SPA1001 is 1FXS only. SPA3102 is 1FXS + 1FXO... if you don't need FXO
why not get the SPA2102? It should be cheaper and you have the extra
port for future use.
On Sat, Mar 15, 2008 at 12:08 AM, Thermal Wetland
<[EMAIL PROTECTED]> wrote:
> That is good to know.
>
> I will be using the the device to
That is good to know.
I will be using the the device to connect a bunch of analog phones back to a
centralized server.
Since each office has only one ethernet connection do I want to:
1. Install a small switch to let a SPA1001 & PC use the one drop
2. Install a SPA3102 to let the PC share the one
Feel free to ping me off list. I've setup quite a few Cavtel PRI's with *.the
paperwork they asked you to setup?
Typically, they only send 4 digits.
Do you have the questionnare they asked you to fill out?
dwright at d2 - tech dot com.
From: [EMAIL PROTECT
its not a bad device - I have 2 problems with it. It doesn't do echo
cancellation very well & is particularly badly matched to the PSTN here in Oz.
Hint: keep it well cooled - echo goes up badly when its hot& it runs very hot
if there is no ventilation. I use the 3102 to bridge a mythtv box i
That is awesome. I don't know why the manual doesn't mention that.
I want to have the device use a static IP & the computer use DCHP from a
central DHCP server...sounds like it won't be a problem.
Thanks.
On Thu, Mar 13, 2008 at 8:14 PM, W.Kenworthy <[EMAIL PROTECTED]> wrote:
> sipura 3102 set
Matt Riddell wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Tzafrir Cohen wrote:
>> Let's be more specific here, folks:
>>
>> What version numbers?
>>
>> Asterisk, spandsp, agx-addons / rx-tx-fax?
>
> Asterisk: yesterday's 1.4 SVN
> SpanDSP: tried with pre 15, 16 and 18
> AGX-Addons
Thanks. I am in Philly. I may have to configure the extensions.conf well to
pass the incoming channels.
On 3/14/08, Steve Totaro <[EMAIL PROTECTED]> wrote:
>
> On Fri, Mar 14, 2008 at 8:13 PM, broadband Voice
> <[EMAIL PROTECTED]> wrote:
> > I had Cavalier turn up a T1 PRI. How can I put in the DI
On Fri, Mar 14, 2008 at 8:13 PM, broadband Voice
<[EMAIL PROTECTED]> wrote:
> I had Cavalier turn up a T1 PRI. How can I put in the DIDs to direct to
> Asterisk. Here is a log
>
>
>
> Zaptel Tool (C)2002 Linux Support Services, Inc.
> ⤠T2XXP (PCI) Card 0 Span 1
>
John Meksavan wrote:
> Asterisk Users,
>
> I am running Asterisk-1.4.11 on a Debian "Etch" system. On an
> occasion, when customer calls into my Asterisk Box, I get this error
> messagefrom Asterisk "CallerID returned with error on channel Zap/3-1" ,
> causing all my zap channels to be busy.
I had Cavalier turn up a T1 PRI. How can I put in the DIDs to direct to
Asterisk. Here is a log
Zaptel Tool (C)2002 Linux Support Services, Inc.
⤠T2XXP (PCI) Card 0 Span 1
ââ[3;10Hâterfaces â
â[3;37Hâ â â
Mark Hamilton wrote:
> I don't think the link that Lee gave works.
>
Oh boy, you were talking about the link to download the software and I
completely misunderstood. My mistake, the link is fixed to download the
software.
Here's the direct link:
http://www.datatrakpos.com/pos/datatalk/downlo
Mark Hamilton wrote:
> I don't think the link that Lee gave works.
>
>
> Also, I wrote a Windows based utility for viewing AMI packets and testing
> AMI
> commands. It's Freeware:
>
> http://www.voip-info.org/wiki-Asterisk+GUI#OperatorManagerInterfaces
> Look for Manager API Test Utility
>
>
zap show channel 3 is giving me that I am using
relaxdtmf (actually I placed the relaxdtmf before the
assiging for the channel -> 3).
I increased the volume rxgain and txgain and now the
voice is hearable good, but still the detection for
the dtmf has duplication.
But, if I let the voice message
Asterisk Users,
I am running Asterisk-1.4.11 on a Debian
"Etch" system. On an occasion, when customer calls into my Asterisk Box, I get
this error messagefrom Asterisk
"CallerID returned with error on channel Zap/3-1" , causing all my zap
channels to be busy. So, I cannot make any calls i
On Fri, 14 Mar 2008, Ken D'Ambrosio wrote:
> Hi, all. I'm trying to do some rudimentary testing of an Asterisk
> system, but, for various reasons, I have to do this covertly, which
> means I'm paying out-of-pocket. So I'm looking for somewhere that will
> do *cheap* SIP and/or IAX termination
I would just try parsing the message in my code to make it work. I know this is
not a full proof solution but is ok for now, till I get something better.
Regards,
Sanjay.
- Original Message -
From: "Mark Hamilton" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Disc
I don't think the link that Lee gave works.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: March 14, 2008 5:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Logs for Call generated by
Thanks Lee, Will try to match on Parameter received in message.
Regards,
Sanjay.
- Original Message -
From: "Lee Jenkins" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Friday, March 14, 2008 11:27:51 PM (GMT+0530) Asia/Calcutta
Subject: Re: [a
Hi, all. I'm trying to do some rudimentary testing of an Asterisk system,
but, for various reasons, I have to do this covertly, which means I'm
paying out-of-pocket. So I'm looking for somewhere that will do *cheap*
SIP and/or IAX termination, preferably with at least two simultaneous
calls, and
You need to download and install the system from the iso image. This puts
everything into place on the system.
On 3/14/08, Jeff Johnson <[EMAIL PROTECTED]> wrote:
>
> I downloaded Druid from the svn, but can't seem top get the install.pl
> to run. It appears that the svn is assuming that Asterisk
I downloaded Druid from the svn, but can't seem top get the install.pl
to run. It appears that the svn is assuming that Asterisk is already
installed. Any help would be appreciated.
Jeff
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Totaro
Sent
Actually, relaxDTMF helped me.
I realized I was putting it in the wrong location in the zapata.conf
to get it to apply to my channels.
'zap show channel #' will tell if you if RelaxDTMF is enabled on a channel.
-- James
On Fri, Mar 14, 2008 at 10:55 AM, bilal ghayyad <[EMAIL PROTECTED]> wrote:
>
On Fri, 14 Mar 2008, Eric Rees wrote:
> I am having a strange issue with setting the incoming caller id on the
> latest version of TrixBoxCE. Right now I have it setup with a
> cross-over T1 cable to our production Asterisk (1.0.9) box and from the
> Trixbox we can send and receive calls just
Help needed, please.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton
Sent: March 13, 2008 2:16 PM
To: [EMAIL PROTECTED]
Subject: [asterisk-dev] Call failed, reason 0 explanation.
Hello,
I understand that Asterisk interprets SIP error codes as 'reason'. We
Hello,
Didn't get much help on asterisk-dev, so here it is.
Please help.
Thanks.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton
Sent: March 13, 2008 2:10 PM
To: [EMAIL PROTECTED]
Subject: [asterisk-dev] Hardware and CentOS tweaks.
Hello,
We're
I am having a strange issue with setting the incoming caller id on the latest
version of TrixBoxCE. Right now I have it setup with a cross-over T1 cable to
our production Asterisk (1.0.9) box and from the Trixbox we can send and
receive calls just fine. The problem I am having is that if a num
[EMAIL PROTECTED] wrote:
> I am generating an outbound call through the Manager API and bridging it to
> an internal Extension, my problem is I am not able to find the logs for the
> call generated by the Manger API, Since on the same Asterisk server there are
> many users connected and I am rec
Hi Eric;
I decreased the gain (although I was need to increase
it) and the problem somehow resolved partially (much
more better), but now I have a problem in the voice
volume, it is weak. If I increased the volume, then
the duplicatin in the DTMF for the first entered digit
(specifically) will app
[EMAIL PROTECTED] wrote:
> Is there way to get the logs of the call generated by Manager API, or is
> there some other way to achieve same scenario so that I can get the status of
> the call generated by me.
>
>
> Actually I have a scenario where I have to call customers and play a message,
>
On Fri, Mar 14, 2008 at 12:10:21PM -0400, Andreas van dem Helge wrote:
> What's your point? It's configurable on the 2102 as
> switch/nat/autodetect, ht496 as nat/switch. If you have such ISP and
> want to use the device as a router just manually set the NAT option.
My point -- I thought it was fa
I know about them... KCL.net in Miami does (did?) the same assigning
10.x address for basic home connections...
What's your point? It's configurable on the 2102 as
switch/nat/autodetect, ht496 as nat/switch. If you have such ISP and
want to use the device as a router just manually set the NAT opti
Asterisk receives T.38 RTP packet from one SIP peer and sends it to
the other SIP peer, it does not process the packets.
By your argument I can't do T.38 @ 1440bps unless the manufactures of
the Ethernet cable, switch, router, keystone jacks, network cards,
CPU, RAM, etc all paid for the royalties
On Fri, Mar 14, 2008 at 02:32:54AM -0400, Andreas van dem Helge wrote:
> Linksys SPA2102 does. It even has the option to auto-detect so if it
> is assigned an RFC1819 address it will act as a switch and otherwise
> just as a NAT router.
Clearly, someone neglected to tell them about ISPs like Rose.
On Fri, Mar 14, 2008 at 9:01 AM, Rizwan Hisham <[EMAIL PROTECTED]>
wrote:
> I dont know about IAX, but for SIP users you can use the function
> SIP_HEADER(headername) to get the information u need from the sip packets.
> for example you can use SIP_HEADER(From) which will give you the From header
Ricardo Carvalho wrote:
> I made some tests with FAX in Asterisk 1.4 using T.38 between two ATAs
> connected to legacy FAX machines, and realized that only SIP can make
> passthrough in the server while RTP go direct between endpoints. Is it
> possible for RTP data stream also to make passthroug
Ok,
Making progress finally came up with a useful goolge search and found
this..
http://click.rho.cc/Blog/?p=10
I'll try it when I get home and see how far I get.
Looks promising.
Adrian
On 14 Mar 2008, at 14:02, Remco Barendse wrote:
> On Fri, 14 Mar 2008, Adrian Merwood wrote:
>
>> I
I made some tests with FAX in Asterisk 1.4 using T.38 between two ATAs
connected to legacy FAX machines, and realized that only SIP can make
passthrough in the server while RTP go direct between endpoints. Is it
possible for RTP data stream also to make passthrough in Asterisk?
Thanks,
Ricardo Car
Is there way to get the logs of the call generated by Manager API, or is there
some other way to achieve same scenario so that I can get the status of the
call generated by me.
Actually I have a scenario where I have to call customers and play a message, I
do not want to send messages to Manag
On Fri, 14 Mar 2008, Adrian Merwood wrote:
> In my asterisk (Trixbox) server I would like to be able to dial
> numbers from my address book using HUD or the SIP client on my 3G
> phone using numbers in this format.
>
> On asterisk I would like to strip of the + and replace it with an
> internation
I am generating an outbound call through the Manager API and bridging it to an
internal Extension, my problem is I am not able to find the logs for the call
generated by the Manger API, Since on the same Asterisk server there are many
users connected and I am receiving lot of Events back, not ab
How would you relay on Google Apps, as Google requires SSL or TLS
authentication?
How can I configure sendmail to do this?
Actually, sendmail is trying to send email directly, and I get the response
below. I´ll now try Mike Hammett´s solution.
Thanks,
Felipe Trevisan
*Message contents*
The
Mindaugas Kezys wrote:
> Hello,
>
> Higher speeds then 9600kbps are not permited by patents.
>
Would you care to name one that prevents 14,400?
> Regards,
> Mindaugas Kezys
> http://www.kolmisoft.com
> MOR PRO - Advanced Billing Solution for Asterisk PBX
>
>
> -Original Message-
> From:
Hello,
Higher speeds then 9600kbps are not permited by patents.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR PRO - Advanced Billing Solution for Asterisk PBX
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andreas van
dem Helge
Sent: Friday, Ma
On Fri, 14 Mar 2008 13:06:27 +0200, "love U.all"
<[EMAIL PROTECTED]> wrote:
> ur mail erver isn authorized to redirect mails say for example to hotmail
> coz msn deal with it as spam
MSN and hotmail are not a reference in anything related to Internet e-mail.
Unless, that is, you're considering ho
--- Atis Lezdins <[EMAIL PROTECTED]> wrote:
> Hmm, didn't knew that queue_log can be written into
> MySQL..
Asterisk 1.6 beta has that through realtime.
But I'm using a custom import script in earlier
Asterisk versions.
> Is callid in queue_log the same uniqueid?
yes.
> You can do
> somethin
Horwich IT Services (Godwin Stewart) wrote:
> On Fri, 14 Mar 2008 07:29:33 +, Adrian Merwood
> <[EMAIL PROTECTED]> wrote:
>
>
>> Secondly (in the future) I would like to strip off certain country
>> codes and replace them with a local dialing prefix.
>>
>> Can anyone help me figure ths out
Dont know if two speakers can be used differently on any sip phone I
have seen, but you can use this:
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ChanSpy
in 'whisper' mode to implement a similar functionality. The group can
sit on their desk and listen!. Or have another speaker
ur mail erver isn authorized to redirect mails say for example to hotmail coz
msn deal with it as spam> Date: Fri, 14 Mar 2008 02:25:37 -0400> From: [EMAIL
PROTECTED]> To: asterisk-users@lists.digium.com> Subject: Re: [asterisk-users]
Mail Server> > tail /var/log/mail or /var/log/maillog> > On
I dont know about IAX, but for SIP users you can use the function
SIP_HEADER(headername) to get the information u need from the sip packets.
for example you can use SIP_HEADER(From) which will give you the From header
containing the IP address of the caller. You will have to apply regex on it
to ex
Hi,
I just encountered a simple but strange problem. I am using 2 sip phones to
call each other. Whenever i make a call, using softphone or ata, ali only
shows the CallerID(name) and not the number. I have no idea why it does not
show the number. I have tried various things but none have worked. Th
Hi!
Paul Hales wrote:
>
> I just installed Asterisk 1.6 beta5 and moh is not working - is there a
> trick? Or is something wrong with my system?
>
This bug already fixed, you can check latest 1.6 branch or try to use
1.6 beta4. This version must not have this issue.
--
Best regards,
Ig
On Wed, Mar 12, 2008 at 6:43 PM, Joshua Wilson <[EMAIL PROTECTED]> wrote:
> I have recently noticed that druid @ http://www.voiceroute.org has created
> an open source edition of their platform. I downloaded it today and
> installed it on a play system where I have about 20 ip phones ranging from
On Fri, Mar 14, 2008 at 03:51:25PM +1100, Paul Hales wrote:
>
> I just installed Asterisk 1.6 beta5 and moh is not working - is there a
> trick? Or is something wrong with my system?
Could you please be more specific? An trace / config snippets of
whatever does happen?
--
T
Hi,
Cory Andrews had a last minute travel plan change and even the Snom M3
won't help on a plane. We'll reschedule Cory. We're open to any and
all asterisk-related ("asterisk" is a trademark of Digium) discussion
or VoIP related discussion this week. If you're into VoIP, and you
must be if you're
On Fri, 14 Mar 2008 07:29:33 +, Adrian Merwood
<[EMAIL PROTECTED]> wrote:
> Secondly (in the future) I would like to strip off certain country
> codes and replace them with a local dialing prefix.
>
> Can anyone help me figure ths out?
This might get you started: http://howto-pages.org/ast
H,
Just a quick question that has been bugging me for a while.
Most of my address book phone numbers are stored in the format:
+
i.e. +
In my asterisk (Trixbox) server I would like to be able to dial
numbers from my address book using HUD or the SIP client on my 3G
phone usi
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