to exten = 888777,1,blah). Is this something they can
change in Trixbox?
http://pastebin.com/fa8b4f4e I highlighted the lines that contain the s
extension.
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
--
From
get the codec
error.
http://pastebin.com/f5b826d62 I highlighted the lines of interest. 34 is
the peer issue whereas 42 is the codec issue.
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
--
From: Steve Totaro stot
Do you know enough about Trixbox to tell me where they need to fix their
misconfiguration, or is it a Trixbox design flaw?
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
--
From: Steve Totaro stot...@first
. The
remaining two aren't provisioned anywhere. I'm going to be adding another
number shortly.
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
--
From: Steve Totaro stot...@totarotechnologies.com
Sent: Tuesday, February 10
Can anyone help me determine where the problem lies and how to fix it?
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
From: Mike Hammett
Sent: Thursday, January 15, 2009 1:00 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk - Trixbox
My
support told my carrier to fly a kite when we were
having T38 issues.
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
--
From: Steve Totaro stot...@totarotechnologies.com
Sent: Monday, February 02, 2009 9:36 AM
To: Asterisk
They are running Trixbox 2.6.1.10 and I'm running Asterisk 1.2.12.1.
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
--
From: Mike Hammett asterisk-us...@ics-il.net
Sent: Thursday, January 29, 2009 1:47 PM
To: Asterisk
debugging when that number isn't even present on that machine?
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
--
From: Adrià Vidal adriavi...@gmail.com
Sent: Friday, January 16, 2009 2:44 PM
To: Asterisk Users Mailing List
this
debugging output? I was calling 8159911010. My server is 208.100.1.33.
Theirs is 208.1.87.235. I solved the s@ problem on the other server by adding
insecure settings, but that didn't seem to solve it on this one.
http://pastebin.com/f5151341f
-
Mike Hammett
Intelligent Computing Solutions
Has anyone had any luck with Attrafax? I'm looking to use it as the T.38
gateway (PRI in, T.38 out).
--
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com
Thanks for the help. I still had a misconfiguration in my res_odbc.conf, but I
figured it out and it appears my voicemail storage is working. I haven't had a
chance to get to the phone on the extension I'm using for it.
--
Mike Hammett
Intelligent Computing Solutions
http://www.ics
;accountcode=2
context=ics
secret=
username=9826
fromuser=8157879826
insecure=very
host=63.175.151.3 ;voip.essex1.com
fromdomain=63.175.151.3 ;voip.essex1.com
;canreinvite=no
;disallow=all
;allow=ulaw
--
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
insecure=very
host=63.175.151.3 ;voip.essex1.com
fromdomain=63.175.151.3 ;voip.essex1.com
--
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: Mike Hammett
To: asterisk-users@lists.digium.com
Sent: Thursday, March 13, 2008 9:13 AM
have to do so the
outside world accepts emails from my Asterisk box? It is behind a NAT.
--
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk
line, or various programs
# that require network functionality will fail.
127.0.0.1 aiur.ics-il.net Aiurlocalhost.localdomain localhost
::1 localhost6.localdomain6 localhost6
--
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message
I am the ISP. ;-)
I'll have to look into that smarthost deal as there is no reverse DNS at
this time (my upstream's server times out).
--
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: Erik Anderson [EMAIL PROTECTED
?
--
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
___
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) exited non-zero on
'SIP/9826-ac087500' in macro 'stdexten'
== Spawn extension (macro-stdexten, s-NOANSWER, 1) exited non-zero on
'SIP/9826-ac087500'
--
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: Mike Hammett
or two
times I've had any issue, he has been quick to respond and took care of me.
--
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: Sigma Networks [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
*bump*
--
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: Mike Hammett [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, February 21, 2008 11:55 AM
I couldn't figure it out on my own. I tried to purchase a Smartnet for the
phone, but the original 7960 is not supported.
Is it technically possible and if so, what would it cost me to have someone
remote into my network and upgrade my SCCP 7960 to the latest SIP firmware?
--
Mike
I was doing it because of the volume on the server. It is very easy to miss
a message or 10 or 100 on a list of this traffic.
--
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: Tilghman Lesher [EMAIL PROTECTED]
To: Asterisk Users
That I am. I'll contact you off list.
--
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: Sigma Networks [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday
back from 63.164.210.14
Really destroying SIP dialog '[EMAIL PROTECTED]' Method: REGISTER
--
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: Mike Hammett
To: asterisk-users@lists.digium.com
Sent: Wednesday, February 20, 2008
It is, however, heavily trafficked and easy for someone to miss an email.
--
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: Benny Amorsen [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, February 25, 2008 3:44
I thought it was odd, but I've had other devices work properly with that
information.
--
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: Chris Bagnall [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion
*bump*
--
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: Mike Hammett
To: asterisk-users@lists.digium.com
Sent: Wednesday, February 20, 2008 4:52 PM
Subject: [asterisk-users] Coppercom and Asterisk
My provider has
messagecount: Failed to
obtain database object for 'asterisk'!
== Spawn extension (rwest, 300, 1) exited non-zero on 'SIP/2441-ac047f90'
--
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
___
-- Bandwidth and Colocation
It was my understanding that voicemail.conf referenced MySQL and not
asterisk.
--
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: Tilghman Lesher [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
PROTECTED]
Event: registration
Content-Length: 0
---
Aiur*CLI
--
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: Alex Balashov [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users
.com (63.164.210.14)
Change setting to use outbound Proxy
--
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
âmemcpyâ
combine_wave.c:991: warning: incompatible implicit declaration of built-in
function âmemcpyâ
make: *** [combine_wave.o] Error 1
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: Steve Johnson [EMAIL PROTECTED]
To: [EMAIL
--
File to patch:
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: Patrick [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, January 15, 2008 11:19 AM
Subject
Never mind, I got it. I needed a -p0
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: Patrick [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, January 15
$RIGHT
# eof
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
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http
Does what I have in the dialplan look right or am I way off base with being
able to use that script?
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: Steve Johnson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; Asterisk Users Mailing List
*bump*
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: Mike Hammett
To: asterisk-users@lists.digium.com
Sent: Tuesday, November 20, 2007 12:27 PM
Subject: [asterisk-users] e911
One of my providers has a different SIP
Then I could just make downstream-phones my current outbound context and
everything would do what I'm after. I got what you're saying.
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: Dave Miller [EMAIL PROTECTED]
To: Asterisk Users
?
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
___
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asterisk-users mailing list
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] Different Networks
On 9/7/07, Mike Hammett [EMAIL PROTECTED] wrote:
If it has nothing to do with Asterisk, then why does every other device
work
as its supposed to?
You never answered as to whether or not you're able to get out past
your gateway with any other network applications on your
*bump*
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: Mike Hammett [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, September 07, 2007 3:25 PM
Subject: Re
*bump*
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: Mike Hammett
To: asterisk-users@lists.digium.com
Sent: Thursday, September 06, 2007 10:05 AM
Subject: [asterisk-users] Different Networks
I have multiple
If it has nothing to do with Asterisk, then why does every other device work
as its supposed to?
An MGCP ATA routes out that interface.
A laptop routes out that interface.
That server traceroutes out that interface.
Asterisk doesn't link up.
-
Mike Hammett
Intelligent Computing Solutions
-scripted
accountcode=12
callerid=*
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
___
Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/
--Bandwidth
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
___
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asterisk-users mailing list
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I've been trying to send messages to the list for the past 24 hours, but they
just aren't going through.
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: Mike Hammett
To: asterisk-users@lists.digium.com
Sent: Wednesday
Agreed. This conversation is working just fine, but the important messages
I'm trying to get to go through aren't.
I've never had consistent success from posting to asterisk-users.
Asterisk-biz seems to work all of the time.
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics
*nods* I verified more than once and even copied + pasted to make sure.
Obviously my ping message went through, but my others have not.
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: Bill Andersen [EMAIL PROTECTED]
To: Asterisk
and I appreciate it much.
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: Jared Smith [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, September 05, 2007
I am looking for a gateway that has several FXS ports and uses IAX. I have
a need for 16 ports, but will accept 6 or 8 port gateways as well.
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com http://www.ics-il.com
/
/reginfo
msg msg.bypassInstantMessage=1
msg.mwi.1.subscribe= msg.mwi.1.callBack=299
msg.mwi.1.callBackMode=contact/
/msg
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com http://www.ics-il.com
Now that MCI and Verizon are one, they're probably on legacy MCI. MCI was
also the one that was doing the wholesale SIP pre-merger.
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent
Why would calls be coming in on the Guest IAX account?
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike Hammett
Sent: Monday, June 04, 2007 6:56 PM
To: 'Asterisk Users Mailing List - Non
to that customer were going to the default
context, despite the fact that I explicitly defined the context I wanted the
calls to go to in all entries in iax.conf.
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com http://www.ics-il.com
set myself up as a Linksys Partner, and have spent hour(s) on the
phone with them, but it still doesn't work. Ideas?
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com http://www.ics-il.com
___
--Bandwidth
Yeah, I was trying to have it match the caller ID with what they're dialing
so that I don't have a separate entry for every customer.
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rizwan
the voicemail system? If they call their own number, how do I get Asterisk
to recognize that and take them to the voicemail system?
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
___
--Bandwidth and Colocation
If it is easy, could you enlighten me? I have another thread on caller ID
matching, but I haven't received any positive responses.
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED
(NUM)},4,Hangup()
exten = 555*,1,NoOp(${CALLERID(num)})
exten = 555*,2,Hangup
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users
] SIP NAT
Mike Hammett wrote:
I hate SIP. The only reason I'm doing this is that its cheaper than
deploying the server to a colo facility. My provider has given me a
non-standard IP block, so I can't do typical routing.
I have Asterisk server - MT w\NAT - PPPoE - MT - Provider.
I
Discussion
Subject: Re: [asterisk-users] SIP NAT
According to sip.conf.sample the answer is...well, I guess you can look
in /path/to/src/asterisk/configs/sip.conf.sample and see for yourself.
Mike Hammett wrote:
If I have several local networks, can I specify that?
-Original Message
I hate SIP. The only reason I'm doing this is that its cheaper than
deploying the server to a colo facility. My provider has given me a
non-standard IP block, so I can't do typical routing.
I have Asterisk server - MT w\NAT - PPPoE - MT - Provider.
I setup a dst-nat on 5060 to the
clients?
Did you look at the nat setting sin sip.conf?
Do you have a static public address that can be routed to the Asterisk box?
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike Hammett
Sent: Thursday, March 29, 2007 11:52 AM
To: 'Asterisk Users Mailing List
?
(Benoit Panizzon)
15. Re: Cisco 30VIP Phone (Jason Parker)
16. SIP NAT (Mike Hammett)
17. Re: maximum simultaneous calls (Matthew J. Roth)
18. RE: SIP NAT (Alexander Lopez)
19. Re: Multi-line phones - Asterisk uses wrong callerid (Drew Gibson
I have a 501 with traditional power and a 301 with PoE. I rightfully assumed
that the traditional power from the 501 would work on the 301.
How do I get the PoE to work? Do I use the Polycom PoE cable in addition to
whatever PoE injection method I use? I have a Cisco PoE injector that works
on my
I was previously having an issue with a Polycom phone and Polycom support
said that Asterisk didn't play well with Polycom firmware versions 1.6.7 and
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
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 1.6.7 and
newer due to SIP compatibility issues. I believe I heard a lot
I enabled some more detailed debugging and logging as per someone else a few
posts ago and I saw that the permissions on MySQL were set incorrectly. I
granted all, but what are the least permissions this user should need?
How do I register to other servers? It seems to be ignoring the register
[EMAIL PROTECTED] asterisk]# cat res_mysql.conf
;
; Sample configuration for res_config_mysql.c
;
; The value of dbhost may be either a hostname or an IP address.
; If dbhost is commented out or the string localhost, a connection
; to the local host is assumed and dbsock is used instead of TCP/IP
PROTECTED]
Content-Type: text/plain; charset=ISO-8859-15
Brian Capouch wrote:
Mike Hammett wrote:
Could someone provide some steps for troubleshooting Realtime? I can't
see any signs that it's working. I followed and double-checked a few
different guides around the net, but haven't been able
I have multiple IP addresses on my box. My provider just changed my eth0 IP
off to another interface (lo:9) and a new IP on eth0. Nothing works anymore
because calls to the old IP address are being answered by the new IP
address. How do I straighten this out?
To: asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=us-ascii
On Wed, Feb 21, 2007 at 09:52:32AM -0600, Mike Hammett wrote:
I go to my Zaptel 1.4.0 folder and run ./configure; make linux26; make
install and I don't see any errors. This is out of my
--
Message: 14
Date: Wed, 21 Feb 2007 16:52:10 -0600
From: Mike Hammett [EMAIL PROTECTED]
Subject: [asterisk-users] Snom 320 password
To: asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=us-ascii
A client of mine has a Snom 320. Usually when he comes in each
Could someone provide some steps for troubleshooting Realtime? I can't see
any signs that it's working. I followed and double-checked a few different
guides around the net, but haven't been able to figure it out.
___
--Bandwidth and
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
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=windows-1252; format=flowed
Mike Hammett wrote:
Does anyone know why when calling out with an ATCOM AG-188 registered
with IAX (havent tried SIP), there is no ring.
Is this that you hear
Does anyone know why when calling out with an ATCOM AG-188 registered with
IAX (haven't tried SIP), there is no ring.
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: text/plain; charset=windows-1252
Mike Hammett wrote:
Im attempting to setup Asterisk 1.4.0 CDRs to use MySQL.
Modules show like cdr_mysql.so tells me it is loaded.
Reload cdr with MySQL started or stopped makes no difference in the
errors.
Ideas
I go to my Zaptel 1.4.0 folder and run ./configure; make linux26; make
install and I don't see any errors. This is out of my modprobe.conf:
install tor2 /sbin/modprobe --ignore-install tor2 /sbin/ztcfg
install torisa /sbin/modprobe --ignore-install torisa /sbin/ztcfg
install wcusb
A client of mine has a Snom 320. Usually when he comes in each morning, it
is asking him for a password. A power cycle brings it back to normal
operation. How do I troubleshoot this further?
--Mike
___
--Bandwidth and Colocation provided
I'm attempting to setup Asterisk 1.4.0 CDRs to use MySQL.
Modules show like cdr_mysql.so tells me it is loaded.
Reload cdr with MySQL started or stopped makes no difference in the errors.
Ideas?
___
--Bandwidth and Colocation
I currently have a customer that a previous employee setup with
Gentoo\Asterisk. I'm looking to migrate to AsteriskNOW. They have a custom
menu, which I would assume is easily replicable in AsteriskNOW. The only
other thing I can think of is the sound bites for the menus. Does anyone
have any
Where do I find more out in regards to the echo-cancelling component you
mentioned?
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, January 23, 2007 8:08 PM
Where do I find more out in regards to the echo-cancelling component you
mentioned?
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, January 23, 2007 8:08 PM
previous
times they've been slow to respond.
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
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constantly all day whereas we only did approximately a
half hour of testing.
I take the new phone back to my office and it now has 0% packet loss.
So, do I have two broken phones or is there something else wrong?
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
troubleshoot this issue?
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
___
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Has anyone actually gotten ASTTAPI to work? I
can't seem to get it to work, yet I have other TAPI setups (SNAP and xtelsio)
working fine. I have noticed that SNAP and Xtelsio act differently.
Etelescript is the application that will be calling TAPI.
Mike HammettIntelligent Computing
I don't know everything that's going on as someone
else has been working on the project, but it hasn't really been going anywhere,
so I had some questions.
We've got some Snom 320s with Asterisk 1.2.9.1 (I
believe). All was well (with a previous release), but the phones started
to get real
of the country that are only 55 ms away.
--
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
If you ping on the SIP port the message has to go through the
application layer - which takes some time considering it is an embedded
system with a small CPU. That part should be ok
I'm looking for an ATA\Voice Gateway that runs IAX
and has several ports (8 would be nice). I am looking to avoid devices
that use the same firmware as the ATCOM devices as I found them to be buggy (and
a PITA to find the proper update).
--Mike HammettIntelligent Computing
I currently have a single server with a few SIP and
IAX upstreams for origination and termination with IAX clients. I am
adding a second server that will have a much higher capacity and will be
handling a larger call volume. However, this second server is not going to
be geographically near
. Next step?
Geographically diverse servers, and I'm afraid of a call coming in to a
server that don't know what to do with it when another server knows exactly
what to do with it.
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: [EMAIL
http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Patterns
says:
! wildcard, matches zero
or more characters immediately
(only Asterisk 1.2 and later, see note)
Note: The exclamation mark wildcard, which is
available only in Asterisk 1.2 and later, behaves
Is there a non hardware limit to the limit of
concurrent connections that can go over a trunk?
So IAX trunking is preferred, can * do any other
trunking?
Mike HammettIntelligent Computing
Solutionshttp://www.ics-il.com
___
--Bandwidth and
Is there a non hardware limit to the limit of
concurrent connections that can go over a trunk?
So IAX trunking is preferred, can * do any other
trunking?
Mike HammettIntelligent Computing
Solutionshttp://www.ics-il.com
___
--Bandwidth and
instead of IP is its only issue.
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, February 09, 2006 3:11 AM
Subject: Asterisk-Users Digest, Vol 19, Issue 59
Send
Can TDMoE be used for non-voice
applications?
Can another box be setup with TDMoE on the other
side to dump it back out via T-1?
How does this compare with an off-the-shelf TDM
over Ethernet or IP device?
Mike HammettIntelligent Computing
Solutionshttp://www.ics-il.com
Reason I ask is I may have a non-voice T-1 replacement project going on and
I'm investigating my various options. Costs may be about the same for
turn-key and DIY.
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: [EMAIL PROTECTED
As evident in the SuperDial script and others based
upon groups, you can place a call into a group, which can have a limit on the
number of concurrent calls. Can a call belong to multiple groups?
IE: I have only a limited number of channels to upstream X.
Downstream Y is only paying me for
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