I chaged to use from 'rxfax' to 'txfax' and I succeeded to receive the
file from * to the FAX under HT488(firmware 1.0.1.2).
My OS is 2.6.11-1.27_FC3smp, CVS-v1-0-04/20/05 with ztdummy.
spandsp is fun!
I made a file:
Channel: SIP/4881
MaxRetries: 0
WaitTime: 20
Application: txfax
Data:
Hi,
Recently, I've just installed asterisk along with AMP..
Everything seems to work fine, just when I tried to record my voice via
ivr, asterisk won't play the file if I call it.
When I test by dialing *99, the record is played, but when I call
straight to the digital receptionist, it just
On Fri, 27 May 2005, Waldo Rubinstein wrote:
I'm planning on setting up some remote agents and before doing so, I
did some simple PING tests to measure latency. The average latency I
got was 250ms. Does anyone have experience in terms of quality of
calls when there is such high
On Sat, May 28, 2005 at 01:51:56AM +0200, Stefan Gofferje arranged a set of
bits into the following:
Hi folks,
I'm trying the latest mayday chan_sccp and found it difficult to have
more than three speeddials. I have defined only one line but can't get
more than three SDs. On console, *
Hello *'s,
I have question regarding Quintum Tenor AXT800 VOIP gateway can anyone
integrate it with asterisk if anyone what is the scenerio i have
scenerio which is quite simple but i am confused about it whether it
is possible or not :
I integrate it with asterisk for interanet no PSTN at all
I recently installed [EMAIL PROTECTED] to give it a shot, really my first
time using asterisk. I've got my sipura 3000 working fine with
asterisk as an extension and I can dial out just fine, but incoming
calls to my toll-free DIDs are not working. I get the following
error:
May 28 04:57:47
Hello everybody.
I have an customer asterisk 1.0.5 running well since 3 monthes, 2 TDM
cards 4 FXO, 4 FXS. Since one week, unable to pass call between Zap and
Sip getting the exception on 15, channel 1
The * box is connected to an eads PBX and it seems that failure started
when they make
Hello list,
I am glad to announce that XC-AST version 0.9.0 is out today.
New functionalities include:
* Though not yet available to the end user, this release inclued the basis
of the Outbounds Call Manager that will be released for 1.0. If you update
from a previous version, have a look at
I've submitted to Julien http://chan-sccp.sourceforge.net/ my patches
the diff file is attached to this mail
Apply it to the CVS head
cvs -d:pserver:[EMAIL PROTECTED]:/cvsroot/chan-sccp login
cvs -z3 -d:pserver:[EMAIL PROTECTED]:/cvsroot/chan-sccp co
-P chan_sccp
it does include:
- added
Great log. You carefully stripped out anything that might possibly be
useful for diagnosise. However, the answer will be the same in any case
- See http://www.soft-switch.org/foip.html
Regards,
Steve
Zen Kato wrote:
I chaged to use from 'rxfax' to 'txfax' and I succeeded to receive the
I am sure this has been addressed somewhere else, but I
havent found it
Is there a way to make multiple extensions have their MWI
light flash, all for the same common voicemailbox? And to make it even
trickier, what if its a mix of SIP and ZAP channels?
Id like to be at my
Hi
Can anyone provide me with a Manchester (0161) UK DID number, preferably
IAX2 but SIP is ok too, that I can use for my incoming calls? Call volume
will be low.
The critical thing is that DTMF must be correctly passed 100% of the time,
unlike Sipgate, my current (free) provider, whose DTMF
On Saturday 28 May 2005 14:31, Tom Fanning wrote:
Hi
Can anyone provide me with a Manchester (0161) UK DID number, preferably
IAX2 but SIP is ok too, that I can use for my incoming calls? Call volume
will be low.
Yeh, Sipgate's price is good (hey you can't argue with £0 setup and £0 per
Thanks for your quick reply.
Do you have experience with them? Does their DTMF work properly? Any chance
of posting the juicy bits of your config files if you use them please?
Cheers!
Tom
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Gavin
SYED ADEEL ALI wrote:
I've configured SIP softphone to work with asterisk n it's working
fine.
but i m unable to connect my netphone IP phone I've connected my phone
to LAN and assigned an IP address to it but how can i make call... plz
tell me step wise.
What model ?
Greetings to all!
Please, I want to buy a SIP or iax2 supporting HARDWARE phone which can
directly be connected to my Asterisk PBX from Dubai. and your
recommendations are highly appreciated.
Thanks
Kumara
___
Asterisk-Users mailing list
Greetings,
I am new to all this VoIP stuff and have been having a bit of a hard
time getting my soft phone working as a SIP client thru Asterisk. I
apologize to start off with such a simple question and hope it's ok to
post this and see what others have done.
THE GOOD NEWS:
I have successfully
Greetings,
I am new to all this VoIP stuff and have been having a bit of a hard
time getting my soft phone working as a SIP client thru Asterisk. I
apologize to start off with such a simple question and hope it's ok to
post this and see what others have done.
THE GOOD NEWS:
I have successfully
Actually, it looks like I'm getting this problem on all my phones. When I was
testing my phones, most worked pretty well with an occasional complaint from
the Polycom.
I've moved them now to a different location and the ISP must have different
NAT translation going on that make it more difficult
A DID number was dedicated to receive fax, but i have the problem when
getting fax call,
which call will become a normal phone call and no fax was printed. When
fax is detected,
the fax extension is executed and dial the extension of the HT486 device
(firmware 1.0.5.22).
Somehow sending fax out
Is anyone actually using Enum or Dundi for internet voip routing?
Both approaches seem to be rather silent on this list, cvs, etc.
Is either (or both) approaches something that should be considered
for small businesses?
How about small itsp's?
Rich
qualify = yes is what is causing the messages. You can assign a value
rather than yes. like 1000 or something or you can remove the qualify
statement alltogether. The message is just a warning. Eliminating the
warning does not eliminate the lag problem.
- Original Message -
$3,000 u$d/mo in Senegal Africa.
- Original Message -
From: Adam Vocks [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, May 27, 2005 10:09 PM
Subject: RE: [Asterisk-Users] Survey: E1 prices
Not sure about
A DID number was dedicated to receive fax, but i have the problem when
getting fax call,
which call will become a normal phone call and no fax was printed. When
fax is detected,
the fax extension is executed and dial the extension of the HT486 device
(firmware 1.0.5.22).
Somehow sending
Rich Adamson a écrit :
Is anyone actually using Enum or Dundi for internet voip routing?
Both approaches seem to be rather silent on this list, cvs, etc.
Is either (or both) approaches something that should be considered
for small businesses?
How about small itsp's?
I'm using enum but I
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Steve Underwood
Sent: Friday, May 27, 2005 6:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] G729 vs. gsm
Well, it does to anyone without hearing
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Adam Goryachev
Sent: Friday, May 27, 2005 7:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] CRM integration (was RE: CallerID)
of client hardware.
Its obvious that Steve never looses, even when he's wrong, so arguing
about it to him won't get anywhere.
As for g729, I was pleasantly surprised by the quality.
I may be old fashioned, but the purpose of my phone system is to
communicate voice with other people, mostly in a business
Hi,
I have been testing out a new voip phone from Comdial called
EP300.
I have it working with Asterisk with very good results. This
phone is the best phone I have tested with a price under
$150. The only issue with this phone is programming the
buttons. I ran a ethereal trace on it with a comdial
city), with no outbound shaping.
I don't care about what anyone else says, I am impressed with g729 and
if it weren't for 1) the cost of transcoding and 2) the lack of g729 on
FreeBSD I'd be using it more.
The digium g729 codec works fine on Freebsd 5.X.
Chris
Well shame on me. There they are, plain as day.
I stumbled upon some posts in the FreeBSD mailing list complaining about
the lack of the g729 codec on FreeBSD, and assumed that was still the case.
Thanks for pointing that out,
Tim
snacktime wrote:
city), with no outbound shaping.
I
snip
Could you use javascript, or java from within the browser, which is
both portable, and likely to work on ANY browser
that way there is no installation as such, just visit the page, and
leave a browser window open (minimised) which is 'listening' for
connections ??
Sigh...
Browsers
Hi,
Via PSTN, if I want * to automatically pick up after the first ring how
would I do it? At the moment it picks up after the 4th ring.
Thanks in advance
Phil.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Tom Fanning
Sent: Saturday, May 28, 2005 11:20 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] CRM integration (was RE: CallerID)
Could you use
On Saturday 28 May 2005 20:21, Rusty Shackleford wrote:
D'oh!
I had misread the PP's statement and assumed he meant a bareback
browser window.
You are, of course, quite right. A Java app could handle this, but we
are still left with the issue of having to install SOMETHING, even if it
is a
Hi,
Can anyone tell me if UK CallerID support has been added to the CVS for the
x100p card ??? I'm new to * so please excuse me if this question has been
asked a million times already.
Thanks in advance
Phil.
___
Asterisk-Users mailing list
does anyone have information about the purpose and the structure of
this file - maybe a sample?
you can find it on the cisco website. Look for telephony locale.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
snip
Browsers don't listen. They inititiate a connection, process the
requested transaction with the web server, and close the
connection. The simply can't be used to listen for an
arbitrary connection.
Actually, I don't think that you are quite right here.
The guy mentioned
pressing the End call softkey while in call produces the message
That key is not active here. Do I have to configure something
special for that to work or is it just not yet implemented?
I did fix it (and more button template issues) this week, waiting for
julien to test and commit it to
On Saturday 28 May 2005 20:33, [EMAIL PROTECTED] wrote:
Hi,
Can anyone tell me if UK CallerID support has been added to the CVS for the
x100p card ???
http://www.voip-info.org/wiki-Asterisk+and+UK+Caller+ID
Please, please always try the wiki first :)
Cheers,
Gavin.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Tom Fanning
Sent: Saturday, May 28, 2005 12:39 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] CRM integration (was RE: CallerID)
No installation as
On Saturday 28 May 2005 12:34, Rusty Shackleford wrote:
This is meaningless drivel.
Hardly. Each conversion introduces the equivalent of gen loss. Two
such conversions are easily encountered, especially when dealing with a
third-party network, and will produce (in MY subjective opinion)
On Saturday 28 May 2005 00:53, Tim Pushor wrote:
Its obvious that Steve never looses, even when he's wrong, so arguing
about it to him won't get anywhere.
It's obvious to anyone who's been doing this for any amount of time that he's
not wrong here. But hey -- don't let reality get in the way
On Saturday 28 May 2005 12:42, Rusty Shackleford wrote:
Browsers don't listen. They inititiate a connection, process the
requested transaction with the web server, and close the connection. The
simply can't be used to listen for an arbitrary connection.
Don't be so closed-minded. Yes,
Hi Gavin,
Thanks for the reply. I did look at the wiki and read the page you pointed
at. According to the wiki mark didn't want to add caller id for the x100p
card. I found patches for the x100p card and just this minute re-compiled
and they work a treat. Was just wondering if the patches had
On Saturday 28 May 2005 21:41, [EMAIL PROTECTED] wrote:
Hi Gavin,
Thanks for the reply. I did look at the wiki and read the page you pointed
at. According to the wiki mark didn't want to add caller id for the x100p
card. I found patches for the x100p card and just this minute re-compiled
Jefferson,
Thanks a lot
Worked for me.
Let me know what are you doing !
Miguel
Hi Miguel,
Try to use a PUBLIC PROXY outside BRAZIL.
I don4t know exacly what4s happening ... but
routing direct from brazil doesn4t reach Areski4s
pressing the End call softkey while in call produces the message
That key is not active here. Do I have to configure something
special for that to work or is it just not yet implemented?
you can quick apply this patch
Index: sccp_device.c
I recently began using the curl cmd to do an external callerid
lookup on my own customer database. I've noticed certain lookups will cause
a crash and not show anything in the messages file or the console. The curl
command is connecting to an external webserver which has a oracle db
I am impressed, I have been trying this for sometime using the SIP image and
the only difference I can create is a 'single' and a 'double' ring on the
phone. I use the 'single' ring for phone calls and the 'double' ring for the
doorbell. I would love to be able to choose a ring tone based on
Hi Gavin,
Can you recommend a good card/modem for the UK? Looks like I'm going down
the same track as you. At first I was going for the TDM400 dev kit but
they are not certified or available yet in the UK and I wanted to play :-)
Cheers,
Phil.
On Saturday 28 May 2005 21:41, [EMAIL
On Saturday 28 May 2005 23:31, [EMAIL PROTECTED] wrote:
Hi Gavin,
Can you recommend a good card/modem for the UK? Looks like I'm going down
the same track as you. At first I was going for the TDM400 dev kit but
they are not certified or available yet in the UK and I wanted to play :-)
The
If extension 101 calls 102 and user 102 hits # and then 103,
the caller ID of 103s phone says 102. Ive been looking for a way
to have 103s Caller ID show the person that is being transferred not the
person transferring.
So if my receptionist answers the phone and transfers it to
one of
On Saturday 28 May 2005 23:54, Adam Vocks wrote:
So if my receptionist answers the phone and transfers it to one of my
techs, I want my techs phone to display the caller ID of the person who
called the receptionist.
Does anyone have a solution to this problem?
Hi :)
show application dial
mmm lets say u export a partition and is then mounted on /mnt/ast-voicemail
you can try to symlink both directories...
/var/spool/asterisk/voicemail/ - /mnt/ast-voicemail/vmail-spool/
/var/lib/asterisk/voicemail/ - /mnt/ast-voicemail/vmail-lib/
but im an asterisk
I do not see the 'o' option. I'm using 1.0.7???
Adam
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Gavin Hamill
Sent: Saturday, May 28, 2005 6:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
I was referred to this URL:
http://www.thevoipconnection.com/store/catalog/
product_16221_SOYO_G668_VoIP_Telephone.html
- Waldo
On May 27, 2005, at 7:32 PM, Isamar Maia wrote:
Do you have any link? Isn't it PA-1688 Chip?
Isamar
On Fri, 27 May 2005, Waldo Rubinstein wrote:
Has anyone
Decarpentrie Guy wrote:
Hi all,
In order to do loadbalancing between my two *, i wanted to stock all
things concerning voicemail on a NFS partition...
I see that the voicemail system put his files onto two differents
directories :
/var/spool/asterisk/voicemail/mycontext etc.
and
Yes. it's a PA1688. IMHO, it works well for Home users but don't even
think to use it for business applications.
The great thinkg is that it works with IAX2.
Isamar
On Sat, 28 May 2005, Waldo Rubinstein wrote:
I was referred to this URL:
http://www.thevoipconnection.com/store/catalog/
Check this notice:
http://hackaday.com/entry/1234000640041977/
--
Guillermo Salas M.
Telconet S.A. Manta
Calle 15 y Av. 24 Esq.
Phone : 593 5 262 8071
Mobile: 593 9 985 5138
SIP : [EMAIL PROTECTED]
e-mail: [EMAIL PROTECTED]
www : http://www.telconet.net
http://www.telcocarrier.net
Installed the new version and, yep, thats
what I wanted!
Thanks asterisk developers.
Adam
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam Vocks
Sent: Saturday, May 28, 2005 6:12
PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE:
I read about the PA1688 and, yes, it says to support IAX2. However,
reading the PDFs on the Soyo G688, I found no reference to IAX2 at
all. How certain are you that the Soyo G688 is based on the PA1688?
Also, why do you not recommend using it for business apps?
Thanks,
Waldo
On May 28,
Tim Pushor wrote:
Its obvious that Steve never looses, even when he's wrong, so arguing
about it to him won't get anywhere.
I have been known to loose arguments. Not too many, but some. :-)
As for g729, I was pleasantly surprised by the quality.
I fully agree.
I may be old fashioned,
Is it possible to give a caller three goes at an extension number then
hangup?
exten = s,1,Zapateller(answer|nocallerid)
exten = s,2,PrivacyManager
exten = s,3,Ringing(1)
exten = s,4,NoOp(${CALLERID})
exten = s,5,SetMusicOnHold(random)
exten = s,6,Background(silence/1)
exten =
Waldo,
The external material quality is business-level. It looks like a toy.
Depending on the business you will provide this as a solution, it will not be
acceptable. But... it can be relative.
For business, I would recommend http://www.act-tel.com.tw
Forget the support from Soyo. Just confirm
Yeah sure (this is what I use):
[closed]
exten = s,1,SetVar(CNT=0) ;set the counter variable
exten = s,2,Goto(s-toolong,1)
some code here.
;here we have the timeout extension that makes sure we incerement and
test the counter
exten = t,1,Goto(s,2)
;this is the extension
Can you shoe your dialplan? asterisk should pick up as soon as it has
CallerID, which on american callerid is at the 2nd ring. If you
disable callerid in zapata.conf than it should pickup as soon as it
rings.
On 5/28/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi,
Via PSTN, if I want * to
I'm working on a multi user database for *, and I'm debating with
myself on something and thought I would throw it out for comment.
I have a design where context names are stored in the database in the
following format:
__username__contextname__
Now, in the web interface it's easy to parse the
It isn't made Soyo. Its an ODM job from one of the makers in Shenzhen. I
forget which, off hand. Hunt for that, and it should tell you which of
the standard PA168 firmwares is right for it.
Regards,
Steve
Isamar Maia wrote:
Yes. it's a PA1688. IMHO, it works well for Home users but don't
Greetings to all!
Please, I want to buy a SIP or iax2 supporting HARDWARE phone which can
directly be connected to my Asterisk PBX from Dubai. and your
recommendations are highly appreciated.
Thanks
Kumara
___
Asterisk-Users mailing list
Hello everyone,
Even though a lot of it was a bit last minute, several of us from the
commnunity made it to Baltimore to help Digium with their booth at
ISPCon. It was a great time.
Gregory Boehnlein, Brian Capouch, Christian Savinovich, Kristian
Kielhofner (me), and John Todd (not
Anyone have the time and webspace to post a quick recording of a
sample conversation in both codecs? If you want to get even more
tricky, perhaps samples of music on hold in both as well? Or noisy
environments?
Obviously not very scientific and prone to a wide margin of error
(background noise,
A price range would be helpful if you want useful suggestions. Or
perhaps major features you're looking for?
A Budgetone is very good if you only want to spend $60US. But at the
same time, the Cisco 7970 is an awesome phone at $570US.
Do you need a business quality speakerphone? Single or
Marie wrote:
Anyone have the time and webspace to post a quick recording of a
sample conversation in both codecs? If you want to get even more
tricky, perhaps samples of music on hold in both as well? Or noisy
environments?
This kind of quickie test is worthless. In doing serious codec
I am in the process of making and SS7-to-VOIP implementation (basically
Astersisk/SS7 implementation). I have researched the subject, but the
results are not encouraging. There is an open SS7 initiative that is
basically stalled. Is there anyone who can post their opinions or
experience or
We use g729 for everything, and it sounds great. Have no sound clips
handy, but on decient equipment it works very very well.
Preston Garrison
direct: 877-748-4142
fax: 310-774-3901
cell: 623-748-4140
-Original Message-
From: Marie [EMAIL PROTECTED]
To: Asterisk Users Mailing List -
On Sun, 2005-05-29 at 00:21 -0700, VOIP Consultant wrote:
I am in the process of making and SS7-to-VOIP implementation (basically
Astersisk/SS7 implementation). I have researched the subject, but the
results are not encouraging. There is an open SS7 initiative that is
basically stalled.
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