Hi!
Philip Prindeville wrote:
Yes, you can tickle an SPA94x or 962 and have it fetch a config from a
TFTP server... But is there no way to simply push a couple of lines
of XML config to it directly via an HTTP POST (sans TFTP server)?
If you have HTTP access to this phone, why not? Just
Hi Steven,
I do not live in China but I had the same problem.
Try these 2 params inside zapata.conf:
busydetect = yes
hanguponpolarityswitch = yes
It worked for me.
Giorgio Incantalupo
Steven O'Reilly wrote:
Afternoon,
I was hoping someone could point me in the right direction. I
Moises,
I try put the line exactly like you send me, saw the time wait getting
longer with the parameter you describe to increment. But the error is the
same as you can see in logs.
Has other way to solve this problem, may I question to my telephony service
de time it's need to send back the ACK
On Thu, Dec 13, 2007 at 11:40:32AM +0100, gincantalupo wrote:
Hi Steven,
I do not live in China
Is live in China good enough? How big is the variance between
different regions, providers(?) etc?
but I had the same problem.
Try these 2 params inside zapata.conf:
busydetect = yes
Hi,
Armin wrote:
How does your dialplan look like? If you have e.g.
exten = _.,1,
in the context for capi incoming calls, then asterisk (chan-capi)
accept these calls even if not all numbers are dialed (transmitted) yet.
we found the reason for our problem, but not yet the
On Thu, 13 Dec 2007, Stefan Guenther wrote:
Hi,
Armin wrote:
How does your dialplan look like? If you have e.g.
exten = _.,1,
in the context for capi incoming calls, then asterisk (chan-capi)
accept these calls even if not all numbers are dialed (transmitted) yet.
we found the
I know one of the guys in the NY Asterisk meetup demonstrated that application
on a polycom display.
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357
+61-2-9016-5642 (Sydney in-dial).
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On
I don't see any evidence that queue metrics can push data to the
phone. I'm really looking for a home-grown solution that pushes
XML/HTML to a phone during a call, like the 7960's.
On 12/13/07, Dovid B [EMAIL PROTECTED] wrote:
Queue Metrics
- Original Message -
From: Peter Pauly
Roger C. Beraldi Martins wrote:
Moises,
I try put the line exactly like you send me, saw the time wait getting
longer with the parameter you describe to increment. But the error is
the same as you can see in logs.
Has other way to solve this problem, may I question to my telephony
I have 2 asterisk servers - serverA and serverC - connected via IAX2.
On serverA, I have a telemarketer hold extension which, if I transfer a
caller into it, loops around playing music please wait messages, until
they give up hang up the phone.
Also on serverA, I have a custom devstate, which
- Original Message -
From: Ade Vickers [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Thursday, December 13, 2007 7:49 AM
Subject: [asterisk-users] How do I do this?
I have 2 asterisk servers - serverA and serverC
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Steve Totaro
Sent: 13 December 2007 14:35
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How do I do this?
- Original Message -
From: Ade
As Steve said, put loglevel=255 in unicall.conf to have more details,
please post them here and I will review them.
On Dec 13, 2007 5:02 AM, Roger C. Beraldi Martins
[EMAIL PROTECTED] wrote:
Moises,
I try put the line exactly like you send me, saw the time wait getting
longer with the
Hi,
Armin wrote:
Again, same question. How does your dialplan look like?
If you have a rule _. (which means 4 digits OR MORE), then there
is a match even if other digits follow. Make sure your dialplan
will accept the call only if all digits are dialed.
here is the relevant part:
On Thu, Dec 13, 2007 at 10:26:59AM -0500, Steve Totaro wrote:
Stefan Guenther wrote:
exten = 940331,3,DIAL(CAPI/g2/${EXTEN:4},20,tr)
How do your zap conf files look?
chan_capi is used in this thread.
--
Tzafrir Cohen
icq#16849755 jabber:[EMAIL
Are there any service providers offering Cape Town DID's?
--
- Eric Smith
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Please, put your zaptel.conf, ok ?
Using zttool can you see what channel the dial is attempting to use ? Is
that a valid channel ?
Luis A P Barbosa.
2007/12/13, Roger C. Beraldi Martins [EMAIL PROTECTED]:
Moises,
I try put the line exactly like you send me, saw the time wait getting
longer
Stefan Guenther wrote:
Hi,
Armin wrote:
Again, same question. How does your dialplan look like?
If you have a rule _. (which means 4 digits OR MORE), then there
is a match even if other digits follow. Make sure your dialplan
will accept the call only if all digits are dialed.
Steve Totaro wrote:
snippage
I suppose you could create a new context on server C, include
it in your internal context, and create an h exten on that
box to handle it locally. I am unsure why what you have does
not work but I assume the unable to transfer is a hint.
Except that, once
Ade Vickers wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Steve Totaro
Sent: 13 December 2007 14:35
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How do I do this?
- Original
Beraldi,
Do you have unused links
2007/12/11, Roger C. Beraldi Martins [EMAIL PROTECTED]:
Dears,
After having set up the board Digium TE420 to receive 3 E1s, I can receive
calls without difficulties. As you can see in the log below:
-- Executing [EMAIL PROTECTED]:1] NoOp(UniCall/14-1,
I have been unable to get callerid name passed from Cisco Callmanager
over a SIP trunk to Asterisk. Only the number is displayed. Has anyone
been successful getting callerid name?
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Tzafrir Cohen wrote:
On Thu, Dec 13, 2007 at 10:26:59AM -0500, Steve Totaro wrote:
Stefan Guenther wrote:
exten = 940331,3,DIAL(CAPI/g2/${EXTEN:4},20,tr)
How do your zap conf files look?
chan_capi is used in this thread.
I have never used
On Thu, 13 Dec 2007, Stefan Guenther wrote:
Hi,
Armin wrote:
Again, same question. How does your dialplan look like?
If you have a rule _. (which means 4 digits OR MORE), then there
is a match even if other digits follow. Make sure your dialplan
will accept the call only if all digits
Chris Bennett wrote:
Hi All,
I am seeking input from anyone who may have seen a similar
configuration and dealt with similar issues to what I'm experiencing.
Configuration:
- 2 sites (site A and B)
- Asterisk 1.2.23 on each site (Trixbox)
- Internet 512/512 symmetric at each site,
Though this is not exactly an asterisk question, I have thoughts of asterisk
integration. Does anyone know how I might get a several digit SMS code and
allow people to SMS it? The second step is, is there a way to charge their
bill x dollars when they do that (ie for a donation to a non-profit
The fax machines will be talking directly to the spa2102 - the problem is
that Asterisk only supports being a T.38 pass-thru and not an end-point. And
I need the ability to fax over WAN links like ADSL... so I'll have a fax
machine plugged into a Linksys SPA2102 which will connect to Asterisk
On Thu, 13 Dec 2007, Stefan Guenther wrote:
Hi,
Armin wrote:
That looks corrct so far, numbers with just 94033 should not be
accepted, because of no match.
I then added a context to match those, too.
Otherwise I would loose these calls.
What type of ISDN line do you have?
4 BRI
On Thu, 13 Dec 2007, Steve Totaro wrote:
Tzafrir Cohen wrote:
On Thu, Dec 13, 2007 at 10:26:59AM -0500, Steve Totaro wrote:
Stefan Guenther wrote:
exten = 940331,3,DIAL(CAPI/g2/${EXTEN:4},20,tr)
How do your zap conf files look?
chan_capi is used in this thread.
I have
Hi Matt,
I've consulted in this space and can answer all of your questions. You
don't specify if you are in the USA so I'll assume you are (lol all the
people overseas are laughing at this point so laugh along :-) ).
You're not going to like the answers I have for you.
SMS Common
Hi,
Armin wrote:
That looks corrct so far, numbers with just 94033 should not be
accepted, because of no match.
I then added a context to match those, too.
Otherwise I would loose these calls.
What type of ISDN line do you have?
4 BRI (Anlagenanschluß) connected to an EICON 4 BRI8M
And
SIP2.0
[EMAIL PROTECTED] wrote: What version of SIP do Asterisk 1.4.x uses.
Regards,
Sanjay.
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Dean,
Thanks... this pricing 'falling on face' will be determined by what our
client says. Somehow I think it will be out of their range :) But we'll
see.
On Dec 13, 2007 12:38 PM, Dean Collins [EMAIL PROTECTED] wrote:
Hi Matt,
I've consulted in this space and can answer all of your
Thanks very much for the help Giorgio, I will give this a try today =)
On Thu, 2007-12-13 at 11:40 +0100, gincantalupo wrote:
Hi Steven,
I do not live in China but I had the same problem.
Try these 2 params inside zapata.conf:
busydetect = yes
hanguponpolarityswitch = yes
It worked for
Ade Vickers wrote:
I have 2 asterisk servers - serverA and serverC - connected via IAX2.
On serverA, I have a telemarketer hold extension which, if I transfer a
caller into it, loops around playing music please wait messages, until
they give up hang up the phone.
Also on serverA, I have
Hi,
Armin wrote:
Okay, but then the setting in capi.conf
isdnmode=MSN
is wrong. For 'Anlagenanschluss' you need
isdnmode=did
Okay , I changed that.
These options are no general options, these are for the port sections
only:
immediate=yes
The example capi.conf says about this
This is not the only way to go, there is another option. I dealt with a
provider that would/could provide a shared short code and route messages by
keyword. Pricing was something like $.06/per SMS with no obligation. I will
try to dig out the vendor that was supplying this. I have no idea
On Thu, 13 Dec 2007, Stefan Guenther wrote:
Hi,
Armin wrote:
Okay, but then the setting in capi.conf
isdnmode=MSN
is wrong. For 'Anlagenanschluss' you need
isdnmode=did
Okay , I changed that.
These options are no general options, these are for the port sections
only:
You cant do premium with a shared short code.
Technically under CTIA regulations you cant even do shared keywords with
multiple companies.
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357
+61-2-9016-5642 (Sydney in-dial).
Hi,
I've had a functioning Asterisk system (1.2.18), which I haven't
reconfigured in any way, that is just now refusing to forward calls. I
only have Polycom phones. When I use the phone's forward feature
(forwarding the phone with extension 204 to extension 206, which used to
work as
Additional info:
hci0: Type: USB
BD Address: 00:18:E7:22:XX:XX ACL MTU: 1017:8 SCO MTU: 64:8
UP RUNNING PSCAN ISCAN
RX bytes:21770594 acl:2032 sco:423505 events:13465 errors:0
TX bytes:16184599 acl:1078 sco:315044 commands:6278 errors:0
Features: 0xff 0xff
I will be out of the office from Dec. 10-13.
Thanks,
Dan Baker
Ursuline Academy
314-984-2828
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On Thu, 13 Dec 2007, Daniel M. Baker wrote:
I will be out of the office from Dec. 10-13.
Thanks,
Dan Baker
Ursuline Academy
314-984-2828
Thanks for the heads-up. We should be finished removing all of your
valuables by then :)
Thanks in advance,
The Asterisk.org development team has released Libpri versions 1.2.7 and 1.4.3.
These releases fix one small compilation error that occurred with the newest
release of glibc.
Thank you for your support!
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The Asterisk.org development team has released Zaptel versions 1.2.22.1 and
1.4.7.1. These releases contain one small change and are otherwise the same as
1.2.22 and 1.4.7. The change is to support the new TE122 card from Digium.
Thank you for your support!
-- Mensaje reenviado --
From: Eric C. [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Date: Sun, 9 Dec 2007 19:55:51 -0500
Subject: [asterisk-users] One server, multiple companies
Hello all,
Just starting to setup asterisk v 1.4.11 and need to run three distinct
Steve Edwards wrote:
On Thu, 13 Dec 2007, Daniel M. Baker wrote:
I will be out of the office from Dec. 10-13.
Thanks for the heads-up. We should be finished removing all of your
valuables by then :)
What do you mean 'by then'? I had everything, by the end of day on the
Hi there,
We've got a problem connecting Asterisk with a TE205P to a TeleWest
E1 ISDN line in the UK.
We get a lot of HDLC Bad FCS (8) on Primary D-channel errors, and
every so often the Primary D-channel goes down and all the calls got
dropped.
We've fully tested the card and
busydetect = yes
hanguponpolarityswitch = yes
Which of the two?
busydetect will work almost always. But it is suboptimal: it may sotimes
accidentally detect running calls. And it takes a few seconds to detect
a hangup.
Do you mean we need to adjust the value of busycount (larger than
Hello
I was wondering why there doesn't seem to a Windows version of Zaptel,
making the Digium and its clones unavailable for a Windows PBX.
Is the Zaptel/Zapata combo too *nix-centric?
Thanks.
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On Wed, 12 Dec 2007 21:42:49 -0600, Michael Graves
[EMAIL PROTECTED] wrote:
You can keep the POTS line but remote call forward to your ITSP.
Yup, but
1) the telco that handles the POTS line charges us for the connection
between our POTS number and the ITSP, with the caller obviously paying
for
Hi Mick -
I've had a functioning Asterisk system (1.2.18), which I haven't
reconfigured in any way, that is just now refusing to forward calls. I
only have Polycom phones. When I use the phone's forward feature
(forwarding the phone with extension 204 to extension 206, which used to
work
Hi Noah,
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Noah Miller
Sent: Thursday, December 13, 2007 21:02
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 1.2.18 and Polycom
phones
On Fri, 14 Dec 2007 03:09:40 +0100, Vincent wrote:
On Wed, 12 Dec 2007 21:42:49 -0600, Michael Graves
[EMAIL PROTECTED] wrote:
You can keep the POTS line but remote call forward to your ITSP.
Yup, but
1) the telco that handles the POTS line charges us for the connection
between our POTS number
Thanks Steven and Giorgio that will work. There are a couple of other
solutions depending on which telco you are connecting to and which exchange
they are using. E1 offers more issues than PSTN.
We are one of Digium's distributors in Asia with full technical support in
China either out of our
Noah,
Turns out I found the problem, BUT I don't understand it exactly. My phones
are on a LAN, and the PBX is on a different IP (Hosted PBX basically).
I had to open out port 5060 on my router (where the phones are). The thing
is, conversations flowed perfectly (with multiple phones at a
Hi Again Mick -
OK stupid question time: Can you successfully make a call from ext 204
to 206? Are those IP's on the phones real? Has any of the IP routing
changed? What does your sip.conf look like?
- Noah
On Dec 13, 2007 9:21 PM, Mike [EMAIL PROTECTED] wrote:
Hi Noah,
-Original
At 19:55 12/13/2007, Vincent wrote:
Hello
I was wondering why there doesn't seem to a Windows version of Zaptel,
making the Digium and its clones unavailable for a Windows PBX.
Is the Zaptel/Zapata combo too *nix-centric?
Thanks.
Windows is a half-baked, dying OS that in essence is
a
Doug wrote:
At 19:55 12/13/2007, Vincent wrote:
Hello
I was wondering why there doesn't seem to a Windows version of Zaptel,
making the Digium and its clones unavailable for a Windows PBX.
Is the Zaptel/Zapata combo too *nix-centric?
Thanks.
Windows is a half-baked, dying OS
Doug wrote:
At 19:55 12/13/2007, Vincent wrote:
Hello
I was wondering why there doesn't seem to a Windows version of Zaptel,
making the Digium and its clones unavailable for a Windows PBX.
Is the Zaptel/Zapata combo too *nix-centric?
Thanks.
Windows is a half-baked, dying OS
On Thursday 13 December 2007 19:55:39 Vincent wrote:
I was wondering why there doesn't seem to a Windows version of Zaptel,
making the Digium and its clones unavailable for a Windows PBX.
Because nobody has done it yet. The real answer is probably more along the
lines of that there's no
On Thu, 13 Dec 2007 22:21:50 -0600, Tilghman Lesher
[EMAIL PROTECTED] wrote:
It is likely to be a very strenuous job to port the framework and all of the
drivers.
Too bad, because there doesn't seem to be any PCI card for FXO/FXS
available for Windows.
Vincent wrote:
On Thu, 13 Dec 2007 22:21:50 -0600, Tilghman Lesher
[EMAIL PROTECTED] wrote:
It is likely to be a very strenuous job to port the framework and all of the
drivers.
Too bad, because there doesn't seem to be any PCI card for FXO/FXS
available for Windows.
Erm,
On Fri, 14 Dec 2007 14:50:28 +1000, [EMAIL PROTECTED] wrote:
Erm, there just might be, take a look at this...:
Ah yeah, forgot about $angoma ;-) I'll restate this as: No card for
home/SOHO use, ie. in the $50-100 range for the single FXO port model.
Tilghman Lesher wrote:
On Thursday 13 December 2007 19:55:39 Vincent wrote:
I was wondering why there doesn't seem to a Windows version of Zaptel,
making the Digium and its clones unavailable for a Windows PBX.
Because nobody has done it yet. The real answer is probably more along
Umm - you could just buy a SPA-3000/3102/3666/etc.
PaulH
On Fri, 2007-12-14 at 05:36 +0100, Vincent wrote:
On Thu, 13 Dec 2007 22:21:50 -0600, Tilghman Lesher
[EMAIL PROTECTED] wrote:
It is likely to be a very strenuous job to port the framework and all of the
drivers.
Too bad, because
I upgraded to Linux kernel 2.6.23-gentoo-r3 and bluez-libs-3.23. I'm
still having the same problem.
What should I do now?
Rob
Rob wrote:
Additional info:
hci0: Type: USB
BD Address: 00:18:E7:22:XX:XX ACL MTU: 1017:8 SCO MTU: 64:8
UP RUNNING PSCAN ISCAN
RX
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I've have installed a new Asterisk 1.4.15 system after having previously
used a 1.2 CVS head (from 10 Sep 2005). Both systems are pentiums though
the newer one is actually a slower processor.
On the new system, playback of gsm files is noticeably
hi,
I had a similar problem with FreePBX last year, follow this mail
thread, it might help solve ur problem
http://lists.digium.com/pipermail/asterisk-users/2006-October/169568.html
On 12/13/07, Paul Hales [EMAIL PROTECTED] wrote:
Using the 'read' function you should be able to do something
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