Hi,
I am experimenting with receiving faxes in asterisk:
exten = in_fax,1,Macro(faxreceive)
exten = in_fax,2,system(tiff2ps -2eaz -w 8.5 -h 11 ${FAXFILE} | ps2pdf
- ${FAXFILE}.pdf)
exten = in_fax,3,system(cp ${FAXFILE}.pdf
/var/www/faxes/${CALLERID(number)}.pdf)
exten =
They're for inbound only though some of them provide termination services
-- Original message -- From: "VIC IP Communications" [EMAIL PROTECTED]
Hi,Companies like DIDx and Sixtel, when they state DIDs at $XX.XX per month and $XX.XX per minute/monthly,do these
I had made a patch a while back so it retrieved the emailbody and
emailsubject from the users table in mysql.
Let me see if I can dig it up
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: April 18, 2006 9:05 AM
To:
Marco Mouta wrote:
I forgot to write: When i hangup the call, it hangs correctly!
On 4/18/06, *Marco Mouta* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Hi all,
I've a Planet VIP-150T VoIP Hardphone connected to Asterisk. When
I'm in a call and i press Hold button, the
Is this message
"normal"???
Apr 18 16:26:29 WARNING[1229]: channel.c:1323
ast_hangup: Hard hangup called by thread 51792816 on Local/[EMAIL PROTECTED],1ZOMBIE,
while fd is blocked by thread 51792816 in procedure ast_waitfor_nandfds!
Expect a failure
RegardsDov
Jim Houser wrote:
I have our Avaya connected to Asterisk using NI D channel protocol
over a standard ESF/B8ZS span. It works great.
span=1,0,0,esf,b8zs
Shouldn't you be getting your timing from the Avaya?
Doug
___
--Bandwidth and Colocation
Dov Bigio wrote:
Is this message normal???
Apr 18 16:26:29 WARNING[1229]: channel.c:1323 ast_hangup: Hard hangup called
by thread 51792816 on Local/[EMAIL PROTECTED],1
mailto:Local/[EMAIL PROTECTED],1ZOMBIE ZOMBIE, while fd is blocked by
thread 51792816 in procedure ast_waitfor_nandfds!
Ok.
First of all , be
sure Redfone ethernet link and the Asterisk ethernet link are both on the same
switch segment.
Then try an pri
intense debug on asterisk console. I believe (not sure), this link is
not at IP level but ethernet level 2
It can help to
determine if packets get stucked
Well this is disappointing. Time to find somebody else...
--
Wes
-Original Message-
From: NuFone Operations [mailto:[EMAIL PROTECTED]
Sent: Tuesday, April 18, 2006 3:44 PM
To: [EMAIL PROTECTED]
Subject: NuFone Update: DIDs
Effective 3pm EST Today, April 18th, 2006 Telesthetic, the
Does Sixtel provide E911 service? have tried it out.
-- Original message -- From: "Kaleb L. Kunzler" [EMAIL PROTECTED] Being a sixTel customer I can tell you how sixTel bills. They charge $X.XX per month for a DID, they also charge per minute inbound (a certain rate)
Im using SixTel as a test (Opened
account w/ $10) and am happy with them so far In their basic
service package, they dont charge a monthly fee, and its outbound
only, and you get charged for every minute. I paid for a DID, which is
$1.50 or so per month, and it lets me receive inbound
Daaah, you are correct.
A typo on my part, not a cut paste from my actual build.
Make that span=1,1,0,esf,b8zs
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle
Sent: Tuesday, April 18, 2006 2:37 PM
To: Asterisk Users Mailing List -
Thanks!
technically you have your asterisk connected to the avaya (not the other way),
since your config below indicates that asterisk is doing CPE side signalling,
which menas the avaya must be doing network side signalling.
nevertheless, the end result is the same!
Anybody have any recommendations? IAX service preferred.
-Seth
On Tue, 2006-04-18 at 15:48 -0400, Wes Baehr wrote:
Well this is disappointing. Time to find somebody else...
--
Wes
-Original Message-
From: NuFone Operations [mailto:[EMAIL PROTECTED]
Sent: Tuesday, April 18,
So it could be BRI, PRI or maybe even Analog there??
I guess what I'm asking is it predominantly ISDN there or not
Thanks for the input about the card and chan-capi
:-)
--
Chris
- Original Message -
From: Armin Schindler [EMAIL PROTECTED]
To: Chris Earle (CBL) [EMAIL PROTECTED];
I am open to suggestions as well.
On 4/18/06, Seth Remington [EMAIL PROTECTED] wrote:
Anybody have any recommendations? IAX service preferred.
-Seth
On Tue, 2006-04-18 at 15:48 -0400, Wes Baehr wrote:
Well this is disappointing. Time to find somebody else...
--
Wes
Seth Remington wrote:
Anybody have any recommendations? IAX service preferred.
Teliax has been good to me.
Darrick
--
Darrick Hartman
DJH Solutions, LLC
http://www.djhsolutions.com
___
--Bandwidth and Colocation provided by Easynews.com --
I have made available the base Asterisk sounds (i.e. as included with
Asterisk) in male UK English in gsm format. They are released under the
Creative Commons Attribution 2.5 License.
They are available as a single compressed tar file via: -
http://www.tel.net/
The Asterisk sounds (as in the
Darrick Hartman wrote:
Seth Remington wrote:
Anybody have any recommendations? IAX service preferred.
Folks, please.
This thread has the potential to become a torrent.
It belongs on -biz as it has nothing to do with Asterisk itself.
B.
___
Guys, this is a weird question but has anybody disabled the blind button
that appears on polycoms or know if you can disable the use of blind
transfers on polycoms to make any transfer attended?
Thx!
___
--Bandwidth and Colocation provided by
I could be wrong but off the top of my head I think that it is in the
features section of the config file.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton
Krall
Sent: Tuesday, April 18, 2006 4:47 PM
To: 'Asterisk Users Mailing List -
Hello everybody, I'm installing an Asterisk 1.2.7.1 with GNUDialer
0.98-puff18. It also has zaptel from CVS. My FXO is an X100P Clone.
The agents from GNUDialer log ok, and everything is fine until the
GNUDialer makes a call, as soon as it engages (the phone starts to
ring) asterisk crashes with
In article [EMAIL PROTECTED],
Steve Kennedy [EMAIL PROTECTED] wrote:
I have made available the base Asterisk sounds (i.e. as included with
Asterisk) in male UK English in gsm format. They are released under the
Creative Commons Attribution 2.5 License.
They are available as a single
I've just set up my dial plan to use hints for each of my extensions,
and I've set up a buddy watch list on my Polycom IP-501's, however, I
keep getting the following messages every 5 or so minutes:
-- Incoming call: Got SIP response 500 Internal Server Error
back from 192.168.1.106
--
I have an application where I need to send outbound prerecorded
messages. The Asterisk call file process works fine if I am sending
the call via SIP or IAX, but not via ZAP over a PRI channel. The
destination device (my cell phone) never rings. The only unusual thing
I see is on the fifth
Hello All,
Phone: Cisco 7960G
Asterisk 1.2.7.1
libpri 1.2.2
zaptel 1.2.5
OS: Fedora Core 4
TDM2400P w/8FXO
When dialing an outbound number, sometimes all the digits are not
dialed properly on the outside line. In the dial plan I added a
SayDigits to the outbound rule and it properly
On 4/18/06, Damon Estep [EMAIL PROTECTED] wrote:
Yep,
I agree,
Just watch out for regulatory issues if you are in the USA, handing a
CUSTOMER a TDM interface vs. a SIP/VoIP interface falls under a much
different regulatory and jurisdictional set of rules...
regulatory issues only apply if
On Tue, Apr 18, 2006 at 08:51:02PM +, Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Steve Kennedy [EMAIL PROTECTED] wrote:
I have made available the base Asterisk sounds (i.e. as included with
Asterisk) in male UK English in gsm format. They are released under the
Creative
SellVoIP are great. Actually the rates are fine, but I like
the quality as well.
I can't say I ever needed phone support.
- Original Message -
From:
[EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Tuesday, April 18, 2006 21:54
J1 is just a T1, the J2 is also very common and likely what you would
see. J2 = 48B+D if I remember correctly.
On 4/18/06, Chris Earle (CBL) [EMAIL PROTECTED] wrote:
Hi all,
general query here --- I'm about to set up an asterisk box for use in Japan
but can't figureout if it's all ISDN there
Armin,
Why not use OCN etc and connect directly to their sip server?
If you use OCN .office you can get multiple lines and multiple
numbers...The quality is great.
Jason
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users
that is 47B+1D
On 4/18/06, Andrew Latham [EMAIL PROTECTED] wrote:
J1 is just a T1, the J2 is also very common and likely what you would
see. J2 = 48B+D if I remember correctly.
On 4/18/06, Chris Earle (CBL) [EMAIL PROTECTED] wrote:
Hi all,
general query here --- I'm about to set up an
I think this should have been sent to Chris..
Chris,
Why not use OCN etc and connect directly to their sip server?
If you use OCN .office you can get multiple lines and multiple
numbers...The quality is great.
Jason
___
--Bandwidth and Colocation
avez vous trouvé une solution ?
jean-louis curty
2006/1/6, maingault [EMAIL PROTECTED]:
Hi,
I'm a new user of Aterisk, and I have to configure a VoIP Gateway.
I have an Alcatel PBX with an E1 card, connected, for the moment, to a local carrier.
I would like work with a french VoIP provider,
On Tue, Apr 18, 2006 at 09:30:36PM +0100, Steve Kennedy wrote:
I have made available the base Asterisk sounds (i.e. as included with
Asterisk) in male UK English in gsm format. They are released under the
Creative Commons Attribution 2.5 License.
They are available as a single compressed tar
I am having a strange problem with [EMAIL PROTECTED] 2.7 (Asterisk
1.2.5) with a
TE210P card and Unicall. I have compiled everything and Unicall seems
to be working well. The only problem we are having is that the outgoing
voice is a bit distorted. When someone from the inside calls
On Tue, Apr 18, 2006 at 06:17:22PM -0400, Tzafrir Cohen wrote:
Any chance of a better license?
http://people.debian.org/~evan/ccsummary.html describes why Debian
considrs version 2.0 of the same license problematic. The reasons
mentioned are pragmatic reasons (of the sort of: the license may
I see, I misunderstood what you meant by customer. It is not you at
one end and the customer at the other, you are doing the work for a
customer that owns both endpoints...
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of C F
Sent:
Greetings,
Somewhere on my asterisk system, a calls come in in a certain
context, for example, from-sip or from-pstn.
Then the calls gets routed through the dialplan, and a macro gets
called, and another one and then the call needs to be redirected
to another number in the same initial context.
I think sipsak can do that for you
TWV wrote:
I should probably add that I tried by creating my own section in
sip_notify.conf but is it possible that you can't include a body there (only
headers)?
When I just add a line (with no =) it is sent in the notify message, but
with no whitespace
I thought this was Local/[EMAIL PROTECTED]
If the number is unique in your configuration, and the context is
already included somehow in your default-context, you can just do a
Local/number, give it a try
Edwin Groothuis wrote:
Greetings,
Somewhere on my asterisk system, a calls come in
Hi,
I'm having problems with the voicemail, the app keeps exiting in 3-5 seconds.
Any considerations will be appreciated.
Thanks,
D.K.
Debug messages:
Apr 18 20:53:41 DEBUG[26150] pbx.c: Launching 'Goto'
Apr 18 20:53:41 DEBUG[26150] pbx.c: Launching 'VoiceMail'
Apr 18 20:53:41 DEBUG[26129]
Hi Carlos. Please provide the version of unicall. Also, are you using
MFCR2 or PRI? and finally, provide us with the codecs that you have
tried. PSTN in Mexico should be ALAW, so try to avoid transcoding and
use ALAW in the phones too, we had bad issues with ILBC when the calls
were coming from
Do you know if you can tweak gains if using unicall? I tried it once and if
you move the gains on zaptel using a te110p with unicall on E1, when gains
are +2 or -1, calls do not complete, forget even about faxing :)
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL
That's great news! Seen sangoma is beating digium :)
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of yusuf
|Sent: Tuesday, April 18, 2006 10:57 AM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users]
I cant seem to find such an option on the xml config files so far :(
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Jonathan k. Creasy
|Sent: Tuesday, April 18, 2006 3:56 PM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject:
So I have * box shorewall/linux NAT firewall internet - WRT54G
with openwrt - IP500
I have 5060, 4569, and 1 through 2 forwarded to * box from internet. I
have tried everything I can think of on the wrt to get it to work but it
appears, looking at tcpdump that my
Hi Anton,
If you need to tweak gains for FAXing, something is badly wrong. The
modems operate over a wide range of signal levels.
Steve
Anton Krall wrote:
Do you know if you can tweak gains if using unicall? I tried it once and if
you move the gains on zaptel using a te110p with unicall on
How can i get this uniq call session id?
--
~Shaun
[EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
Each call has a unique callid - I used that for a dialplan a short while
ago, to do a very similar job to what you are doing...
Paul Hales
Technical Manager
AsteriskIT
-
First of all, try sending it to the
asterisk-biz list.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Rich
Sent: Monday, April 17, 2006 10:53
AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Don't
see my post
Hi Folks,
I have posted a
*${UNIQUEID}*: Current call unique identifier
- From the Asterisk Variables webpage
(http://www.voip-info.org/wiki/view/Asterisk+variables)
regards,
Paul Hales
--
Paul Hales
Technical Manager
Asterisk IT
mob: 0434 225 491
Shaun wrote:
How can i get this uniq call session id?
List,
The past few days the asterisk service on my server has
crashed several times. I have had it running for months and have made no
changes to it.
When it crashes, I am unable to make calls or gain access to
the CLI. The service has been stopped. If I try to start it again
On Tue, 18 Apr 2006 19:15:33 -0500, Moises Silva wrote
Hi Carlos. Please provide the version of unicall. Also, are you using
MFCR2 or PRI? and finally, provide us with the codecs that you have
tried. PSTN in Mexico should be ALAW, so try to avoid transcoding and
use ALAW in the phones too, we
On Tue, Apr 18, 2006 at 06:57:32PM -0700, [EMAIL PROTECTED] wrote:
If the number is unique in your configuration, and the context is
already included somehow in your default-context, you can just do a
Local/number, give it a try
The issue is, I don't want it in the default context (which is
Carlos,Unicall is for signalling only, i.e., it only takes part in setting up and tearing down a connection, therefore, it has nothing to do with speech quality.I'd think what you're experiencing is phone related and/or transcoding.
Regards,
___
Ronald Hartmann wrote:
Good Day List,
I am looking to see if anyone is willing to share their working
configs with me.
I would be happy to add to wiki and document steps to get it to
work with asterisk.
I am looking for both Welgate configs as well as sip.conf and
To do that you need to get the remote ip address and port of the sip peer!
I found the function:
${SIPPEER(exten:ip)
But how can I get the port???
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sippeer
You have all sorts of info BUT... I see no port!!! :-(
I can't believe that
Dear Hirosh,
I already knew about that :-), and tried it with success.
However, that was not my question!
I asked how you can make the phone browse to a certain http:// URL,
initiated from the server side.
So essentially remote control the phone to open an XML file from some server
in its
On 4/17/06, Joseph Rothstein [EMAIL PROTECTED] wrote:
Is anyone using a PRI to Ethernet bridge, or any other kind of E1 GW thatwould allow failover to an alternate Asterisk box without manually switchingthe cable? This one is a litteexpensive(
When dialing an outbound number, sometimes all the digits are not dialed
properly on the outside line. In the dial plan I added a SayDigits to the
outbound rule and it properly reads back the entire number that was entered
on the phone before dialing.
Is asterisk dialing too quickly, is
Try adding the following to sip.conf
--
[general]
progressinband=no
-Jon
Brent Torrenga wrote:
Anyone experience the double ringing when calling out over TelIAX? I am
using a Cisco 79[46]0, and do not use the r option in the Dial() command.
I always thought that the r is what
On Wed, 19 Apr 2006 09:22:50 +0700, CC Jay wrote
Carlos,
Unicall is for signalling only, i.e., it only takes part in setting up and
tearing down a connection, therefore, it has nothing to do with speech
quality.
I'd think what you're experiencing is phone related and/or
transcoding.
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