On Mon, Nov 13, 2006 at 11:20:27PM +0100, Christian wrote:
Hi,
Thanks for that info so i need to install the mpg123 player?
I am not able to do make mpg123 as before.
mpg123 is availble, as usual, as a deb package. Etch/Sid have a decent
version of it, unlike the old and buggy version in
On 11/13/06, Vicky [EMAIL PROTECTED] wrote:
IF your asterisk server is behind NAT and no port forwarding is done then
how can that static ip user/device reach it . You will have to
keep asterisk server in static ip or do
port forwarding to accept connections from
outside .
i've understand but
Hello All.
I am stumped, please help me out..
I have the following setup:
VOIP provider = VOIP GW (asterisk GW1) = VOIP server (asterisk - VS1)
The gateway is there to get around the limitations running on the VOIP
server.
I can call out from and receive calls VS1 no problems at all.
Where is your DMZ pointed?
Best regards,
Al Bochter
Bochter Services
http://www.BochterServices.com/?t=Email
Are you outside of the US?
Do you need to call US Toll Free Numbers?
We can help you save money on calling US toll free numbers.
Email for information: [EMAIL PROTECTED]
(Cellular)
On 13 Nov 2006, at 13:15, Ondrej Valousek wrote:
Hi Dean,
I will check that site - thanks for the hint.
The biggest problem I see with authentication and I do not think
mexuar could help me here (and I am definitely going to pay $2000
for it :-)
But it is another story...
Well, it
Hi there, I am looking around, is there anyone did any integration asterisk talk to / connect to MS SQL? Thanks-- Regards, Sharon Lim *Good memories are to be folded neatly and tucked away into the back pocket *
___
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On 13 Nov 2006, at 16:18, Tzafrir Cohen wrote:
On Mon, Nov 13, 2006 at 07:10:12AM -0500, Brian Rogan wrote:
On Mon, Nov 13, 2006 at 12:46:14PM +0100, nik600 wrote:
Hi
i have an application developed with bayonne.
Recentely i'm experiencing some problems and i am planning to
migrate
to
Yes asterisk can do that . If you mena for call records then see http://www.voip-info.org/wiki-Asterisk+cdr+mysqlAlso see
http://www.voip-info.org/wiki/view/Asterisk+cmd+MYSQLOn 14/11/06, Sharon Lim [EMAIL PROTECTED]
wrote:
Hi there, I am looking around, is there anyone did any integration
I figured out the problem. In one of the parameters' name, one underscore was missing. That was causing the whole thing to not work. It was like thisExtension_2_ 1/Extension_2
_This underscore marked in red was missing.Once I typed that underscore, everything went back to normal, and now all
How about meassuring it directly? For starters, take a look at zttest.c .
(Though it could use some slightly better accuracy).
Not sure how accurate is zttest.c.
Will run some test to see it's accuracy.
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HI, all
Can I disable send e-mail feature in the voicemail application?___
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To UNSUBSCRIBE or update options visit:
Dear all,
I just want to change the format of the VM_DATE (used in the mail sent
to the user).
I'm french and the format is not very acceptable for my users
Friday, November 10, 2006 at 04:28:43 PM could be replace by
Vendredi 10 Novembre 2006 à 16h28
Any idea ?
Thanks a lot,
--
Kelly,
Could there be a mismatch at the branch switch? Such as ethernet
interfaces operating at half-duplex when the switch is at full-duplex.
This usually manifests itself as dropped packets. I have an older Dell
box that cannot seem to negotiate with a Cisco switch. 50% of the time
it comes
Ondrej Valousek napisał(a):
Hello Michal,
Thank you for the hint!
Can I ask you for your script so I have some idea how it works?
I have apache already running on my * box.
OK. Python code for my script is below. Read it while you must prepare
it to your
environment (prefixes, cellular
exten = _91900NXX,1,Playback(the-number-u-dialed)
exten = _91900NXX,n,Playback(has)
exten = _91900NXX,n,Playback(a-connect-charge)
exten = _91900NXX,n,Playback(sorry-cant-let-you-do-that2)
exten = _91900NXX,n,Busy()
exten = _9NXX976,1,Playback(the-number-u-dialed)
exten =
Hi,
I've recently got some Polycom 501 601 phones.
I have buddy watch working showing the status of users listed in the
directory.
I would like to also have the status of the trunks (ZAP via TDM2400E SIP)
on the IP601 Sidecar display, but I cannot so far find any info on this?
Thanks,
I am planning to use asterisk with Digium TDM2404E card as a media
gateway to terminate traffic to Cell phones. Anyone got this working
before with no problems, specially with Answer/Disconnect supervision?
Thanks
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On 11/13/06, Dovid B [EMAIL PROTECTED] wrote:
How much did the hardware cost you to set this up for your door ?
From memory... the strike was around $95, the relay board between $25
and $50, and the power supply was only a few dollars, so you could do
it all for under $200.
--
2006/11/10, Anselm Martin Hoffmeister [EMAIL PROTECTED]:
I'd go with parallel softphones on LAN-connected and/or WLAN-connected
PCs and see wether they have the same problem. That could rule out the
provider or confirm the WLAN or WLAN phone implementation make the
problem.
We have tested a
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Peter Howard
Sent: 14 November 2006 00:38
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Problem found Re: [asterisk-users] Headaches with
Video over SIP
Found the problem.
You need to install unixodbc also
-Message d'origine-
De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Tony Mountifield
Envoyé : mardi 14 novembre 2006 11:22
À : asterisk-users@lists.digium.com
Objet : [asterisk-users] Re: Is asterisk able to integrate with MS SQL
In
In article [EMAIL PROTECTED],
Sharon Lim [EMAIL PROTECTED] wrote:
Hi there,
I am looking around, is there anyone did any integration asterisk talk to /
connect to MS SQL?
Look for the package FreeTDS and install it. Then build Asterisk and it will
include the TDS driver that can log CDRs
Haha cool, I'm the Belgian guy in Sofia you've met before, i asked for
exactly the same reason as you :)
Zoa.
Anton Tinchev wrote:
Zoa wrote:
Can you tell us how you do the testing ?
3-4 different ways. All gives same results, so test are pretty valid.
1. Interrupt counting inside the
If you don't mind using linux, linux can do some fairly intense load
balancing all built in. Check out the Linux Virtual Server project. As
for WAN failover, if you again don't mind using linux, you can script a
simple ping to the internet (I would ping at least 3 hosts) and if that
fails, fail
Which card, BTW?
TDM400 analog
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http://lists.digium.com/mailman/listinfo/asterisk-users
Us Toll Free is only 800, 866, 877 and 888
--
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Steven
http://www.glimasoutheast.org
Doug Crompton [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
Ok so ONLY 900 numbers are pay.
Next question 18XX numbers. are they all toll free? Is there any
space in 8xx that is
Dear All,
I have this problem which is preventing me from switching to
voip system andstill working on that old siemens pbx, we have fax machines
that we attached to ATA called planet and when we try to send a fax locally
between the fax machines it works great but when we try to get a fax
I'm extremely pleased with the ZyWall 35 (ZyXel) - well worth it at
about GBP350. The Linksys and Netgear dual-WAN routers are a pile of
rubbish.
The ZyWall has good management tools and a lot of options. Rock solid
in use.
Todd- Asterisk wrote:
I've been looking for this as well.. I
oops sorry i thought its my sql didnt notice it's MS SQL :D On 14/11/06, Tony Mountifield [EMAIL PROTECTED]
wrote:
In article [EMAIL PROTECTED],Sharon Lim
[EMAIL PROTECTED] wrote: Hi there, I am looking around, is there anyone did any integration asterisk talk to / connect to MS SQL?Look for the
On Tue, 14 Nov 2006, Doug Crompton wrote:
Ok so ONLY 900 numbers are pay.
Next question 18XX numbers. are they all toll free? Is there any
space in 8xx that is used otherwise?
Whoa. No, only 800, 888, 877, 866 and futuree NPA's 855, 844, 833, and 822
are toll free.
Here in
We've never used it in a load balancing situation
but it works great in a failover config.
- Original Message -
From:
Dean Collins
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Tuesday, November 14, 2006 10:41
AM
Subject: RE:
Here's a question maybe someone can help me with:
My extension looks like this:
exten = 1006,1,MP3Player,http://audio-mp3.ibiblio.org:8000/wcpe.mp3
When I try this extension, the following output appears in the CLI:
Nov 13 12:47:51 NOTICE[8422]: app_mp3.c:111 timed_read: Poll timed out/errored
Hi all,
I have one interface zaptel
TE110P with 10 channels enabled for my service provider. Alone the first
channel is enable to receive call from this service provider and when a make
call using other channel, like Zap/2, the channel is enable from the PBX
Siemens EWSD version 10 to
Are you looking for load balancing or
failover.
Also is there a cheaper way of
implementing load balancing than $845 appliance?
Cheers,
Dean
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Todd- Asterisk
Sent: Tuesday, 14 November 2006
9:26 AM
To:
Doug Crompton wrote:
I had a 19xx rule in asterisk and realized when I was trying to dial an
area code 978 in MA that that was not a good idea. Is there a more defined
rule for 900 space of non pay vs. pay codes?
_1900NXX
_NXX976
___
LONDON, UK (14th November 2006) - Bicom Systems announced today it has
released its first freeware software to the Asterisk Community, OutCall.
This is to be the first of similar releases of proprietary tools that can
assist users with getting the most out of Asterisk and will also be released
just dont enter any email address while creating extension / mailbox ;)On 14/11/06, Ma Zhiyong [EMAIL PROTECTED]
wrote:
HI, allCan I disable send e-mail feature in the voicemail application?___--Bandwidth and Colocation provided by Easynews.com --
Hi Jason,
I was looking for an external solution outside of my asterisk box so
that I can load balance my other website/email traffic as well.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Jason
Sent: Tuesday, 14
I've been looking for this as well.. I need to support up to 20 VOIP phones over Internet as the Asterisk server is off-site. We'll have multiple cable modems or DSL routers. I found this device which looks promising - does anyone have any experience with this?
810 is an area code in Michigan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Doug
Crompton
Sent: Tuesday, November 14, 2006 10:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 900 rules
Ok so ONLY 900
There are several Caribbean countries within the 8XX range, as well as
more toll free, and regular area codes.
Here's the full list at NANPA:
http://www.nationalnanpa.com/nas/public/npasInServiceByNumberReport.do?method=displayNpasInServiceByNumberReport
Doug Crompton wrote:
Ok so ONLY 900
Dean,
A small Linux box will make a very effective router (and firewall if
required) and give load balancing/failover capabilities. I've done it in
the past (many moons ago!)
A link from my bookmarks: http://lartc.org/ - can be a little scary
depending on your knowledge of ip routing and linux
Tim Panton wrote:
On 13 Nov 2006, at 13:15, Ondrej Valousek wrote:
Hi Dean,
I will check that site - thanks for the hint.
The biggest problem I see with authentication and I do not think
mexuar could help me here (and I am definitely going to pay $2000 for
it :-) But it is another
I just received two TDM400P cards, but I'm having problems with them.
The full info is at:
http://pastebin.com/824079
Extra, I'm using :
libpri-1.2.3
zaptel-1.2.10
On a x86 stable Gentoo box.
Kernel: 2.6.17
gcc-4.1.1, glibc-2.4-r4
Is that an hardware problem? Should I try the other card?
--
I had a 19xx rule in asterisk and realized when I was trying to dial an
area code 978 in MA that that was not a good idea. Is there a more defined
rule for 900 space of non pay vs. pay codes?
Doug
___
--Bandwidth and Colocation provided by Easynews.com
Ok so ONLY 900 numbers are pay.
Next question 18XX numbers. are they all toll free? Is there any
space in 8xx that is used otherwise?
Doug
On Tue, 14 Nov 2006, Eric ManxPower Wieling wrote:
Doug Crompton wrote:
I had a 19xx rule in asterisk and realized when I was trying to dial
http://hotbrick.com/
- Original Message -
From:
Dean Collins
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Tuesday, November 14, 2006 9:38
AM
Subject: RE: [asterisk-users] Dual Wan
Router with Failover
Are you looking for
load
Sorry for the crosspost (this was also posted to
asterisk-at-uc-dot-org) but I haven't got a response.
I have a cell phone added to a queue as a local extension (member =
Local/299). I want the cell phone to be able to reject calls to the
queue without the person sitting in the queue being hung
Not all 8XX numbers are free. 800, 888, 877, 866, ans 855 are free. I
don't remember when 855 is scheduled to start being issued.
Try checking your local phone book.
Doug Crompton wrote:
Ok so ONLY 900 numbers are pay.
Next question 18XX numbers. are they all toll free? Is there
On 11/14/06, Stelios Koroneos [EMAIL PROTECTED] wrote:
JR Richardson gave a very nice presentation at Astricon on how to do that with
DUNDI
As I understand it JR Richardson's DUNDi solution does not support
IAX. It uses regcontex which I believe is only available with SIP.
(please correct me
Sweet, now that is interesting
http://hotbrick.com/produto.asp?tipo=2codPro=22
anyone have any comments on the load
balancing capability of these?
Cheers,
Dean
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Mailing List
Sent: Tuesday, 14
you could use any number of the linux firewalls out there then. I know
some folks are really happy with smoothwall. I just use some hacked up
iptables scripts myself.
Jason
The place where you made your stand never mattered,
only that you were there... and still on your feet
Dean Collins
There are several dual-WAN routers with load balancing and failover,
including the Xincom Twin-WAN series that I have tested OK with SIP (as
NAT): http://www.xincom.com/twinwan.php . Their other products probably
work, too.
Keep in mind that load balancing on these devices assigns
Hi all,
When I listen to my voicemail or when the caller reviews his message the volume
is too high in the playback. It is allmost distorted. The other sounds of the
PBX sounds great, its just the voicemail that is being plaied. Any thoughts?
many thanks,
Christian
Incorrect :) IAX2 most definitely does support regcontext.
Also, I think what he means is the phone specific information must be
exactly the same from system to system or the failover won't be as
seamless as you expect. A lot of phones support some sort of SRV
records, so in the event of a
One more thing i would like to point out is that softphones like sjphone use some freeware stun server to detect nat on network (as a client ) . Asterisk(asclient) cannot use external stun server to detect nat type automatically so i think thats why it isnt able to make calls while softphone works
In article [EMAIL PROTECTED],
Vicky [EMAIL PROTECTED] wrote:
Yes asterisk can do that . If you mena for call records then see
http://www.voip-info.org/wiki-Asterisk+cdr+mysql
Also see
http://www.voip-info.org/wiki/view/Asterisk+cmd+MYSQL
The OP was asking about MS SQL (i.e. Microsoft SQL
Did not know how to make up a subject line for this.
I have a dial plan that allows a caller can try my cell phone. And that's
fine. If the call cannot be made, it sends caller back to voice menu.
However, I'd like a way for the caller to elect to go back to the voice menu,
if they end up
Try this subject line if you will.
On 11/14/06, joe a. [EMAIL PROTECTED] wrote:
Did not know how to make up a subject line for this.I have a dial plan that allows a caller can try my cell phone.And that's fine.If the call cannot be made, it sends caller back to voice menu.
However, I'd like a way
Hello,
Here is our setup:
asterisk-A --LAN-- nat-router --Internet-- asterisk-B
A and B have appropriate friend entries in their sip.conf with a
qualify=yes.
The router forwards anything on sip,iax and sip/rtp ports to A.
The problem: SIP/A remains UNREACHABLE for SIP/B, however A sees B. No
On Fri, 10 Nov 2006, Todd- Asterisk wrote:
I'm preparing to deploy a small number of Grandstream BT101's and
GXP2000's to a remote location (which I won't have access to). I'd
like to have them pull a config file from my server - I'm almost
there...
The phones are looking for the config
On Tue, Nov 14, 2006 at 10:39:00AM -0500, Steve Sobol wrote:
On Tue, 14 Nov 2006, Doug Crompton wrote:
Ok so ONLY 900 numbers are pay.
Next question 18XX numbers. are they all toll free? Is there any
space in 8xx that is used otherwise?
Whoa. No, only 800, 888, 877, 866 and
Thanks for the reply matthew, basically I've been looking at going to a
dmz model for a while as currently everything runs through a single
sbs2003 server and when it's churning drives doing something sometimes
audio errors occur.
Cheers,
Dean
-Original Message-
From: Matthew
We've had great results with Astrocom powerlink for load balancing
outbound wan connections.
==
Jeronimo Romero
EUS Networks
Email: [EMAIL PROTECTED]
Cell: 917-332-7238
Office: 212-624-5943
Web: www.euscorp.com
==
-Original Message-
From:
You need to modify app_queue.c to hold off on bridging until the receiving party has accepted the call. If the receiving party rejects (hangup, digit other than '1', timeout, etc), leave or put the calling party back in at close to the same
There are many none free area codes in the 8xx space. Just google area
codes.
.Brian
Doug Crompton wrote:
Ok so ONLY 900 numbers are pay.
Next question 18XX numbers. are they all toll free? Is there any
space in 8xx that is used otherwise?
Doug
On Tue, 14 Nov 2006,
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Tuesday, November 14, 2006 9:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Load balance Asterisk servers?
Incorrect :) IAX2 most definitely does support
Thanks- they did respond. I got a new template, but was asked to not
share it for now - it'll be on their website in a few days pending
committee approval
thanks
Todd
On Nov 14, 2006, at 12:50 PM, Gordon Henderson wrote:
On Fri, 10 Nov 2006, Todd- Asterisk wrote:
I'm preparing
By the time you purchase PCI cards for you extensions (FSO ports)you would
be better off purchasing SIP phones like Grandstream GXP 2000 this will
give you a fully featured PBX IP phone for about the same cost or less
than FSO ports. Asterisk will have no problem running 25 or more SIP
phones
I see the Grandstream website now has the new config templates posted
with all the happy P commands...
http://grandstream.com/y-configurationtool.htm
Todd
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To
On 11/14/06, Aaron Daniel [EMAIL PROTECTED] wrote:
Incorrect :) IAX2 most definitely does support regcontext.
Also, I think what he means is the phone specific information must be
exactly the same from system to system or the failover won't be as
seamless as you expect. A lot of phones support
It appears to be up there now. From the header:
## Configuration template for GXP-2000 firmware version 1.1.1.14
Look here for details on the North American Numbering Plan:
http://www.nanpa.com/reports/reports_npa.html
The report named Non-Geographic NPAs In Service lists the Toll Free and
Premium assignments.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Am Dienstag, den 14.11.2006, 12:28 -0500 schrieb joe a.:
Did not know how to make up a subject line for this.
I have a dial plan that allows a caller can try my cell phone. And that's
fine. If the call cannot be made, it sends caller back to voice menu.
However, I'd like a way for the
Hi James,
We're running SIP 1.6.6.0036 on the 3.1.3.0131 BootROM.
Did you come up with any reason/fix for this?
CP
On Nov 13, 2006, at 11:00 PM, James Andrewartha wrote:
Anthony Rodgers wrote:
Greetings,
Has anyone noticed that attempting to place a call from the Placed
Calls list on a
Have you tried setting up a hint for a ZAP channel?
exten = foo,hint,ZAP/bar
Then make a directory entry for foo in your Polycom directory for foo -
just as you would if the hint was for a SIP channel.
CP
On Nov 14, 2006, at 4:26 AM, Robert Jenkins wrote:
Hi,
I've recently got some
We use only IP connections to our asterisk boxes. Given this our
origination/termination providers
usually send/receive traffic to/from our network on a single IP or
limited number of IPs.
In a DUNDi Asterisk Cluster, would each of the boxes need to be able
to connect to our
On Tue, Nov 14, 2006 at 03:41:08PM +0100, Stephen Wingfield wrote:
LONDON, UK (14th November 2006) - Bicom Systems announced today it has
released its first freeware software to the Asterisk Community, OutCall.
To avoid any confusion: free here means a limited license for one copy
per user.
At 02:08 AM 11/14/2006, you wrote:
From memory... the strike was around $95, the relay board between $25
and $50, and the power supply was only a few dollars, so you could do
it all for under $200.
Do be careful with these. I was installing one and discovered that I
could open it by the
List,
Im a Cisco certified Network guy with little telecom experience (BRI/PRI
at the time) so please forgive my terminology. I am showing interest
after the Network World SHSU October 4 article. We have 3 offices
(Hub-Spoke T1 Frame relay to the remote offices(Data voice on separate
T)). Each
On Tue, 2006-11-14 at 02:10 -0800, Steve Langstaff wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Peter Howard
Sent: 14 November 2006 00:38
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Problem found Re:
James R. Stevens wrote:
Have we enough info to ask:
1) 1 server or several?
1 for each location, if the T1 goes down, they still will have a phone
system. Each Asterisk system can trunk to each other via IAX.
2) Channel bank or not?
If you want to supply dial tone
Tzafrir Cohen wrote:
On Tue, Nov 14, 2006 at 03:41:08PM +0100, Stephen Wingfield wrote:
LONDON, UK (14th November 2006) - Bicom Systems announced today it
has released its first freeware software to the Asterisk
Community, OutCall.
Tzarif,
Thanks for your contribution in clarification of
Gustavo Felisberto wrote:
I just received two TDM400P cards, but I'm having problems with them.
The full info is at:
http://pastebin.com/824079
Extra, I'm using :
libpri-1.2.3
zaptel-1.2.10
On a x86 stable Gentoo box.
Kernel: 2.6.17
gcc-4.1.1, glibc-2.4-r4
Is that an hardware
I am seeing the following in my log file (standard trixbox install).
One seems to be complaining about an error in the dialplan but it
won't tell me what file or what line. The other (maybe related) is
complaining about a channel lock.
How to do go about trying to figure out what the problem is
On Tue, 2006-11-14 at 12:00 -0700, David Thomas wrote:
On 11/14/06, Aaron Daniel [EMAIL PROTECTED] wrote:
Incorrect :) IAX2 most definitely does support regcontext.
Also, I think what he means is the phone specific information must be
exactly the same from system to system or the failover
Hello,
Anselm Martin Hoffmeister Try adding username=200 which fixed things for
me. Alternatively, Try using a username that does NOT begin with a digit -
I saw a flaky softphone some time ago that would screw completely with a
numeric username.
Dovid B The error you are getting is that
On Tue, 2006-11-14 at 13:09 -0700, David Thomas wrote:
We use only IP connections to our asterisk boxes. Given this our
origination/termination providers
usually send/receive traffic to/from our network on a single IP or
limited number of IPs.
In a DUNDi Asterisk Cluster, would each of the
Hi thereI am trying to change the rtp packet size of my Cisco 7940 from 10ms to 20ms. Does anyone know how I can do this.Codec: ULAWSIP firmware: 8.2Bootload ID: PC03A300Thanks.Naija Man
___
--Bandwidth and Colocation provided by Easynews.com --
Roger Gulbranson wrote:
On Tue, 2006-11-14 at 17:07 -0500, Roger Gulbranson wrote:
On Tue, 2006-11-14 at 21:42 +, Gustavo Felisberto wrote:
Gustavo Felisberto wrote:
I just received two TDM400P cards, but I'm having problems with them.
The full info is at:
http://pastebin.com/824079
Hi,I have 2 simple asterisk servers linked over IAX. I want to know if it will be possible to transfer my voice mail messages in mailbox of SIP_PHONE1 on Atserisk1 to mailbox of SIP_PHONE2 on Asterisk2.SIP_PHONE1 --Asterisk1 ---IAX2-- Asterisk2
Hi,
I've not yet used a TDM400, only a 2400.
Silly question first, are you connecting the power cable? I don't know what
happens if you leave it off.
I found that if I have the zaptel asterisk services enabled, the
card/drivers do not initialise correctly.
To get my system working, I ended up
When you say DUAL T1 card from Digiim. Are you thinking One T for voice
coming in the other T going to the remote office(s)?
Why Dual T1 card?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle
Sent: Tuesday, November 14, 2006 2:59 PM
To:
James R. Stevens wrote:
When you say DUAL T1 card from Digiim. Are you thinking One T for voice
coming in the other T going to the remote office(s)?
Why Dual T1 card?
1 for the direct link to the PSTN and the other to the Adit
Doug
-- Ben Franklin quote: Those who would give up
Thanks, will do more research on that part. By the way, Im trying to do IVR where caller enter the pin the retrieve some information out of the MS SQL. I am wondering, what is the constraints or how to go about it. As per said MS SQL is about CDR. Now like i want to match and retrieve data out of
Hi!I am getting inbound caller ID fine bout not out.I am in Sweden and suing Rixtelcom /POrt80 as provider.anyone knowing what is wrong?Also is anyone knowing about Swedish voices to trixbox/Asterisk? I have male now and am looking fro female voices.
RegardsMattias-- Mattias
Hi!I am getting inbound caller ID fine bout not out.I am in Sweden and suing Rixtelcom /POrt80 as provider.anyone knowing what is wrong?Also is anyone knowing about Swedish voices to trixbox/Asterisk? I have male now and am looking fro female voices.
Sorry if I have missed a previous answer on the
Am Mittwoch, den 15.11.2006, 01:06 +0100 schrieb Mattias Andersson:
Hi!
I am getting inbound caller ID fine bout not out.
I am in Sweden and suing Rixtelcom /POrt80 as provider.
anyone knowing what is wrong?
Assuming that is a SIP provider, it is not your job to set the callerid
but the
Now I have answer to my own question, i.e. No, they don't. Grandstream Phones unfortunately are not very advanced in remote provisioning system, and they don't have one single file serving the whole installation, instead every phone needs its own configuration file. Then this file has to be
Now I know that this has something to do with the NAT for sure. But why and how, I don't know. I changed DHCP settings on my router to expire lease time in 1 hour. Then I could see on Asterisk CLI phones getting registered every hour. Then I changed it back to 7 days. But still some phones kept on
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