Softphone or hard phone doesnt matter if the service provider has the
right connections to provide E911 service. We are setting
up E911 compliance right now with our service. Its not as easy as just
updating the address, it take time, its not instant.
Kyle
Matt wrote:
Greetings to all,
Can
Is there something significantly different in 1.2.x sip.conf that would
prevent clients from registering with the server?
We installed 1.2.3 and copied over sip.conf from a production 1.0.9 box
and clients can not register.
Asterisk just replies with unauthorized.
Here is a sample from
I just did a quick office poll and everyone agreed if a party candidate
did this to them, they would vote for the candidate's opponent. The office
is rarely unanimous in political matters so this was a pretty interesting
result to me.
I'm pretty sure the feeling is universal.
Like I said
Here in Arizona I have gotten recorded calls from politicians.
There are apps to allow for this feature. If you want to know more
contact me offline.
Kyle
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On Tue, 2006-02-14 at 14:53 -0500, Matthew Crocker wrote:
Hello,
I have a Lucent MAX TNT, (DS-3, 672 modem ports, 28 PRIs). I'd like
to be able to direct an inbound fax call into my TNT, have it answer
the fax and send the image file over to Asterisk, or some other
system to deliver
Please ignore my last query about DTMF on ZAP, turned out to be an echo can
issue.___
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To UNSUBSCRIBE or update options visit:
We've setup automatic printing of faxes from asterisk (spool to a net queue
or direct to local printer), use the sendmail capability of emailing to an
executable (alias). Why send an image over to asterisk from the TNT?
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
On Tue, 14 Feb 2006, Matthew Crocker wrote:
I have a Lucent MAX TNT, (DS-3, 672 modem ports, 28 PRIs). I'd like to be
able to direct an inbound fax call into my TNT, have it answer the fax and
send the image file over to Asterisk, or some other system to deliver to an
e-mail address(s). I'm
Yes, we have and we just got rid of them because of it. We use higher
end phones like Polycom, Snom and Cisco now.
On 2/14/06, Ronald Voermans [EMAIL PROTECTED] wrote:
Hi all,
At our customer site i've installed one asterisk server with 20 Grandstream
GXP2000's. Firmware 1.0.1.9. When
I have the same problem on the same server...But I have just 3 PCI slots and the 3 are with digium cards. One of then is a TE406P with only one link connected, so there are a lot of red alarms.I'd like to have the blue light back on my server =) ..
Any one already solve this?On 11/17/05, Simone
Ok, got my last issue sorted, now another one. I can call out fine on this zap
channel which is connected to a carrier access bank 1 channel bank, using
asterisk 1.2 (aah2.0), I can call out, call other extensions such.. but I
cannot call into this zap extension, it always says the user is on
On Tue, 14 Feb 2006, Ronald Voermans wrote:
At our customer site i've installed one asterisk server with 20
Grandstream GXP2000's. Firmware 1.0.1.9. When someone dials the
customer, the receptionist picks up, and does an attended transfer (the
'grandstream way') to a collegue. Most of the times
after upgrade from 1.0.x to 1.2.x i cannot send faxes
my topology:
PSTN-wct4xxp-asterisk- -sip- ata (ht496,ht488,asus vp100) - samsung sf2500
fax
is there someone with this scenario? it is working?
thanks
(ip connectivity is good, codec alaw, 0% success)
log:
Feb 13 23:50:35 DEBUG[27914]
Yes. I have customers doing that all the time. They are service provider
specialize in political campages. Going back to topic, it he is only doing this
one time, why doesn't just find a service provider company to do it.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
First impressions telling me you want to check your phone settings. What
phone are you using and what are the config settings?
Mark
-Original Message-
From: Tomislav Parèina [mailto:[EMAIL PROTECTED]
Sent: Tuesday, 14 February 2006 9:01 PM
To: Asterisk Users Mailing List -
That's not a bad way to do it, but I'm also getting headline news. I
have something that works (once I like it I'll tweak the wiki with it),
but wouldn't mind something better.
Podcasts for weather by zip are available from 'pirateweather.com'.
These allow offloading the festival-like functions
Hi Giorgio:
That seems like a kind of a kludge. I would rather have the program work
right, than adding a work around. Dan of Littlejohnsconsulting has told
me of one problem in ARI that he is fixing but I do not understand how
it will fix the issue yet?? I will let you know as I find out
Hi Folks,
Can anyone give me some good recommendations for VoIP providrs that
support Asterisk PBX's? We're based in Georgia and I having a hard time
finding anyone
Regards,
Jim
PS - If you could CC me in on the reply I would greatly appreciate it!
jim(-A T-)linux-sp.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Tuesday, February 14, 2006 11:06 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Solution for 1 time blast of
200, 000
On Feb 14, 2006, at 4:33 PM, [EMAIL PROTECTED] wrote:
On Tue, 14 Feb 2006, Matthew Crocker wrote:
I have a Lucent MAX TNT, (DS-3, 672 modem ports, 28 PRIs). I'd
like to be able to direct an inbound fax call into my TNT, have it
answer the fax and send the image file over to Asterisk, or
We are using the same phones in our office with firmware 1.0.1.13 and
have no issues.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom Vile
Sent: Tuesday, February 14, 2006 4:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
What you are doing is changing the priority of packets that you are sending to the internet, you'll have to throttle the bandwidth for incoming packets (or better yet, have sprint do it on their router). What you are doing will help if you are getting bad calls when someine is uploading something
Ouch, Sprint wants $200 for the priviledge. I couldn't get approval for
that yet until we are closer to switching over more lines to voip.
Is it possible to do something equivelent or close without Sprint's help?
It seems like they are implmenting the equivelent of:
service-policy input
Well, I'm not real sure on whether I like the idea or not bug
Anyway, here is an app that I wrote for something similar to this. It
was for notifying customers of events,etc.
http://www.astpp.org/index.php?n=Misc.AutoDialOut
Darren Wiebe
[EMAIL PROTECTED]
Ron Senykoff wrote:
Hi,
Hello,The cards you referenced are BRI cards. We are located in México and do not use ISDN. Instead I could install a optional E1 interface to the D-1232 and purchase for example a TE110P card from Digium.But this solution will cost a lot more money and I am limited with my budget. I dont
Speaking of script launching, how would one fire-and-forget a script in
Asterisk? It seems that as it is currently configured, if the called
hangs up, the script aborts.
Thanks,
Bob McDowell
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bob
On Tue, 2006-02-14 at 17:26 -0500, Matthew Crocker wrote:
On Feb 14, 2006, at 4:33 PM, [EMAIL PROTECTED] wrote:
On Tue, 14 Feb 2006, Matthew Crocker wrote:
I have a Lucent MAX TNT, (DS-3, 672 modem ports, 28 PRIs). I'd
like to be able to direct an inbound fax call into my TNT, have it
On Tue, 14 Feb 2006, Matthew Crocker wrote:
On Feb 14, 2006, at 4:33 PM, [EMAIL PROTECTED] wrote:
nvfaxdetect / nvbackgrounddetect, along with spandsp+rxfax/txfax is your
answer.
http://www.voip-info.org/wiki-NVFaxDetect
I didn't think this was scalable to a PRI full of fax calls. I'm trying
To:
Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
I've got several issues with AGI/FastAGI
1. When an AGI script sends a command to Asterisk via stdin, why does Asterisk
block and not return a result until the command is complete? Specifically,
If you define a sip peer, wheather or not you put an entry in the field
OUTBOUND CID, if you
dial an external extension (let's say an extension on another asterisk
server, connected via IAX2 connection) the callerid
received by the foreign asterisk is device YOURNUMBER: i.e device 567
If you take
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Peter Corlett
Sent: Tuesday, February 14, 2006 9:01 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Solution for 1 time blast of
200,000 recorded calls
Ron Senykoff
Thanks for all your responses. The reason we would not go through a
provider is that I run Asterisk phone systems, we have access to
bandwidth, and I can do this myself for a fraction of the cost.
Cheers
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On Tue, 2006-02-14 at 11:34 -0500, Time Bandit wrote:
When agent tries to transfer a phone call (*2 - att transfer) he hangs up.
Why? When a phone call isn't from queue then att transfer works fine.
In features conf I have *1 for recording, *2 for att transfer and #1 for
blind. In
I'm trying to figure out how Asterisk decides how often it will send SIP
NOTIFY's to an ATA when a voicemail message is waiting for the user on the
server.
From watching, it seems to be completely random. Sometimes 10 seconds
apart, then 33 seconds, then 13 seconds, etc. Each time causes a ring
Hi All --
I've got a Polycom 601 with the sidecar unit all working with extension
hints and what Polycom calls the Buddy Watch feature. I can see the state of
extensions, but there seems to be a limit of 7 that I can monitor at any one
time.
I've put in a call to my distributor (this is how
Hi Freddi
Thanks for the reply. Neat ideas there, but a couple of issues.
1. Don't want to have to jump around between the FastAGI and the dial plan. Our
plan is to have NO customer data in the dialplan, as all data will be contained
within MySQL. We don't want to have to make _any_ edits to
http://connect.voicepulse.net
They support astrisk, with iax2 :)On 2/14/06, Jim Robinson [EMAIL PROTECTED] wrote:
Hi Folks,Can anyone give me some good recommendations for VoIP providrs thatsupport Asterisk PBX's?We're based in Georgia and I having a hard timefinding anyoneRegards,JimPS - If
sorry its really
http://connect.voicepulse.com/On 2/14/06, andrew matthews [EMAIL PROTECTED]
wrote:
http://connect.voicepulse.net
They support astrisk, with iax2 :)On 2/14/06, Jim Robinson
[EMAIL PROTECTED] wrote:
Hi Folks,Can anyone give me some good recommendations for VoIP providrs
Title: Firmware version 1.3.1 released for Aastra IP phones
Aastra Telecom has released SIP v1.3.1 firmware for the Aastra range of
IP phones (480i, 480iCT, 9112i and 9133i).
The firmware and release notes (no updated admin and user guides yet)
are available for download at:
www.interast.com
Kyle
andrew matthews wrote:
sorry its really
http://connect.voicepulse.com/
On 2/14/06, *andrew matthews* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
http://connect.voicepulse.net
They support astrisk, with iax2 :)
On 2/14/06, *Jim Robinson* [EMAIL
Mike Pollitt wrote:
I've got a Polycom 601 with the sidecar unit all working with extension
hints and what Polycom calls the Buddy Watch feature. I can see the state of
extensions, but there seems to be a limit of 7 that I can monitor at any one
time.
Polycom is apparently not going to
is there recomended source files for t.38 pass? latest cvs does not work
for me.
is it possible
publish working src?
turby
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On Sun, Feb 12, 2006 at 04:30:50PM -0700, [EMAIL PROTECTED] wrote:
I've seen a similar problem before. Span 3 was throwing errors for
(what seemed to be) no reason at all. After some testing it seemed
that the number of errors thrown on Span 3 had a relationship to the
temperature inside
--- [EMAIL PROTECTED] wrote:
Hello,
I have an IAX2 trunk like this running well with
IAX2 and SIP users mixed at
each side.
Runing like a charm :-)
Don't forget to add username definition from this
example.
To avoid too much load for your CPUs with
transcoding, tempt to have only
no cdp enable
-
1) Is it possible to have CAS framing with no R2?
You cannot have CAS and PRI at the same time. Thats for sure. Also CAS
goes hand-in-hand with R2. CAS means Channel Associated Signalling and
MFC-R2 is the Signalling
I've got a situation here that I thought was trivial. I have two
phones, and an asterisk box.
The first phone knows about g723, alaw and g729, as does the second phone.
sip.conf has allows for those codecs.
Now, within the dialplan I make a determination which codec will be used
(this is a
Hi,
I want have upgrade my ATA186 v2.15ms to H.323/SIP
ButI dont have a cisco acount yet
can some body help me with the ata18x-v2-16-030401a-1.zip file ?
THX .
weiming.
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From memory, it's really down to making the right selections in sip.conf
We did a large installation, with phones at the Head Office using g711
and phones at remote sites using g729.
Asterisk happily transcoded for us. Which was great.
PalH
On Wed, 2006-02-15 at 15:37 +1100, Lisa Wolf wrote:
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