We have Panasonic D500 and Asterisk with TE110P hybrid setup successfully
and it is possible to route DID to a PRI card in D500 if it is set up as EXT
card, and extension number of PRI card is defined for example as 2XX. This
gives you opportunity to have extensions 200 - 299 routed to PRI
looking at setting up an asterisk box at my home-- what VOIP providers
are you all using with the best results (and low costs! lol)
thanks
Bernie
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Adam Goryachev wrote:
On Mon, 2005-07-18 at 23:04 -0500, Kristian Kielhofner wrote:
The new firmware and bootrom already require 4mb flash, which the 301,
501, and 600 have. You can't load firmware 1.5.2 on the 300 or 500!
Are you sure of that?? I don't recall seeing that noted
hello
how can i install meetme application without Zaptel
interface. and if this is not posible then how to
install zaptel module.
any helpful link
thanks in advance
Kamran
__
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam
Adam,
Who did you buy them off?
Kind regards
Michael Felder
IT Medic Australia Pty. Ltd.
P: 03 9557 2213
F: 03 9557 2214
M: 0419 568 217
E: [EMAIL PROTECTED]
http://www.ITMedic.com.au
Keeping your computer systems healthy.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Hi,
-Original Message-
So I won one of these on ebay, in the auction it says it has the RJ45
ports on it but it doesn't :(
If I were to get an analog adapter would I be able to use the video
portion of this or am I SOL? The auction requires me to pay for
shipping back, so I end
On Tue, 2005-07-19 at 02:02 -0400, Bernie Courtney wrote:
looking at setting up an asterisk box at my home-- what VOIP providers
are you all using with the best results (and low costs! lol)
thanks
Bernie
That is a hard question based on what you have stated. First if you are
using
The IP300 will run SIP v1.5.2, as will the IP500 and IP600. I have tested
them with bootrom v1.6.2
The issue is not that the IP300 and IP500 phones do not have enough memory
to run SIP v1.5.2, it is that they do not have enough memory to run bootrom
v3.x.x AND sip v1.5.2. The added features of
The size of the sip.ld file is NOT an indication of the size of the firmware
for a particular model.
The sip.ld file is a coposite file that encapsulates individual firmware for
the different models. IE: the sip v1.4.1 sip.ld image contains individual
firmware for the IP300, IP500, IP600 and
dbruce wrote:
The IP300 will run SIP v1.5.2, as will the IP500 and IP600. I have tested
them with bootrom v1.6.2
The issue is not that the IP300 and IP500 phones do not have enough memory
to run SIP v1.5.2, it is that they do not have enough memory to run bootrom
v3.x.x AND sip v1.5.2. The
dbruce wrote:
The size of the sip.ld file is NOT an indication of the size of the firmware
for a particular model.
Yes, but it is an indication of the overall size of the firmware image,
regardless of the model, and that is what I was talking about.
The sip.ld file is a coposite file that
Hello,
I have the exact same question as you. Did you find an answer?
We are using asterisk at the office and the incoming line is an ISDN
(HFC-PCI card with zap_hfc driver from bristuff 0.2.0 RC3a).
And I have a problem, when both ISDN B channels are in use (i.e. 2
calls in progress) it
David Burgess wrote:
Matt Riddell wrote:
David Burgess wrote:
Hi,
I am new to the list.
I have just *re*-installed and rebuilt asterisk from the head
branch and I am left with the problem that the sound bounces
around. When installing zaptel I get the following message from
ztcfg.
Noah,
I've encountered the same problem (same negative timestamp log
messages, and garbled audio). The last version of CVS HEAD that worked
on my system without any problems was July 4...
Steve
On 7/18/05, Noah Miller [EMAIL PROTECTED] wrote:
Hi -
I've just been testing out the latest CVS
Dear All,
I am currently working on asterisk cvs-head version in order to use
realtime with mysql, 2 asterisk servers with duplicate mysql databases,
one asterisk server is serving the sip phones and the data is logged to
the database and replicated to the other asterisk database, when the first
Would like to exchange with Sphinx users, ...
bye
Ronald
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hi,
Adeel Ali schrieb:
Jul 18, 2005 2:55:13 AM net.sf.asterisk.util.impl.JavaLoggingLog info
INFO: Received connection.
Jul 18, 2005 2:55:14 AM net.sf.asterisk.util.impl.JavaLoggingLog error
SEVERE: Resource bundle fastagi-mapping is missing.
Jul 18, 2005 2:55:14 AM
I have been trying to make Broadvoice match incoming Caller ID and do
specific things based on the number received, but due to Broadvoice
requiring the s to start off an incoming extension, I cannot get
this to work. Has anyone been able to do this? Here are some examples
of my setup:
from
I'm also getting a lot of the following sip messages as well under FreeBSD.
sip_xmit of 0x87fb01c (len 507) to 192.168.123.94 returned -1: Address
family not supported by protocol family
- Original Message -
From: Steve Hsieh [EMAIL PROTECTED]
To: Asterisk Users Mailing List -
weicheng jiang [EMAIL PROTECTED] wrote:
I just talked to someone at http://www.luxoncomm.com,
he said authentication via http will be supported in
the next firmware release (2.8), due out in a couple
of weeks.
While we're at it - is anybody aware of a 802.11g capable phone? My home
WLAN is
Alexander Szlezak [EMAIL PROTECTED] wrote:
System, while not even one of the Billion cards work in my AMD Athlon
2400+ (Via) Chipset System (pci performance to low - you might have
some cpu throttling enabled ). Maybe it's just the board/bios etc.
I get this message also in my VIA KT133
hi,
im installing latest asterisk from cvs on solaris
9. but when i run make i got this error
/bin/sh: build_tools/make_version_h: cannot
executemake: *** [include/asterisk/version.h] Error 1
what i did was chmod 777 all files under
asterisk/build_tools/
and when i run make again i got
Arik Funke [EMAIL PROTECTED] wrote:
I am looking for inexpensive isdn card that supports nt mode with asterisk.
HFC. Get the cheap 25 Euro card from Conrad. (955078)
on the PBX4Linux page: http://isdn.jolly.de/cards.html Or will asterisk
work ONLY with cards with hfc chipsets?
Let's say it
Actually, your logic is flawed...
It would be correct id the images contained in the composite file were
static, but they are not...
sip.ld v1.4.1 (approx. 7.7 meg) contains 5 images: IP300 , IP500, IP600,
IP4000 and one unknown image (probably for the short lived IP100/IP400
customized model).
The bandwidth calculator at
http://www.asteriskguru.com/tools/bandwidth_calculator.php
will tell you exactly how much bandwidth your calls will take, in both
directions.
Zoa.
Karlheinz Hagen wrote:
How can I calculate the quantity of concurrent users using a
bandwith of 512Kbps ?
I have a number of internal SIP phones and a number of external SIP clients
with the server running Asterisk on the boundary between the two. ie the server
has two network cards with an internal private address and an external public
address. For security reasons no routing is allowed between
Maybe it is not just me going crazy. I have garbled audio based on
July 17. And I thought it was me messing up the ztdummy setup.
David Burgess
On 7/19/05, Steve Hsieh [EMAIL PROTECTED] wrote:
Noah,
I've encountered the same problem (same negative timestamp log
messages, and garbled audio).
Zoltan Szecsei wrote:
David Burgess wrote:
Matt Riddell wrote:
David Burgess wrote:
Hi,
I am new to the list.
I have just *re*-installed and rebuilt asterisk from the head
branch and I am left with the problem that the sound bounces
around. When installing zaptel I get the
I though that Asterisk would transcode between codecs! All my SIP devices
support G729a 711a/u but my ISDN PRI/BRI lines use 711a/u. Asterisk is quite
happy to accept a call from a SIP device using G729a and then complains that it
can't translate into G711 to go onto the ISDN network. Does
Bernie Courtney wrote:
looking at setting up an asterisk box at my home-- what VOIP providers
are you all using with the best results (and low costs! lol)
thanks
Bernie
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Hi,
I set alaw = 1-7 in /etc/zaptel.conf hoping to make my zap channels
the same as the PSTN. This caused the levels to be about 20dB too high,
as well as being distorted. I adjusted the txgain and rxgain settings in
/etc/asterisk/zapata.conf to sane levels, but now I still have rather
distorted
Hi,
I'm currently building an asterisk system which should work as gateway
between SIP phones and ISDN. Most parts are working very fine, but one
problem occurs and I am not able to solve or debug it.
Telephony from ISDN to SIP (a Sipura Hardphone) is working very well,
but if the SIP Phone
AFAIK you need a license from Digium if you want to transcode to/from
G729a...
Hope this information is correct and it helps
Regards
Guido Hecken
I though that Asterisk would transcode between codecs! All my SIP devices
support
G729a 711a/u but my ISDN PRI/BRI lines use 711a/u. Asterisk is
You could try somthing like this in extensions.conf:
[broadvoice-incoming]
exten = s,1,AGI(db.agi)
exten = s,2,Answer
; handle case of no CallerID
exten = s,3,gotoif($[${CALLERIDNUM} = ]?broadvoice-nocallerid|s|1)
; special handling for specific NPANXX
exten = s,4,gotoif($[${CALLERIDNUM:0:6} =
Madhawa Jayanath wrote:
o Bernie,
1) best results www.nufone.net
2) low cost www.voipjet.com
Anyone able to find NuFone's rates? I have been looking for them on
their site. I need international rates and UK Mobile.
--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int: (305)
On Tue, 19 Jul 2005, Torsten Hoefler wrote:
Hi,
I'm currently building an asterisk system which should work as gateway
between SIP phones and ISDN. Most parts are working very fine, but one
problem occurs and I am not able to solve or debug it.
Telephony from ISDN to SIP (a Sipura
I have tidied up the script and added some help text, feel free to
download and maybe improve.
http://www.masonc.com/phoneconf
Usage: Usage: ./phoneconf [config|help] phonemacaddress extention
username context
./phoneconf help will print syntax info
./phoneconf 0004f201aa11 500 MyPhone
Hi,
Is there any way to pick up a remote phone/extension wich is ringing
for a long time but no one is availabe to answer that call? OR the
scenario is like, suppose my extension is 1000, and I am working at a
colleagues desk. At this time my phone is ringing in my desk and I
want to pick up the
hi
we are also working in asterisk pbx .and we have try it with softphone (diax
and xlite) it works good
but the problem we got is when installed the TDM04B and try to phone the pstn
we got this problem.;
ul 19 11:25:33 NOTICE[2924]: app_dial.c:764 dial_exec: Unable to create channel
of type
I though that Asterisk would transcode between codecs! All my SIP devices
support G729a
711a/u but my ISDN PRI/BRI lines use 711a/u. Asterisk is quite happy to accept
a call from a SIP
device using G729a and then complains that it can't translate into G711 to go
onto the ISDN
network.
Hi all!, i
search for some information about to setup my asterisk box withe-mail
notification when a I call the voicemail application. Voicemailapplication
works fine in the Dial Plan but nothing happens with emailnotification ...so
what i need to know about this?...wiki pages did not
Hello,
I found, that in CVS Head, in chan_sip.c, there's some support of
asterisk. I have one question -- how does it map extensions to sip
user names? When my client subscribes to other extensions' presence,
they see all users online, but it may be because of voicemail
fallback. Is there a way
I was waiting for everyone to reply so here is mine.. Check out the Mediatrix
web site. There are no downloads or lists of resellers who might have this
provisioning software that is normally included with purchase.
I'm not interested in your relationship with voipsupply.
Mediatrix does
Message is no translator path exists for channel type CAPI (native 8) to 256
[EMAIL PROTECTED] 19/07/05 13:44:29
I though that Asterisk would transcode between codecs! All my SIP devices
support G729a
711a/u but my ISDN PRI/BRI lines use 711a/u. Asterisk is quite happy to accept
a call
On 18/07/05 17:06 -0700, Michael D Schelin wrote:
I was waiting for everyone to reply so here is mine.. Check out the
Mediatrix web site. There are no downloads or lists of resellers who might
have this provisioning software that is normally included with purchase.
You may be
Had the same problem and found that chan_capi-cm solved it. Watch out about the
change to the dial command syntax which not does not let you specify outgoing
MSN. It now seems impossible to specify outgoing MSN. The excellent perl
Op_panel also seems unable to show events on a CAPI line button
Had not seen a response on the following question - wondering if
anyone may have any insight on this?
Original Question-
Is there a way to log SIP response codes without enabling verbose
logging? Reason being is that from time to time I see a call fail on
our primary provider and
Rich Adamson wrote:
I though that Asterisk would transcode between codecs! All my SIP devices
support G729a
711a/u but my ISDN PRI/BRI lines use 711a/u. Asterisk is quite happy to accept
a call from a SIP
device using G729a and then complains that it can't translate into G711 to go
On Tue, 2005-07-19 at 05:40 -0400, Nate Kapi wrote:
[snip]
[broadvoice-incoming]
exten = s/760899,1,AGI(db.agi)
exten = s/760899,2,Answer
exten = s/760899,3,Goto(menu,s,1)
exten = s/_760899,1,AGI(db.agi)
^^^
exten = s/_760899,2,Answer
^^^
exten =
I am new to this list. What is the process of reporting something that is
completely busted? Or do we just patiently wait?
On 7/19/05, Chris Stenton [EMAIL PROTECTED] wrote:
I'm also getting a lot of the following sip messages as well under FreeBSD.
sip_xmit of 0x87fb01c (len 507) to
What does your 'show translation' look like?
Can you copy/paste the specific *.conf entries for the sip devices
and capi?
Message is no translator path exists for channel type CAPI (native 8) to 256
[EMAIL PROTECTED] 19/07/05 13:44:29
I though that Asterisk would transcode between
Juraj Bednar wrote:
Hello,
I found, that in CVS Head, in chan_sip.c, there's some support of
asterisk. I have one question -- how does it map extensions to sip
user names? When my client subscribes to other extensions' presence,
they see all users online, but it may be because of
Yes, I've purchased 20 G729a licenses and I know that * uses them OK
[EMAIL PROTECTED] 19/07/05 13:27:04
Rich Adamson wrote:
I though that Asterisk would transcode between codecs! All my SIP devices
support G729a
711a/u but my ISDN PRI/BRI lines use 711a/u. Asterisk is quite happy to
I'm having some weird behavior happening with my current configuration.
I'm running asterisk 1.0.9 and I've tried placing outbound calls using the
Originate action in the Manager API.
I'm following directions from the voip-info wiki on placing a call from an
outgoing channel.
1st action:
The way I fixed this with another provider that had similar behavior
was to patch chan_sip.c so that it pulled out the DNIS value from the
to: tag in the SIP header and then threw that it into the DNID channel
variable. Then, I took the common extension ('s' in your case) and did
a Goto with the
When I'm connected with two clients in the same LAN of the asterisk server
using the same codec and the IAX protocol, I have no media passing through
the asterisk. But, when these clients are on the Internet, all the media
flow pass through the asterisk server. Is that way that it works? I
On Tue, Jul 19, 2005 at 08:52:14AM -0300, Gustavo A. Gonzalez wrote:
Hi all!, i search for some information about to setup my asterisk box with
e-mail notification when a I call the voicemail application. Voicemail
application works fine in the Dial Plan but nothing happens with email
On Jul 18, 2005, at 3:13 PM, Michael D Schelin wrote:
Here is a letter I sent them for my $150 paper weight.
The forum is not a place to post ransom notes. You've added zero
benefit to any reader here, nor to yourself, since you didn't actually
ask a question in your email.
On Tuesday 19 July 2005 01:10, trixter http://www.0xdecafbad.com wrote:
the same thing as paper. I have had multiple systems with the same mac
address on the same network for other purposes and did not have problems
with those systems talking to each other or other devices on the network
I'm
On Monday 18 July 2005 18:13, Michael D Schelin wrote:
first time, I'm very disappointed that you have removed the
configuration CD that should have been shipped with the Mediatrix 2102
just to get a few more bucks. I have contacted mediatrix and they have
informed me that the CD's is shipped
Hi,
I am facing some
problem with DBdel it says Error deleting key from database. Please let me know
whats going wrong.
here is the syntax
what iam using
exten =
_*73.,1,DBdel(CF/${EXTEN:3})
thanks
Somesh
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Maybe it is not just me going crazy. I have garbled audio based on
July 17. And I thought it was me messing up the ztdummy setup.
I've encountered the same problem (same negative timestamp log
messages, and garbled audio). The last version of CVS HEAD that
worked
on my system without any
Why didn't I think of using that command...
It shows all - for G729a which is presumably why I'm having a problem
I have purchased 20 licenses from Digium, downloaded binary, registered the
binary correctly, placed it in the correct directory and it is listed
specifically in SIP.conf
I'm sure
Wendell Almeida Silva wrote:
When I'm connected with two clients in the same LAN of the asterisk server
using the same codec and the IAX protocol, I have no media passing through
the asterisk.
That is because IAX has a SIP-like reinvite mechanism. So by default,
whenever it can, asterisk will
On Tuesday 19 July 2005 08:28, David Burgess wrote:
I am new to this list. What is the process of reporting something that is
completely busted? Or do we just patiently wait?
If you're running CVS HEAD be advised that this is the live development code,
it mostly works most of the time but
On Monday 18 July 2005 18:11, Chad Osmond wrote:
Remove Callerid and set immediate=yes
Callerid is sent between the first and second rings, so asterisk has to
wait for it.
There should be no need for immediate=yes on FXO POTS ports. The first ring
will cause the dialplan to be entered at
Wow ! No reply... May be I must talk about X100P instead of X101P ?
Is someone has yet encountered this kind of no sound problem when bridging
two FXO lines like this (first as input, second as output) ?
Any idea ?
TIA.
Best Regards,
Francois BERGERET,
France.
- Original Message -
Hi,
New version of chan_capi-cm 0.5.4 has fixed my problem.
Regards,
Cyrille
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Cyrille
Demaret
Envoyé : vendredi 15 juillet 2005 11:13
À : asterisk-users@lists.digium.com
Objet : [Asterisk-Users] Strange
how and where to check the archives?
i am a newer for asterisk.
thanks
2005/7/18, C F [EMAIL PROTECTED]:
This is the first time I hear of this idea, this is a great idea. Why
don't you check the archives, since if you are the first one you might
win $10,000 from Digium.
On 7/18/05, Liu
On Monday 18 July 2005 16:54, Erik Espinoza wrote:
I once tried to call in support for digium for 4 IAXy's that I
purchased ($400). They told me to e-mail this mailing list. I
appreciate all the hard work that they did to produce asterisk, I just
don't trust this company to support anything.
asterisk support pickup function.
set the numbers in same pickupgroup.
2005/7/19, Yusuf Iqbal [EMAIL PROTECTED]:
Hi,
Is there any way to pick up a remote phone/extension wich is ringing
for a long time but no one is availabe to answer that call? OR the
scenario is like, suppose my
I'm just using some generic GE phone (analog) connected to a linksys
pap2.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Totaro
Sent: Monday, July 18, 2005 7:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
On Monday 18 July 2005 23:50, Kevin Walsh wrote:
Services could be provided, and money could be made, without resorting
to selling closed source versions of the product. Apparently, the
closed version consists of the contents of CVS HEAD, with various
changes made to increase reliability and
We are currently using Asterisk 1.0.8 and noticing a rather odd problem
that occurs roughly during the middle of the day sometimes. It seems
that Asterisk and Bellsouth are getting out of sync somehow on the the
status of the PRI channels. I get warnings messages like:
Jul 18 10:50:24
Right, so before resolution can be
attempted or had, you lash out on the forum.
If they had told you to stuff it or had
just ignored you, you might have something to complain about.
You are pissed that the ATA is not web
configurable? How in the hell is that VoipSupplys fault? You bought
no translator path exists for channel type CAPI (native 8) to 256. I
understood that Asterisk
You don't mention whether you successfully registered the licenses.
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Welcome aboard.
the list is located at: lists.digium.com
to search just the list use google with site:lists.digium.com
the wiki is located at: www.voip-info.org/wiki-asterisk
IRC is at irc.freenode.net #asterisk
On 7/19/05, Liu Peter [EMAIL PROTECTED] wrote:
how and where to check the
Hi,
I have a bunch of NetPhones that I have bought from MeritCall some time ago for
their service. How can I use this phones (supposed SIP phones) to integrate
with a Asterisk Setup.
I have seen a manual for a similar one but I don´t know If mine are hardcoded
in some way. This devices are
Hi,
I'm trying to configure a quadbri card using the configuration
found in Bristuff.
I know the configuration of telco is point-to-point
and I think the card have to work in NT mode (I presume because I have
not found the documentation about this and when attach
to the ISDN the led become
On 7/19/05, Yusuf Iqbal [EMAIL PROTECTED] wrote:
Hi,
Is there any way to pick up a remote phone/extension wich is ringing
for a long time but no one is availabe to answer that call? OR the
scenario is like, suppose my extension is 1000, and I am working at a
colleagues desk. At this time my
We have run into the very same problem with support. We had one issue where it
was identified as a hardware problem, 1 port out on a four port card. We were
promised a quick replacement. We called back a week later and it still hadn't
been sent. They finally overnighted it, but I was not
On Tue, Jul 19, 2005 at 03:57:59PM +0200, Francois BERGERET wrote:
Wow ! No reply... May be I must talk about X100P instead of X101P ?
Is someone has yet encountered this kind of no sound problem when
bridging two FXO lines like this (first as input, second as output) ?
Any idea ?
TIA.
What
Hi,
i need an recommandation for an ATA adapter to use with an anlog fax maschine.
I would appreciate any hints.
Regards!
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To
Paul Dracevich wrote:
Can that use real time Database.
No, as previously mentioned many times on this list, the 1.0.x series
does not support Realtime.
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..all i know is that if this guy is bitching about a non-existant CD
and is unable to provision
a simple VOIP device, then i would be terribly afraid of his companies
technical ability for their actual VOIP service..
scary..a VOIP provider that can't even provide themselves...
On 7/19/05,
Would anyone care to boast about how they use their hard phone screens?
For instance, could a person place a variable on the screen that indicates
how many waiting calls there are in a certain queue? Could you program the
screen to display any data? Current wait time in a certain queue?
Or am I
Lee Howard wrote:
HylaFAX can use E1 and T1 fax modems just fine (24 and 30 channels each).
Yes ok, with special and expensive hardware. (€ 7000 )
Furthermore, HylaFAX also supports multiport modems (usually up to 8
ports each).
Ok but then I have the E1 problem, I don't have so many anlog
Hi
I am doing
PSTN - Asterisk - SIP - yate - H323 - Telco
When user intiate a call from asterisk, it is pass to yate for SIP - H323 signalling,
and forward the calls to telco. Everything is fine there.
My problem is, i am not getting an actual PSTN ringing tone.
instead i am getting a fake
Hi
I am doing
PSTN - Asterisk - SIP - yate - H323 - Telco
When user intiate a call from asterisk, it is pass to yate for SIP - H323 signalling,
and forward the calls to telco. Everything is fine there.
My problem is, i am not getting an actual PSTN ringing tone.
instead i am getting a fake
All,
Is it possible to have a list of IP addresses to register to?
Example: Service provider has multiple register locations, DC, Chicago,
LA etc...
If one register site is down then I should be able to register to a
different site
automatically.
Is this possible?
THanks,
Jerry
I will 2nd the nufone vote..
awesome service.. haven't had any outages since i have had their
service.
Nufone also has free incoming 18xx DID's too, you just have to pay for
the 0.02/min usage
On Tue, 2005-07-19 at 03:24, Madhawa Jayanath wrote:
Bernie Courtney wrote:
looking at
I am a little pissed when
all other ATA's are configurable from their built in web server.
The 2102 does have a built in Web server.
See manuals at support.bctgroup.ru/mediatrix/2102/
If you have a refurbished unit, perhaps the web server was disabled,
or the password was changed. Try reset to
from the bottom of http://www.voipsupply.com/product_info.php?manufacturers_id=16products_id=334
...
With the Mediatrix 2102, service providers get the product characteristics allowing them to successfully deploy residential IP telephony applications. The Mediatrix 2102 provides a web
Probably doesn't help diagnose the problem but there were also audio
problems experienced with
this cvs version even on LAN / sip2sip / no transcoding
ERROR[1171] UTILS.C:509 TVFIX: WARNING NEGATIVE TIMESTAMP -194931. ...
I will be looking into this issue later today.
Kib Eki wrote:
Hi,
i need an recommandation for an ATA adapter to use with an anlog fax
maschine.
I would appreciate any hints.
There are some hints at http://www.soft-switch.org/foip.html :-)
Regards,
Steve
___
Asterisk-Users mailing list
[EMAIL PROTECTED] ha scritto:
I know the configuration of telco is point-to-point and I think the card
have to work in NT mode (I presume because I have
not found the documentation about this and when attach to the ISDN the
led become green).
If you're connecting the card to a NT1/NT1+, then
Why are there so many attempts to native bridge? The call is actually up
and working by attempt #1 so what is it doing on all those other
attempts? We are only allowing G729 so it can't be codec negotiation.
-Matthew
-- Executing Goto(SIP/3013-3dfa, cytel-outgoing|917034439032|1)
in new
On Tuesday 19 Jul 2005 14:45, Martin Sutherland wrote:
Why didn't I think of using that command...
It shows all - for G729a which is presumably why I'm having a problem
I have purchased 20 licenses from Digium, downloaded binary, registered the
binary correctly, placed it in the correct
Wouldn't it have been easier to do...
register = user:[EMAIL PROTECTED]/1234
register = user:[EMAIL PROTECTED]/2345
And then create a dialplan for extensions 1234, 2345, etc?
-Original Message-
From: BJ Weschke [mailto:[EMAIL PROTECTED]
Sent: Tuesday, July 19, 2005 8:11 AM
To: Nate
What is the CLI output you getting?
On 7/19/05, someshwarak [EMAIL PROTECTED] wrote:
Hi,
I am facing some problem with DBdel it says Error deleting key from
database. Please let me know whats going wrong.
here is the syntax what iam using
exten = _*73.,1,DBdel(CF/${EXTEN:3})
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