Hi,
I prevoiusly has asterisk on a public static ip and had a phone from
a different location registering to the asterisk box. But now we have
dropped the previous connection and the current connection has a
dynamic ip. Is there any way for the phone to register to now-dynamic
ip addressed asteris
Looking to install asterisk for a client and was shopping
around for prices for 6 POTS lines with or an integrated T1 with voice and
data. I called up Sprint and I told
the sales rep that there was going to be a Phone system she said that they
would have to install “key” rotary lines and t
, Manjit Riat <[EMAIL PROTECTED]> wrote:
> I am going to buy some IP phones from them but I sent them an email couple
> of weeks ago and got no reply. Has anyone ordered anything from them? Any
> other places that I can buy from? Sorry if it's a wrong post.
I would give them a
Hi,
I was going to
order the T100P but it is replaced by TE110P. On further reading the TE110P
does not need an external router (The one that separates the data from pstn lines ?). Has anyone got it
configured? And on the wiki it says that the drivers
for some distros don't exist yet. Is
List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] VoipSupply.com
Manjit Riat wrote:
> I am going to buy some IP phones from them but I sent them an email
> couple of weeks ago and got no reply. Has anyone ordered anything from
> them? Any other places that I can buy from? Sorr
I am going to buy some IP phones from them but I sent them
an email couple of weeks ago and got no reply. Has anyone ordered anything from
them? Any other places that I can buy from? Sorry if
it’s a wrong post.
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Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] 8+
line receptionist only setup
Can your receptionist handle 6 active
conversations? Once she transfers the call, it would disappear from those 6
lines.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Manjit Riat
S
Behalf Of Manjit Riat
Sent: Sunday, May 08, 2005 5:09 PM
To: 'Asterisk
Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] 8+ line
receptionist only setup
Hi,
We are looking towards a 8+ CO line setup (20
extensions) in our office but we do not want an IVR(aut
Hi,
We are
looking towards a 8+ CO line setup (20 extensions) in our office but we do not
want an IVR(auto-attendant) feature. All incoming will
be answered by a receptionist. I have read the multi-line configuration for cisco 7960 thread in this list but that way I believe we
could onl
TED]
Sent: Thursday, May 05, 2005 12:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Polycom Images
Manjit Riat wrote:
> Out of curiosity what's the reason? Why would they not sell phones to
> asterisk users? Do they not trust asterisk or t
List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Polycom Images
If you want to purchase them from a distributor as a dealer it is best to
not mention asterisk as they will deny your certification. Use PingTel as
your way in.
- Original Message -
From: "Manjit Ria
I did a make webvmail and I get
the following error on redhat 9.0
No HTTP directory
make: *** [webvmail] Error 1
I have the perl-suidperl rpm
installed and apache installed
Thanx.
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Asterisk-User
Couple of months ago someone had posted to the list that polycom won't
support/provide to customers using polycom phones with asterisk. Is it still
the same way? If not then where do you get the images from coz I am looking
to pick up a polycom.
___
As
Well having one line provisioned on the cisco 7960 gives me two incoming
lines anyways (call waiting).how can I provision the other lines with the
same extension (through the phone or asterisk)?
Or is it something like this
Exten => 101, 1, Dial(SIP/ciscoline1&SIP/ciscoline2&SIP/ciscoline3,20,tr)
Hi,
This may be a dumb
question but I know how to provision lines but what is the use for them. Right
now I just have one line provisioned on my cisco 7690
and I get all incoming calls on that line and make calls on that too. Additional
lines may be mean additional extension numbers. Bu
Not too sure but I have heard of Fry's Electronics using asterisk.
-Original Message-
From: Denis Galvão - iSolve [mailto:[EMAIL PROTECTED]
Sent: Tuesday, April 19, 2005 11:01 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk Business Case - Who is using it!?
Hi
Hi,
I am looking
around for phone for a receptionist that has a list of extensions and so forth
like the cisco expansion module. But I had read
somewhere that the expansion module does not work with asterisk. Please
confirm.
Thank You
--
__
ctually be cheaper than 6-7 POTS
business lines), and a single T100P card...and you don't get the myriad
problems reported on this list involving the TDM cards.
Who is doing the hunting on your main phone number? Or do you not have a
main phone number in this install?
Greg
Manjit Riat wrote
line installation.
They should have the answer:
http://local.sprint.com/home/local/contact/contact_information.html
On 4/13/05, Manjit Riat <[EMAIL PROTECTED]> wrote:
>
>
>
> We are going to be doing an asterisk install with 5-7 lines. So we are
looking to get
We are going to be doing an asterisk install with 5-7 lines.
So we are looking to get two TDM04B cards. Now I believe when you
get your telco(Sprint, etc.) to install the lines they basically just
leave the wires without jacks. Am I right? If so, then can we ask them to
install the jacks or
Hi,
I have a
setup asterisk with 4 extensions with asterisk & 3 extensions on public IP
and one extension behind a NAT. All works great. Today I tried installing X-lite for my friends in Dubai (ISP uses a proxy server/NAT) and I get
this error
Apr 6 10:51:41
NOTICE[1213]: chan_sip.
facts are correct.
Manjit Riat wrote:
> Does any know if SIP ports are blocked in dubai (UAE)? Anyone in UAE
> using FWD or similar services and connecting to SIP proxies in US?
>
>
Does any know if SIP ports are blocked in dubai
(UAE)? Anyone in UAE using FWD or similar services and
connecting to SIP proxies in US?
Thanks.
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The partner list shows digium as one of their partners. So under GPL they
have to provide the source code for the app.
-Original Message-
From: Brian Dingman [mailto:[EMAIL PROTECTED]
Sent: Tuesday, February 01, 2005 11:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sub
Did you use insecure=very in sip.conf ?? or did you use qualify ?
-Original Message-
From: Luki [mailto:[EMAIL PROTECTED]
Sent: Monday, January 31, 2005 8:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Broadvoice problems
What registration
I am thinking of dumping broadvoice
so I need some other VoIP providers that have a las vegas DID and a service better
than broadvoice.
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I somehow got my broadvoice to work
with a weird configuration but now I have to restart my asterisk server every
24 hours otherwise all incoming calls are sent to VM.
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Does anyone know of a speech recognition module (like say
yes or no, or numbers) I guess due to the complexity of speech recognition it
might just be found in commercial applications or am I wrong like always?
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Just installed festival from source and the voice is very
jittery and I get this a lot in the asterisk CLI (at least once on every call)
NOTICE[3236]:
rtp.c:430 ast_rtp_read: RTP: Received packet with bad
UDP checksum
Maybe the packets are malformed so I get the jittery sound.
BroadVoice? If so how do you tell which
number is ring in on or which line to dial out on I have on line
with him now and would like to add two lines..
Thanks, David.
On Thu, 2005-01-27 at 14:14 -0800, Manjit Riat wrote:
> I had a lot of problem with them to set up..
>
> Yo
I tried everything and only got that configuration with all bv servers
listed to work.
-Original Message-
From: Luki [mailto:[EMAIL PROTECTED]
Sent: Thursday, January 27, 2005 8:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Stumped by Broa
I had a lot of problem with them to set up..
You need to register to sip.broadvoice.com
And need to have all of their four servers to listen to incoming calls as
ony one can send it in..
Just posted my config two days ago.
http://lists.digium.com/pipermail/asterisk-users/2005-January/085736.htm
Just installed festival and I get this error in asterisk CLI
SIOD ERROR: wrong type of argument to car
: wholeutt
Jan 27 11:46:09 WARNING[25917]:
app_festival.c:444 festival_exec: Festival returned
ER
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any explanation? what is your register => statement?
I can make calls, but can not recieve them. When i use sip debug, it
seems like asterisk
is getting something from broadvoice, but it fails to ring my phone.
--Dalon
On Wed, 26 Jan 2005 11:09:05 -0800, Manjit Riat <[EMAIL PROTECTE
scussion'
Subject: RE: [Asterisk-Users] A
working BroadVoice config example
Can you also post your
“extensions.conf” so I can compare. I have been trying to get
this working for about a week.
Chris
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Manjit Riat
Se
I am sorry but I think you're mistaken about heavy traffic.
This is my stats (from MySQL)
Query statistics: Since its startup, 746,184,562 queries have been sent to
the server in 16 days..
And I really don't consider that high load.
-Original Message-
From: Vahan Yerkanian [mailto:[E
Yeah Nabeel I guess it was yours. Thanx
-Original Message-
From: Nabeel Jafferali [mailto:[EMAIL PROTECTED]
Sent: Wednesday, January 26, 2005 12:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] A working BroadVoice config example
> [bv-in-1]
I finally got my incoming and outgoing to work on Broadvoice with a configuration file that is no where close
to the one given by them.
Here it Is (sip.conf).
For others who have a working config could u please
share so that I could compare. Thank You
[broadvoice]
type=friend
Is there any way to change the outgoing caller id on BroadVoice
I have tried SetCallerID(Name ) but that does not work
Thanx
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line, most likely at your
firewall/router or your SIP messages contain a non routable address so
BroadVoice is sending your RTP stream to a bad destination. You will
need to include your SIP messages and a network topology if you would
like someone on the list to chime in.
-John
Manjit Riat
Once you compare Postgress and MySQL you will never want to go back to
MySQL.
-Original Message-
From: Robert Augustyn [mailto:[EMAIL PROTECTED]
Sent: Wednesday, January 26, 2005 10:07 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] ANNOUNCEM
I finally got my incoming and outgoing working but outgoing
I cannot hear the called person, but the called person can hear me.
On incoming everything works perfect.
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h
patch not being applied. I had to get the Asterisk
source 1.0.3 then apply the patch and recompile. The reason you are
seeing the 404 is that asterisk is not registering to their service
properly, and that is what the patch fixes.
-John
Manjit Riat wrote:
> Is the Broadvoice service up
Is the Broadvoice service up? I
just signed up with them and started receiving calls in no time but could not
make calls. And after a few minutes I cannot even place calls.
register => [number]:[EMAIL PROTECTED]
[broadvoice]
type=peer
fromuser=[number]
host=proxy.lax.broadvoic
Which would you recommend as far and quality and pricing to
connect to asterisk (including DTMF issues)/
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Thank You for your replies but I should have asked the question in a better
way.
I currently do have a Vonage line as a residential line and keep switching
it back and forth between the home phones and vonage. So I want to get an
additional line just for asterisk.
For that I needed to know if the
Hi,
I am thinking of
signing up with voice pulse connect to connect to my asterisk server and using
it as a regular line. Is it good? Or should I go with vonage
or others ?
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h
Title: RE: [Asterisk-Users] Music On-Hold problem
It should work right off the install..
Make sure you have MPG123 installed and running.
_
From: Computer Onsite Support [mailto:[EMAIL PROTECTED]]
Sent: Sunday, January 23, 2005 3:10 PM
To: asteris
Just got my 7960 that I picked up
from ebay. It looks like it
has a SKINNY image instead of SIP.. where
can I get a SIP image ?
And how do I unlock the phone.. it is stuck at configuring IP, configuring CM based on the
old settings.. tried **# but nothing happens !
__
, Manjit Riat wrote:
> Just got a headset for testing asterisk and am using X-Lite. I plugged
> in the headset into the headset jack and is there any way to configure
> X-lite to use the headset instead of the speakers? Or will I have to
> plug the headset in the speaker jack ?
Manjit
Just got a headset for testing asterisk and am using X-Lite. I plugged in the headset into the headset jack and is
there any way to configure X-lite to use the headset
instead of the speakers? Or will I have to plug the headset in the speaker jack
?
___
Thank you everyone. Makes a lot of sense...
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Thursday, January 20, 2005 7:40 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] E911 Testing !
Joe Gr
I am testing IAXTEL and routing 800 number to them.. Sometimes
the call goes through and the other times it get the following error.
WARNING[20502]: chan_iax2.c:1477 attempt_transmit:
Max retries exceeded to host 69.73.19.178 on IAX2/iaxtel/3 (type = 6, subclass
= 9, ts=631, seqno=1)
I believe the 911 is a serious issue if one does an asterisk
installation in an office. How do you test 911? Won’t they arrest you or
something for dialing 911 for no reason and talking to one of their agents who
could have taken a more important call?
On the other hand what an emergen
That was a really nice description... Can you do 1-14 and I'll do 15 and
16??
Just kiddin.
-Original Message-
From: Ty Carter [mailto:[EMAIL PROTECTED]
Sent: Wednesday, January 19, 2005 10:58 AM
To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial
Discussion'
Subject: RE
My router (1605R) currently does not support QoS. Is there
any open source software available so that I can set one up before the router?
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I believe there is a variable tos in sip.conf in the general context which
can be a keyword or a numeric value. Hope that's what you're looking for.
-Original Message-
From: Dale [mailto:[EMAIL PROTECTED]
Sent: Tuesday, January 18, 2005 10:45 AM
To: Asterisk Users Mailing List - Non-Com
How do I configure Asterisk to accept incoming SIP URL calls?
(sip:[EMAIL PROTECTED])
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To UNSUBSCRIBE or update options
I want to set up my asterisk to receive SIP calls using the
URL [EMAIL PROTECTED] . I have my own DNS
server but would like know what entry goes into it as I have never set up SRV records
before. (if it matter it is a BIND dns
server).
thanx
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