PROTECTED] wrote: I have to re-image one phone, I do not want to setup a small network with DHCP and FTP to get it done. Can I just point the phone at the server manually to try to bypass putting another dhcp server on my network. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] OMG Verizon is terrible
This could go on forever, I mean take your pick Verizon, Att, Bell South any of them. Same story We are the phone company, who else can you call?. We have time and again seen it take weeks to get the order documents created, not the actual order, just the paperwork to create the order. I personally take great joy in finding anyway not to deal with them. The only way I see them changing their ways is by losing enough customers. I think Verizon is learning that lesson, but their response is not to compete and satisfy the customers, but to put the competition in a strangle hold with patents that are vague and broad. Ultimately I think Verizon will suffer from the court decisions more then anyone else, the true nature of their leadership is not to satisfy the customer. Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Big trouble with zap lines
Do you have fxs modules or fxo modules? PSTN connects to fxo, but the signaling is fxs like you have. On 4/20/07, Ricardo Melendez [EMAIL PROTECTED] wrote: Hi, I recently install asterisk 1.4 over fedora 4, I configured 12 zap channels like this In zaptel.conf fxsks=1-4 fxsks=5-8 fxsks=9-12 loadzone=us defaultzone=us in zapata.conf in channels section context=incoming signalling=fxs_ks channel = 1-4 channel = 5-8 channel = 9-12 when i run ztcfg -vv show 12 channels correctly configured whe i run zap show channels in asterisk console this show 12 channels correctly configured when i call to a zap channel like this Dial(zap/1/somenumber,15,r) in the console appear that asterisk is dialing trought this channel to this somenumber but in the line the call never go out nor in, the same happens when dial from outside, the line is ringing until the normal timeout. the PSTN lines used work normally whit normal hardphones (PSTN) zaptel, asterisk, zttool and ztcfg all never send any error message. What could be the problem?? Could be a damaged wildcard My card is wctdm2400p with 12 fxs ports in 3 modules thanks in advance ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Big trouble with zap lines
Do you have fxs modules or fxo modules? PSTN connects to fxo, but the signaling is fxs like you have. On 4/20/07, Ricardo Melendez [EMAIL PROTECTED] wrote: Hi, I recently install asterisk 1.4 over fedora 4, I configured 12 zap channels like this In zaptel.conf fxsks=1-4 fxsks=5-8 fxsks=9-12 loadzone=us defaultzone=us in zapata.conf in channels section context=incoming signalling=fxs_ks channel = 1-4 channel = 5-8 channel = 9-12 when i run ztcfg -vv show 12 channels correctly configured whe i run zap show channels in asterisk console this show 12 channels correctly configured when i call to a zap channel like this Dial(zap/1/somenumber,15,r) in the console appear that asterisk is dialing trought this channel to this somenumber but in the line the call never go out nor in, the same happens when dial from outside, the line is ringing until the normal timeout. the PSTN lines used work normally whit normal hardphones (PSTN) zaptel, asterisk, zttool and ztcfg all never send any error message. What could be the problem?? Could be a damaged wildcard My card is wctdm2400p with 12 fxs ports in 3 modules thanks in advance ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Announce] Web-MeetMe V3.0.1 released
Also - this is probably again a problem of the missing documentation, but let me clarify my problem in detail: If I create a conference, there is a button email participants. If I click that button, nothing happens (). How does the whole email procedure works? How does the web-meetme gather the email addresses of the participants? There is no way how to configure participants to the conference. I have seen this with my setup, I am using the client mode for emails, when using firefox. Strange enough IE works. Most of our users are on IE so I have not researched the why. -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Announce] Web-MeetMe V3.0.1 released
Also - this is probably again a problem of the missing documentation, but let me clarify my problem in detail: If I create a conference, there is a button email participants. If I click that button, nothing happens (). How does the whole email procedure works? How does the web-meetme gather the email addresses of the participants? There is no way how to configure participants to the conference. I have seen this with my setup, I am using the client mode for emails, when using firefox. Strange enough IE works. Most of our users are on IE so I have not researched the why. -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IMAP Voicemail with MS Exchange
I would love to know if you get this working. We use the SMTP features now, but the ability for a message to be managed from either email client or phone and be changes seen in both is the missing link for us. On 4/11/07, Stephen Bosch [EMAIL PROTECTED] wrote: Anthony Rodgers wrote: Hi there, We're trying to get IMAP voicemail storage working on an MS Exchange server - I would be grateful if anyone who has successfully done this could post the magic soup here, as extensive Google searching has yielded nothing other than tantalizing references to it being done without any specifics. I haven't used IMAP voicemail yet, so you'll have to bear with me here. Have you tried configuring Asterisk to save voicemail messages on the Exchange server using IMAP? What was the result? IMAP support in Exchange, as in Outlook, is rough and rather ugly. For obvious reasons it's never been in MS interest to support it properly, as they want people to use their native Exchange server protocol. There's probably a good reason why you want to do it the way you want to, but I'll ask the question anyway -- what about delivering the message over SMTP? -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Different devices for asterisk!!!
If your device is connecting to asterisk as a peer or a friend, the the sip show peers user will show a user agent field. For example I have a linksys phone in my home office that connects as a friend and so if I type sip show peer 1000 - the phones username. I get the following entry Useragent: Linksys/SPA942-5.1.5 Which tells me brand, model and firmware. This field stores the user agent string sent by the device, so each manufacture and even device may give different information. Here is another just to show some of the detail. Useragent: Aastra 480i Cordless/1.3.0.1080 Brcm Callctrl/1.5 MxSF/v3.2.6.26 On 4/7/07, Rizwan Hisham [EMAIL PROTECTED] wrote: Hi all, Im trying dial a user according to the device s/he uses. i mean if the user is using asterisk as a peer, then i have to pass the extension in the dial application like this: Dial(SIP/[EMAIL PROTECTED]) ;so that s/he can perform routing according to the DNID. and if the user is using sipura, linksys or grandstream i dial the user like this, Dial(SIP/user) so is there a way to know what kind of device user has used to register with my asterisk server? Thanx in advance -- Regards Rizwan Hisham Software Engineer ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CRM integration with Asterisk
Will an Outlook dialer run on Linux ?? Works fine on XP, might check your OS :) On 4/7/07, Philipp Kempgen [EMAIL PROTECTED] wrote: S. A. Kamran wrote: I just want to add here that Star Outlook Dialer (Free Edition) has built in integration (through StarJunction) with SugarCRM as well as with Salesforce CRM. It is available for free download at http://www.starutilities.com/staroldialer.htm Can someone tell me how to decompress staroldialer.exe? Tried to make it executable but that doesn't work either: bash: ./staroldialer.exe: cannot execute binary file Is Star Utilities aware of the problem? ;) Happy Easter! Philipp ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Adding DND to dialplan
Brian,
DND is not real hard. You basicly want to to note the extension is set to
DND and then when someone calls that extension you check for DND status and
if it is yes then you go on to voicemail instead of dial. It sounds like you
are miss understanding the dialplan and how to use it. In your sample, do
the macros user-callerid and hangupcall exist? Do the sound files you
specified exist in var/lib/asterisk/sounds? A simple DND would look like so:
exten = *73,1,Answer()
exten = *73,n,Wait(0.5)
exten = *73,n,Set(DB(${CALLERID(number)}/DND)=1)
exten = *73,n,Playback(do-not-disturb)
exten = *73,n,Playback(enabled)
exten = *73,n,Hangup()
and then
When someone calls say extension 1000 I would have a macro check for :
exten = s,n,Set(DNDStatus=$[${DB(1000/DND)} = 1]) = returns a 1 if
enabled or a 0
exten = s,n,GoToIf($[${DNDStatus} = 1]?DND)
exten = s,n(DND),Voicemail([EMAIL PROTECTED],u)
On 4/3/07, Brian McEntire [EMAIL PROTECTED] wrote:
Hello -
I've read Asterisk should be able to activate a do not disturb feature
to turn off the ringers on extensions. I checked the wiki and can't
find documentation for how to do it.
Here's my attempt, added to extensions.conf:
[dnd-on]
exten = _#78,1,Answer
exten = _#78,n,Wait(1)
exten = _#78,n,Macro(user-callerid,)
exten = _#78,n,Set(DB(DND/${CALLERID(number)})=YES)
exten = _#78,n,Playback(do-not-disturbactivated)
exten = _#78,n,Macro(hangupcall,)
[dnd-off]
exten = _#79,1,Answer
exten = _#79,n,Wait(1)
exten = _#79,n,Macro(user-callerid,)
exten = _#79,n,dbDel(DND/${CALLERID(number)})
exten = _#79,n,Playback(do-not-disturbde-activated)
exten = _#79,n,Macro(hangupcall,)
;further down
include = dnd-on
include = dnd-off
- - -
Monitoring asterisk from the CLI, when I dial #78 on an extension, I
just get a fast busy signal and this information is reported on the
CLI:
Apr 3 10:41:33 WARNING[30702]: app_macro.c:144 macro_exec: No such
context 'macro-user-callerid' for macro 'user-callerid'
Apr 3 10:41:33 WARNING[30702]: func_db.c:97 function_db_write: DB
requires an argument, DB(family/key)=value
Apr 3 10:41:33 WARNING[30702]: file.c:504 ast_openstream_full: File
do-not-disturb does not exist in any format
Apr 3 10:41:33 WARNING[30702]: file.c:816 ast_streamfile: Unable to
open do-not-disturb (format unknown): No such file or directory
Apr 3 10:41:33 WARNING[30702]: app_playback.c:106 playback_exec:
ast_streamfile failed on Zap/2-1 for do-not-disturbactivated
Apr 3 10:41:33 WARNING[30702]: file.c:504 ast_openstream_full: File
activated does not exist in any format
Apr 3 10:41:33 WARNING[30702]: file.c:816 ast_streamfile: Unable to
open activated (format unknown): No such file or directory
Apr 3 10:41:33 WARNING[30702]: app_playback.c:106 playback_exec:
ast_streamfile failed on Zap/2-1 for do-not-disturbactivated
Apr 3 10:41:33 WARNING[30702]: app_macro.c:144 macro_exec: No such
context 'macro-hangupcall' for macro 'hangupcall'
- - -
Any tips?
All I really want to do is turn off the ringers / do not ring
extenstions when I've activated DND. Right now I'm just using a hack
which is to shutdown asterisk altogether when I don't want the phones
to ring, which of course also prevents dialing out, it's a
sledgehammer approach and I'm looking for something more typical.
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Re: [asterisk-users] Adding DND to dialplan
You have a syntax error.
exten = _#78,n,Set(DB(${DND/CALLERID(num)})=1)
should read
exten = _#78,n,Set(DB(DND/${CALLERID(num)})=1)
On 4/3/07, Brian McEntire [EMAIL PROTECTED] wrote:
Hmmm...
Had hoped this would be easy, maybe still is, but running into a problem:
When I dial #78, I get a fast busy and these errors on the CLI
[Apr 4 00:39:29] ERROR[4046]: pbx.c:1523 ast_func_read: Function
DND/CALLERID not registered
[Apr 4 00:39:29] WARNING[4046]: func_db.c:87 function_db_write: DB
requires an argument, DB(family/key)=value
- - -
Here is the extensions.conf entry:
[dnd-on]
exten = _#78,1,Answer()
exten = _#78,n,Wait(1)
exten = _#78,n,Set(DB(${DND/CALLERID(num)})=1)
exten = _#78,n,Playback(do-not-disturb)
exten = _#78,n,Playback(enabled)
exten = _#78,n,Hangup()
- - -
It appears to me that Set(DB ... as a function isn't working, isn't
built in, or needs more information.
I saw something about GLOBAL variables, perhaps I can use those instead?
On 4/3/07, Doug Lytle [EMAIL PROTECTED] wrote:
Brian McEntire wrote:
Hello -
I've read Asterisk should be able to activate a do not disturb feature
Instead of using 2 extensions, you can get away with just one. Check
the database entry at the start, if it's already set, remove it. If
it's not there, add it.
[dnd]
; **
; Do not disturb can be set via Asterisk
; instead of the phones by dialing this
; number.
; **
exten = 79*,1,Set(CALLBACK=${DB(DND/${CALLERIDNUM})})
exten = 79*,2,GotoIf($[${CALLBACK} = YES]?79*,3:79*,101)
exten = 79*,3,Set(DB(DND/${CALLERIDNUM})=NO)
exten = 79*,4,Playback(local/stutter)
exten = 79*,5,Playback(de-activated)
exten = 79*,6,Hangup()
exten = 79*,101,Set(DB(DND/${CALLERIDNUM})=YES)
exten = 79*,102,Playback(local/stutter)
exten = 79*,103,Playback(activated)
exten = 79*,104,Hangup()
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little
Temporary Safety, deserve neither Liberty nor Safety.
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Re: [asterisk-users] Speed Dial Application in *
You can build it. I put one in my systems that uses an mysql table that the users can edit via a web interface and then the dialplan does a lookup and dails the number. On 3/30/07, Chris Nighswonger [EMAIL PROTECTED] wrote: Hi all, Is there a speed dial type application in *? The NEC PBX we currently use has a feature which allows any phone to access a system-wide speed dail database simply by keying the speed-dial number and pressing the 'redial' key from any extension. Even using a vinella phone on an sli the user can dial 77+speedial# and access this directory. Does * have a similar feature? Thanks, Chris ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Speed Dial Application in *
You can build it. I put one in my systems that uses an mysql table that the users can edit via a web interface and then the dialplan does a lookup and dials the number. On 3/30/07, Chris Nighswonger [EMAIL PROTECTED] wrote: Hi all, Is there a speed dial type application in *? The NEC PBX we currently use has a feature which allows any phone to access a system-wide speed dail database simply by keying the speed-dial number and pressing the 'redial' key from any extension. Even using a vinella phone on an sli the user can dial 77+speedial# and access this directory. Does * have a similar feature? Thanks, Chris ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PoE - IEEE 802.3af
A POE switch will put power on what ever line is connected to it, so if your polycom plugs into a wall plate with cat 5 cable that runs back to a port on the POE switch then you have power all the way to the phone. On 3/28/07, Mike [EMAIL PROTECTED] wrote: Hi, I'm not clear on how to use Power--over-Ethernet, specifically with Polycom phones. What I understand, is that by buying the Polycom 501 with the 802.3afcable bundle, I simply connect my phone, through the Polycom provided special RJ-45 cable, into a PoE capable switch, and voilà! Is this true? And if so, what happens when the Phone doesn't connect directly to the switch? (let`s say there is wiring in the wall that goes to a patch panel, for example. Do I need to change all the wiring in the office?) Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom and Asterisk
Matt, I am running Polycom 2.1 on both 1.4 and 1.2 svn releases without any problems. What kind of issues did you experience? On 3/28/07, Mike Hammett [EMAIL PROTECTED] wrote: I was previously having an issue with a Polycom phone and Polycom support said that Asterisk didn't play well with Polycom firmware versions 1.6.7and newer due to SIP compatibility issues. I believe I heard a lot of things were fixed\adjusted in 1.4 and was wondering if anyone has had success with Asterisk 1.4 and the latest Polycom firmware releases. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Chan_cellphone and CentOS 4.x
Thanks for the tip, I got the device to pair, now it just won't stay connected. In Asterisk the CLI shows -- Bluetooth Device bruce has connected. -- Bluetooth Device bruce initialised and ready. -- Bluetooth Device bruce has disconnected, reason (104). Device bruce is a Motorola V3, what is reason 104?? On 3/26/07, Jay Milk [EMAIL PROTECTED] wrote: Bruce Reeves wrote: I ran into a problem today while trying to compile chan_cellphone version 17 on a CentOS 4.4 machine. Apparently the bluez and autoconf versions were to old and as I tried to install the latest version, I found that the new bluez-lib would install and allow the chan_cellphone to compile, but bluez-utils required an update to D-sub which in turn required python 2.4 or better. That apparently in not possible on CentOS from what I have read. So I have it compiled and it sees my phone, but paring fails every time, which I think is due to the utils package. All that to ask does anyone have this chan working on CentOS 4? If so can you help me through this? -- Bruce Reeves Nortex Networks How'd pairing fail? I've worked with a lot of BT goodness in linux (albeit ubuntu), but the pin-helper stuff has never really don't it for me. What I have is: 1. hcid.conf referencing /usr/bin/bluepin as a pin_helper; 2. /usr/bin/bluepin containing: #!/bin/bash cat /etc/bluetooth/pin 3. /etc/bluetooth/pin containing: PIN:12345678 watch permissions. Works every time. (my BT setup periodically checks for the presence of various BT capable cellphones and then synchronizes the phone books using gsmtools; also daily downloads pics/videos from phones with that capability) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Chan_cellphone and CentOS 4.x
I ran into a problem today while trying to compile chan_cellphone version 17 on a CentOS 4.4 machine. Apparently the bluez and autoconf versions were to old and as I tried to install the latest version, I found that the new bluez-lib would install and allow the chan_cellphone to compile, but bluez-utils required an update to D-sub which in turn required python 2.4 or better. That apparently in not possible on CentOS from what I have read. So I have it compiled and it sees my phone, but paring fails every time, which I think is due to the utils package. All that to ask does anyone have this chan working on CentOS 4? If so can you help me through this? -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] freepbx - DB Error messages...
You might get a faster response on freepbx/amp mailing list. On 3/24/07, Francesco Peeters (Asterisk) [EMAIL PROTECTED] wrote: Hi all, I am probably missing something ultimately obvious, but I have a problem configuring freepbx... Using Edgy Eft with the cvs freePBX 2.2.1 and followed the Ubuntu installation guide on freepbx.org. System pxe-boots from a server with NFS root on same Using * 1.2 current (from source, not .deb's) Using mISDN-streams (from source, not .deb's) Using freePBX-2.2.1 (from source, not .deb's) Installed everything, and mISDN and * load just fine amportal start works fine as well However I keep getting DB Error's in the GUI... The syslog gives two separate errors: 1) Error 127 when reading table ./asterisk/whatever 2) Table is crashed and needs to be repaired I created a special mysql user for * and did an PERMIT ALL PRIVILEGES on the mysql databases When I log in to mysql as root and do 'SELECT username FROM ampusers ORDER BY username' I get the record list. When I do the same as the * user, I get the 'Table is crashed, blablabla' line. I tried changing the login user for freepbx (ampdbuser) to root, but that doesn't help either, as I keep getting the 127 error... Googling wasn't very helpful, and the freepbx forum admins still haven't approved my account, so I thought I'd try here... Any help appreciated! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sendmail and exchange for voicemail integration
Jordan, Assuming that the voicemail users are email users on the domain for exchange then your DNS entries for MX will take care of most of the work. Sendmail on the Centos installs I have done has required no changes to the default config to work with our exchange servers. You probably will want to make sure that the SMTP protocol on Exchange allows the Sendmail server to relay. On 3/23/07, Jordan Novak [EMAIL PROTECTED] wrote: I am having real trouble getting Asterisk to send to exchange. They are on the same LAN. Does anyone know of a walkthrough for this setup. I have gotten it to work before, but that was to a hotmail account. Jordan Novak ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PC / Phone Combo
Sounds like your looking for a laptop with a good softphone. You take an ultralight laptop and you could probably meet all these requirements plus some. On 3/23/07, Dean Collins [EMAIL PROTECTED] wrote: Hi Chris, What are you hoping to do on the browser? I don't have a solution for you but curious about what your intentions are. Could a wireless display or similar with a softphone application provide what you are looking to achieve? Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Chris Gamble Sent: Friday, 23 March 2007 1:42 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] PC / Phone Combo Is there a phone in the sub-400$ department that has at least a 10 inch display and a nearly modern web browser with keyboard hook-up that is either sip or iax based? I dont want to do video, just have a good web browser built into the phone. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] polycom random reboots
Yes, I recently saw this with a 501, in my case the network drop was the problem. If you have a good tester then run it on the connection. I had another drop near by and just swicthed to it. On 3/21/07, Louis-David Mitterrand [EMAIL PROTECTED] wrote: Hi, At one location we have a user whose Polycom IP430 suffers from erratic reboots. We swapped his phone for a brand new one, but the problem remains. Has anyone seen that? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] polycom random reboots
The one I use the most is a Fluke Net Tool. It can determine polarity problems and I believe has some diagnostics for POE and VOIP. It also can go in-line between a device and the network and help diagnose problems the device is experiencing that the tool would not encounter on its own. On 3/21/07, Louis-David Mitterrand [EMAIL PROTECTED] wrote: On Wed, Mar 21, 2007 at 05:53:27AM -0500, Bruce Reeves wrote: Yes, I recently saw this with a 501, in my case the network drop was the problem. If you have a good tester then run it on the connection. I had another drop near by and just swicthed to it. What kind of test tool would you suggest? Usually we rely on the cabling guys for that but that entails a delay and I'd be interested in knowing how to do it myself. Thanks, ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dlink i2eye
I have seen this product before and wondered, has anyone connected this to Asterisk? http://www.i-2-eye.com/index.html As far as that goes has anyone seen a set top box video phones that work with Asterisk? -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Follow me on multiple numbers..
Have you looked at the Follwo-me feature in 1.4? It can require the answering channel to accept the call. You might take a look. On 3/16/07, Ritesh Agrawal [EMAIL PROTECTED] wrote: Hi Folks, I want to setup a follow me routine so that asterisk can call me on the multiple numbers. I tried some of the samples at voip-info but there is a problem with those examples. I dont have coverage in my home area and my cell phone answering machine picks up the phone right away so my home phone never rings. I also want the caller to be able to leave a voicemail and the cell phone answering machine messes it all up. I have call screening setup so the call gets answered by the cell phone answering machine and it never accepts the call. I would appreciate if someone can help me with the setup. Thanks. R PS: I am always open to making a donation to a charity of your choice. Please help me and help the charity :-) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FW: Seamless Multi Office Asterisk Deployment
on time frames? P.S. I want to thank everyone who replied so quickly, surprised my co-worker and I :D -- Thanks, Brandon Comouche IT Administrator Sno Falls Credit Union -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves Sent: Monday, March 12, 2007 9:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FW: Seamless Multi Office Asterisk Deployment I'm more then happy to share my experiences with anyone, there is just a lot to be said about the things Brandon is trying to accomplish. Take the automatic fail over he mentioned, there are a number of ways to do that and everyone has an opinion. I just want make myself available to help other get from playing with Asterisk like I did to really putting it to use so that people sit back and say wow, my cisco/avaya/nortel can't do that. On 3/12/07, Sean Bright [EMAIL PROTECTED] wrote: Why does everyone want to go off-list? Is this not information that could benefit others? On 3/12/07, Bruce Reeves [EMAIL PROTECTED] wrote: Brandon Your on the right track with what is can do. It will also be good to look into what kind of QOS you can do on the T-1 connections between offices. I have an 8 office setup similar to this and many of your goals I have achieved and would be glad to offer ideas and such if you want to email me off list. On 3/12/07, Brandon Comouche [EMAIL PROTECTED] wrote: Hello I have a brief and a long question about a possible Asterisk deployment I am planning. Long Story Short: I have four total offices, one main and three remote. All offices are connected using dedicated network T1 lines creating one unified network across offices. I would like to know if it is possible to set up an Asterisk system with the following capabilities: - Branch Unification (I know this can be done) - Branch Independence (In case of T1 network Failure, PSTN line failover at each branch) - Roaming Extensions (A user can go to any office and log in to a phone - hopefully check voice mail as well) Basically, I am asking if Asterisk can be a system that will seamlessly operate as one big system and handle failovers as well. After spending hours playing with Asterisk, reading voip-info.org, and watching this list, it seems that Asterisk can handle anything. I just would like re-assurance that I am not chasing a lost cause. A simple Yes or No answer is acceptable to me. Below I have a long version of what I am trying to do if anyone is in the mood to give me more pointers J Brandon (Long Version Follows) Long Story Version: Here is what I have to work with: - Four Offices (One main and three remote) - Dedicated T1 lines connecting three remote offices to one main office (all connections made through the main office) - Will have a T1 Voice line at the main office - Three PSTN lines at each remote office Essentially what I would like to do is create a system comparable to the ShoreTel ShoreGear product line (if you are familiar with it). This system will seamlessly unite all offices as one and provide failover in the case of line outage. It also allows users to roam from phone to phone across offices seamlessly. It has many more features, but those are two main features I am looking for. About 40 total phones will be deployed. To make it even more difficult, I would like all user extensions to start the same (i.e. across offices all extensions are 5### with no discernable pattern). Progress so far: At this time I have determined that I will need a server at each office as well as a T1 card (TE110P) at the Main office and the four port TDM PSTN cards at each remote office. I plan on using the Polycom IP 430 or 501 (Undecided, 501 if required). I have been using TrixBox to this point, would like to continue if possible. It appears that I will want to use DunDi in some fashion to unite the branches. My main roadblock right now is trying to figure out how to get all the information across the offices at the same time (extensions, voicemail). I have successfully had two boxes communicate, but what I am looking for is much more complex I feel. I have thought of synchronized MySQL databases, but I do not know if that will work the way I wish. If anyone reads this far ;) I am looking for suggestions for routes I might consider or places I should/could look for more information. I am relatively new to Asterisk, but I am not afraid to get my hands dirty. If something I said did not make any sense or if there is more information I could provide, I am happy to help where I can. Thank you for your time and assistance. Brandon Comouche
Re: [asterisk-users] Incoming Caller ID
That should be provided by your telco, if your referring to a PRI on a Sangoma T-1 card. On 3/15/07, Rob Vinson [EMAIL PROTECTED] wrote: Does anyone know if I can get Incoming caller id name and number on a sagnoma PRI. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What is the best phone to get when using a headset?
Cory, Are the Polycom phones able to detect that the headset is off-hook? We have had problems with 2 different brands of headsets working fine but the phones seem unaware of the headset's hook state. On 3/14/07, Cory Andrews [EMAIL PROTECTED] wrote: I like any of the Polycom Soundpoint IP Series phones (IP301, 430, 501, 601, 650) paired with a VXI Tuffset 37 with Quick Disconnect Cord. IMO VXI makes the best headsets out there for the money, and they are quite inexpensive. You can find more info on VXI at www.vxicorp.com Cory Andrews -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gareth Blades Sent: Wednesday, March 14, 2007 8:54 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] What is the best phone to get when using a headset? We currently have the Grandstream GXP-2000 phones which generally work very well except that we cannot get find a headset which works reliably with them. Either the sound quality is poor or the other party has difficulty in hearing us. We therefore want to get a couple of different phones and headsets for our customer service people. What would you recommend as a good phone and headset combination? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] _ALERT_INFO replacement in 1.4?
Does SIPAddHeader(Alert-Info:) not do it? On 3/12/07, Nikhil Jogia [EMAIL PROTECTED] wrote: Hi All I have just upgraded from Asterisk 1.2 to 1.4 and am having trouble with with one of my ATAs not ringing. Basically, when I execute the Dial command, an error occurs: Got SIP response 400 In alert-info header: Empty value expected Now in 1.2, I just issued the following command to overcome this problem: Set(_ALERT_INFO=). Now in 1.4, _ALERT_INFO is deprecated, so I have to use SIPAddHeader, but I don't know how, or if there is a way to remove the alert-info header. Here is my dialplan snippet: exten = s,9,Playback(my-greeting) exten = s,10,Wait(1) exten = s,11,SIPAddHeader(Alert-Info: info=bellcore-r4) exten = s,12,Dial(SIP/600SIP/602SIP/603,60,tm) exten = s,13,Set(_ALERT_INFO=) exten = s,14,Dial(SIP/604,60,tm) exten = s,15,Voicemail(su600) exten = s,16,Hangup exten = s,115,Voicemail(sb600) exten = s,116,Hangup As you can see, #13 is deprecated, so extension 604 does not ring. Extension 600, 602 and 603 are all hooked up to Sipura ATAs and need the bellcore-r4 ringtone to differentiate from other incoming lines. Any ideas? Thanks Nikhil Jogia ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FW: Seamless Multi Office Asterisk Deployment
Brandon Your on the right track with what is can do. It will also be good to look into what kind of QOS you can do on the T-1 connections between offices. I have an 8 office setup similar to this and many of your goals I have achieved and would be glad to offer ideas and such if you want to email me off list. On 3/12/07, Brandon Comouche [EMAIL PROTECTED] wrote: Hello I have a brief and a long question about a possible Asterisk deployment I am planning. Long Story Short: I have four total offices, one main and three remote. All offices are connected using dedicated network T1 lines creating one unified network across offices. I would like to know if it is possible to set up an Asterisk system with the following capabilities: - Branch Unification (I know this can be done) - Branch Independence (In case of T1 network Failure, PSTN line failover at each branch) - Roaming Extensions (A user can go to any office and log in to a phone – hopefully check voice mail as well) Basically, I am asking if Asterisk can be a system that will seamlessly operate as one big system and handle failovers as well. After spending hours playing with Asterisk, reading voip-info.org, and watching this list, it seems that Asterisk can handle anything. I just would like re-assurance that I am not chasing a lost cause. A simple Yes or No answer is acceptable to me. Below I have a long version of what I am trying to do if anyone is in the mood to give me more pointers J Brandon (Long Version Follows) Long Story Version: Here is what I have to work with: - Four Offices (One main and three remote) - Dedicated T1 lines connecting three remote offices to one main office (all connections made through the main office) - Will have a T1 Voice line at the main office - Three PSTN lines at each remote office Essentially what I would like to do is create a system comparable to the ShoreTel ShoreGear product line (if you are familiar with it). This system will seamlessly unite all offices as one and provide failover in the case of line outage. It also allows users to roam from phone to phone across offices seamlessly. It has many more features, but those are two main features I am looking for. About 40 total phones will be deployed. To make it even more difficult, I would like all user extensions to start the same (i.e. across offices all extensions are 5### with no discernable pattern). Progress so far: At this time I have determined that I will need a server at each office as well as a T1 card (TE110P) at the Main office and the four port TDM PSTN cards at each remote office. I plan on using the Polycom IP 430 or 501 (Undecided, 501 if required). I have been using TrixBox to this point, would like to continue if possible. It appears that I will want to use DunDi in some fashion to unite the branches. My main roadblock right now is trying to figure out how to get all the information across the offices at the same time (extensions, voicemail). I have successfully had two boxes communicate, but what I am looking for is much more complex I feel. I have thought of synchronized MySQL databases, but I do not know if that will work the way I wish. If anyone reads this far ;) I am looking for suggestions for routes I might consider or places I should/could look for more information. I am relatively new to Asterisk, but I am not afraid to get my hands dirty. If something I said did not make any sense or if there is more information I could provide, I am happy to help where I can. Thank you for your time and assistance. Brandon Comouche An IT Guy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FW: Seamless Multi Office Asterisk Deployment
I'm more then happy to share my experiences with anyone, there is just a lot to be said about the things Brandon is trying to accomplish. Take the automatic fail over he mentioned, there are a number of ways to do that and everyone has an opinion. I just want make myself available to help other get from playing with Asterisk like I did to really putting it to use so that people sit back and say wow, my cisco/avaya/nortel can't do that. On 3/12/07, Sean Bright [EMAIL PROTECTED] wrote: Why does everyone want to go off-list? Is this not information that could benefit others? On 3/12/07, Bruce Reeves [EMAIL PROTECTED] wrote: Brandon Your on the right track with what is can do. It will also be good to look into what kind of QOS you can do on the T-1 connections between offices. I have an 8 office setup similar to this and many of your goals I have achieved and would be glad to offer ideas and such if you want to email me off list. On 3/12/07, Brandon Comouche [EMAIL PROTECTED] wrote: Hello I have a brief and a long question about a possible Asterisk deployment I am planning. Long Story Short: I have four total offices, one main and three remote. All offices are connected using dedicated network T1 lines creating one unified network across offices. I would like to know if it is possible to set up an Asterisk system with the following capabilities: - Branch Unification (I know this can be done) - Branch Independence (In case of T1 network Failure, PSTN line failover at each branch) - Roaming Extensions (A user can go to any office and log in to a phone – hopefully check voice mail as well) Basically, I am asking if Asterisk can be a system that will seamlessly operate as one big system and handle failovers as well. After spending hours playing with Asterisk, reading voip-info.org, and watching this list, it seems that Asterisk can handle anything. I just would like re-assurance that I am not chasing a lost cause. A simple Yes or No answer is acceptable to me. Below I have a long version of what I am trying to do if anyone is in the mood to give me more pointers J Brandon (Long Version Follows) Long Story Version: Here is what I have to work with: - Four Offices (One main and three remote) - Dedicated T1 lines connecting three remote offices to one main office (all connections made through the main office) - Will have a T1 Voice line at the main office - Three PSTN lines at each remote office Essentially what I would like to do is create a system comparable to the ShoreTel ShoreGear product line (if you are familiar with it). This system will seamlessly unite all offices as one and provide failover in the case of line outage. It also allows users to roam from phone to phone across offices seamlessly. It has many more features, but those are two main features I am looking for. About 40 total phones will be deployed. To make it even more difficult, I would like all user extensions to start the same (i.e. across offices all extensions are 5### with no discernable pattern). Progress so far: At this time I have determined that I will need a server at each office as well as a T1 card (TE110P) at the Main office and the four port TDM PSTN cards at each remote office. I plan on using the Polycom IP 430 or 501 (Undecided, 501 if required). I have been using TrixBox to this point, would like to continue if possible. It appears that I will want to use DunDi in some fashion to unite the branches. My main roadblock right now is trying to figure out how to get all the information across the offices at the same time (extensions, voicemail). I have successfully had two boxes communicate, but what I am looking for is much more complex I feel. I have thought of synchronized MySQL databases, but I do not know if that will work the way I wish. If anyone reads this far ;) I am looking for suggestions for routes I might consider or places I should/could look for more information. I am relatively new to Asterisk, but I am not afraid to get my hands dirty. If something I said did not make any sense or if there is more information I could provide, I am happy to help where I can. Thank you for your time and assistance. Brandon Comouche An IT Guy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- sean
Re: [asterisk-users] Asterisk distributed deployment
I just completed a deployment of 8 sites connected via MPLS, and I chose to go with the local * servers option and Sangoma hardware at each site. I then put dundi in place to route calls between sites and will later look at adding LCR. I'm with Steve on the cards, don't skimp on cards or even echo canceling. Most of my sited were 2-5 employees and I used Dell Optiplex systems for their servers, overkill on capabilities, but easy to maintain parts for. On 3/8/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hello all, I post this issue thinking too that could help other people on an asterisk deployment over distributed offices considering both quality, prices, devices and so. Well, i am working on a deployment of a telephony system based in asterisk. My company have a central office with seven remote offices connected all through a VPN. To reduce and evaluate costs i consider solutions like: Asterisk servers on all locations(central and remote offices) or Asterisk on Central office plus FXO Gateways on remote offices, all of this connected through a central asterisk cluster With the first option i have TDM cards seller that offer me DIGIUM (expensive) or OPENVOX (less expensive), but because i not have experience with OPENVOX telephony hardware I cant consider that. So, if Any can give me some good reasons for use OPENVOX against DIGIUM cards i would have solve this question because may build IAX trunks on each office. With the 2nd option I have sellers that offer me gateways: Quintum Tenor AFT400 Planet VIP-480 FO But, again, I don't have experience with asterisk and FXO gateways to think that it is the best solution amen that is the less expensive solution. Another solution that i consider is mount asterisk on central office and IP PBX DIGISTAR preconfigured on remote offices. On the Users Side I was considering the use of Ata's or FXS Gateways, with Ata's I get offers of Audiocodes MP202, GRANDSTREAM HT 386 or Linksys SPA-2002. And with FXS Gateways sellers offers me Quintum Tenor AXG2400, Quintum Tenor AFG800 Thanks for any word that can help me to get this VoIP deployment working and sorry for my english. Cheers G. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT Vonage V-Phone Adapter (Possible Hack)
Steve, If you can get this to work with your own choice of softphone please post back to the list. I've wondered about it myself. On 3/7/07, Steve Totaro [EMAIL PROTECTED] wrote: It would be cool to get one of these and see if it can be hacked and loaded with your favorite SIP or IAX softphone. Looking at the pic, it looks like the dongle is both a soundcard and memory stick. Heck, I would be glad to have it if I could get the soundcard to work. Might as well since it is free after rebate. http://www.circuitcity.com/ssm/Accessories-for-Vonage-V-Phone-VPHONE/sem /rpsm/oid/162059/rpem/ccd/productDetailAccessory.do#tabs Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] rtsavesysname not working in 1.4
David, Here is what is working on my system, I added the following coulmn to the sip table regserver and it is varchar(20) and then set the following items in conf files. asterisk.conf systemname = server1 sip.conf displaysystemname=yes - Olle told me about this rtsavesysname=yes I bet the displaysystemname=yes is the missing setting, I seem to remeber not getting anywhere till I added that. On 3/5/07, David Thomas [EMAIL PROTECTED] wrote: On 3/2/07, Bruce Reeves [EMAIL PROTECTED] wrote: Try renaming you column in the peers table to regserver Thanks for the suggestion Bruce, unfortunately it did not help. Any other thoughts? Does the systemname in asterisk.conf and regserver in field mysql need to be an IP address, FQDN, hostname, or what is the proper format? Regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A New Phone Service - www.virtualphoneline.com
Or the fact that www.virtualphoneline.com is part of DIDXchange and of course you love it since you work for supertec.com, didxchange.com, and virtualphoneline.com On 3/5/07, Singer Wang [EMAIL PROTECTED] wrote: Follow this link http://www.virtualphoneline.com/signup/index.php?Referral2=vpl072625129bOID=10 non commerical eh? care to remove that Rferreal2= part? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie extensions.conf question
Chris,
Here is how I might use this, I have a context called inside, is where each
of my extensions is dialed from. On my home box it looks like this.
[inside]
exten = 1000,1,Dial(SIP/1000,20,t)
What I would probably do is add the Notify command to each of my extensions
before my Dial, like so
[inside]
exten = 1000,1,Notify(${CALLERIDNUM}|${CALLERIDNAME}|${EXTEN}/sunnybook)
exten = 1000,n,Dial(SIP/1000,20,t)
As a side note, if you are trying to get a screen pop, there are several
programs that connect to the manager API and will create a screen pop when
the a dial event is triggered. I mainly us snap, which runs on windows, and
I can configure it to watch a certain extension and display the call
information for that extension. So instead of having a special line in my
dial plan, I have a program filtering through events on the manage
interface. Hope this helps.
On 3/2/07, Chris Griffin [EMAIL PROTECTED] wrote:
I'm still stuck on just exactly where in my extensions.conf file I
should put the code below.
Chris Griffin
[EMAIL PROTECTED]
On Feb 28, 2007, at 9:55 PM, Patrick wrote:
On Wed, 2007-02-28 at 23:28 -0600, voiplist wrote:
Thanks for the link..
As for Google, I know how to use it. I searched for Sven Slezak's
Notify and other variations and got Squat..
Yes I had that too initially. The trick is to remove the 's from Slezak.
Then the first link that pops up is the link I gave below.
On 2/28/07, Patrick [EMAIL PROTECTED] wrote:
On Wed, 2007-02-28 at 22:04 -0600, voiplist wrote:
What does this module do?
On 2/28/07, Chris Griffin [EMAIL PROTECTED] wrote:
I've installed Sven Slezak's Notify module.
http://mezzo.net/asterisk/app_notify.html
Google is your friend.
Regards,
Patrick
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Re: [asterisk-users] rtsavesysname not working in 1.4
Try renaming you column in the peers table to regserver On 3/2/07, David Thomas [EMAIL PROTECTED] wrote: I am trying to have asterisk update the system name in my realtime peers, but it does not seem to be working. Here is what I've done so far. - added systemname = mysystemname in asterisk.conf - set rtsavesysname=yes in sip.conf. - created a table called sysname in my peers table in mysql - restarted asterisk - rebooted my phone to force a re-register Is there something I'm missing? Thanks! David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Realtime
What do you have setup in the res_mysql.conf file and extconfig.conf files? Have you installed the asterisk addons for 1.4 to get support for mysql? On 3/1/07, Mike Hammett [EMAIL PROTECTED] wrote: queue show Show status of a specified queue realtime load Used to print out RealTime variables. realtime update Used to update RealTime variables. restart gracefully Restart Asterisk gracefully Aiur*CLI realtime load You must supply a family name, a column to match on, and a value to match to. I am using Asterisk 1.4.0 and MySQL. It appears that the only realtime options are for loading and updating specific items from the database. The only database options seem to be for dundi. Under modules, all I could find is: Aiur*CLI module show like pbx_realtime.so Module Description Use Count pbx_realtime.soRealtime Switch 0 1 modules loaded --Mike -- Message: 12 Date: Thu, 01 Mar 2007 13:02:23 -0500 From: Brian Capouch [EMAIL PROTECTED] Subject: Re: [asterisk-users] Asterisk Realtime To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=windows-1252; format=flowed Mike Hammett wrote: Could someone provide some steps for troubleshooting Realtime? I cant see any signs that its working. I followed and double-checked a few different guides around the net, but havent been able to figure it out. You don't say which version you're running. I *think* the syntax is the same for both: realtime driver-name status will show you the status. For postgres it's pgsql for driver name (that's what I use). I think the other driver ids are mysql and odbc. If you don't see yourself connected, that's where to start. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SER / IAX solution
If re-invites are allowed then once both IAX endpoints are connected to Asterisk and the call is active the server will attempt to step out of the call. This actually works for both sip and IAX. On 2/27/07, Joseph [EMAIL PROTECTED] wrote: I find IAX connection with FWD very unreliable so I think I'll have to roll out my own SIP Express Router as I want to communicate with few SIP clients. So I hope this the right solution. I'm new to SER and to my understanding SER is like a road-map it tells the SIP Clients where they are so they can communicate directly with each other without going through a central server, am I right? What is the equivalent solution for IAX? If I have 5 clients registered to my box and all of them want to talk to each other the connection would go through my Asterisk server and that is not acceptable as they will kill my upload bandwidth; I want them to communicate with each other. What is FWD using for IAX clients? -- #Joseph ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] moving WiFi phone
In my experience having ap's with the same SSID and 3 channels of separation overlapping worked if the phone could roam. On 2/14/07, Ronald Wiplinger [EMAIL PROTECTED] wrote: Can anybody tell me how I can set-up multiple access points with overlapping coverage, so that a moving WiFi phone user can continuesly use the phone. bye Ronald Wiplinger ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] colors in the console
I have seen this when I have restarted the server from the asterisk CLI and not a service asterisk restart command. I'm not sure as to why, but I always assumed it had to do with the safe_asterisk file. On 2/12/07, Earle Clubb [EMAIL PROTECTED] wrote: Lacy Moore - Aspendora wrote: I'm wondering if anyone else has experienced this. Up until a few days ago, when accessing the CLI from my terminal program (Private Shell), the output was in color. I haven't upgraded, rebuilt, or to my knowledge, changed anything in Asterisk that would change this. My terminal settings were the same as well. I have two computers that I access the CLI regularly on, and neither show color anymore. When I disconnect, Private Shell shows the disconnect in red, just like before. This tells me that Private Shell is still doing color. What controls the color coding in the CLI? I found something in the source about it, but again, since it has been recompiled, this should not have changed. Is there a config file somewhere that I'm too blind to find? Thanks! -- Lacy Moore Somewhere I wish I wasn't I believe that only the CLI console provides color: e.g. asterisk -c. Connecting to an already-running asterisk process will not provide color: e.g. asterisk -r. Earle ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
