Title: Need help connecting Alcatel 4400 PBX to Asterisk
Hi there
I have a TE110P card fitted in my linux box running :
Linux version 2.6.9-5.ELsmp ([EMAIL PROTECTED]) (gcc version 3.4.3 20041212 (Red Hat 3.4.3-9.EL4)) #1 SMP Wed Jan 5 19:30:39 EST 2005
I followed the installation steps on
Dovid B wrote:
Is there any advantage of getting a T1 card with a channel bank over
2-3 FXO cards ?
Thanks.
channel bank is more friendly to faxes and modems (v90 can work too)
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asterisk-us
> Hi Jamie -
Hi Noah,
>> Has anyone here reprogrammed their Polycom features keys using
>> sip/ipmid.cfg?
>>
>> If so I would be really grateful if someone could send me an example
> Here's the "keys" line that I use for one of my clients:
> key.IP_500.37.function.prim="DialpadPound"
> key.
Does anyone see anything wrong here?
CLI> realtime load sipusers name 1000
Column
Name Column Value
id 1
name 1000
callerid "Don" <1000>
Hi all,How can i originate a call from someone outside my sip-network (for example my PSTN home number) to one of my SIP number?I can originate a call from my SIP-network using this parameters in Originate call command :Channel = SIP/0041435215301Context = defaultExten = 00982166501553Priority = 1C
On Wed, 2006-11-01 at 03:23 -0800, Ehsan Khosrowshahi wrote:
> Hi all,
>
> How can i originate a call from someone outside my sip-network (for
> example my PSTN home number) to one of my SIP number?
>
> I can originate a call from my SIP-network using this parameters in
> Originate call command :
If I understood your question correctly, you just need to reverse everything.
Channel = OUTGOING TRUNK i.e. ZAP/00982166501553
Context = default
Exten = internal extension that points to -> 0041435215301
Priority = 1
CallerID = 0041435215301
This will first initiate the call to the number 004143
Hello Users...I'm Strucked in Call parking...I'm Using the Asterisk-1.1.11 version in My FC5 box,In That there is feature.confI'm Using SIP channel By using Asterisk + OpenSER
[general]parkext => 9006 ; What extension to dial to parkparkpos => 9007-9009 ; What ex
Hi
Does anyone know how I can check if a callerID is more than
2 digits.
I am setting up my phones so that if the callerID is 3
digits the phones ring one way if it is more than 3 digits it rings another
i.e. internal calls and external calls.
exten => ,1,GotoIf($["${CALLERIDNU
In article <[EMAIL PROTECTED]>,
Ehsan Khosrowshahi <[EMAIL PROTECTED]> wrote:
> How can i originate a call from someone outside my sip-network (for example
> my PSTN home
> number) to one of my SIP number?
>
> I can originate a call from my SIP-network using this parameters in Originate
> call c
Hi,
Is there any possibility to have md5 encoded passwords in the IAX users
database? I notice the "secret" AND/OR "md5secret" columns always have to
contain the password in plain text even when you set the "auth" column value
to md5?!?
Am I missing out something? Any ideas on how to correct this
The following will:
exten => s,1,GotoIf($[${LEN(${CALLERID(num)})}=2]?50)
On 11/1/06, Scott Pinhorne <[EMAIL PROTECTED]> wrote:
Hi
Does anyone know how I can check if a callerID is more than 2 digits.
I am setting up my phones so that if the callerID is 3 digits the phones
ring one way i
raviprakash sunkara wrote:
In Extension.conf .. I'm confused to give the Dial planning..
You don't need to do anything in the dial plan for parking. Just
transfer the call to your parking extension and Asterisk will take it
from there.
Doug
_
Ken
If these are older comdials then they are just analog phones with
"extra signaling". The extra signaling could be on the main twisted
pair (likely) or on the next twisted pair as data (9600 baud modem)
like some of the nortels do. Always remember that it would cost the
companies a ton to ma
On 11:53, Wed 01 Nov 06, Scott Pinhorne wrote:
> Hi
>
>
>
> Does anyone know how I can check if a callerID is more than 2 digits.
>
> I am setting up my phones so that if the callerID is 3 digits the phones
> ring one way if it is more than 3 digits it rings another i.e. internal
> calls and e
On Wed, 2006-11-01 at 11:53 +, Scott Pinhorne wrote:
> Hi
>
>
>
> Does anyone know how I can check if a callerID is more than 2 digits.
>
> I am setting up my phones so that if the callerID is 3 digits the
> phones ring one way if it is more than 3 digits it rings another i.e.
> internal c
Hi,
In Asterisk 1.2.7, my AEL code looks like this:
macro callForwardHunt(numargs,numlist,typelist,ttr)
{
for(x=1;${x}<${numargs}+1;x=${x}+1)
{
CUT(number=numlist,-,${x});
CUT(type=typelist,-,${x});
NoOp(${number});
Hi Wendy,
I got this info from digium developers, that caller id name
transfer/display (asterisk/iphone <-> pbx/clasic phone)) using
ISDN/Q.SIG should work,
so, do you have possibility to confirm this, if it realy working in
practice (with siemens hipath idealy)? thanks
PJ
Origin
Scott Pinhorne wrote:
Hi
Does anyone know how I can check if a callerID is more than 2 digits.
I am setting up my phones so that if the callerID is 3 digits the phones
ring one way if it is more than 3 digits it rings another i.e. internal
calls and external calls.
exten => ,1,G
That worked great
Many Thanks
-Original message-
From: "C F" [EMAIL PROTECTED]
Date: Wed, 1 Nov 2006 06:57:28 -0600
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] ${CALLERIDNUM}
> The following will:
>
Hi,
When playing a wav-format (( low compression),(wav49-format?)) file with
Windows Media Player, it plays the file and then sometimes bombs out with an
error about how the file is corrupt or unsupported. If you listen to the
file in wavepad you will hear the whole file, in Media Player the la
Thanks for help me with this issue. I've this scenario, a PANASONIC KX domain
and an ASTERISK domain, each one with their own pool of extensions, incoming
calls are recived by the PANASONIC KX as a gateway from PSTN to the office.
Once a call is recived by the PANASONIC,it bridge the call to ASTERI
Dear
How can I customize a2billing to have two groups
One have service to play its balance and the second group
do not play the balance.
Regards
*
No employee or agent is authorized to conclude any binding agreement on behalf of Xp
I love the Snom phones as well. The function keys are great and easy to use.On 10/31/06, mitcheloc <[EMAIL PROTECTED]
> wrote:My vote is definitely for Snom, I've worked with Cisco phones foryears, but the Snom is much better integrated, and the feature buttons
can be retooled for any environment,
Dear
How can I customize a2billing to have two groups
One have service to play its balance and the second
group do not play the balance.
Regards
*
No employee or agent is authorized to conclude any binding agreement on behalf of
Hello,
All the biggest gateways manufacturers do that.
Search for Aliwei, Audiocodes, Patton, etc...
Best Regards,
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Noc Phibee
Envoyé : lundi 30 octobre 2006 20:51
À : Asterisk U
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Johann Steinwendtner schrieb:
> I 'm trying T.38 faxig with spandsp using rxfax/txfax as fax terminal.
> As another endpoint I 'm using Grandstream HT 486 ATA and a fax machine.
> Has anybody success with the HT486 as T.38 terminal ?
I've successfull
http://www.aztech.com/prod_iptelephony_ip150.htmlaztech rawks... the lcd has backlighting and methinks is snom inside
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http:/
I was reading Octobers online edition of Wireless Asia and
came across a company called CDyne (www.cdyne.com)
They build a number of web services applications but among
other things they have an application which you can fill out your details on a web
page will some time in the future
This is incorrect. The data is still packetized and passed through IP which
provides the same echo cancellation and distortion issues as a call that
passed through an FXO/FXS card.
Ejay Hire
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ed
Sent: Wedne
That looks like a rebranded Snom 300 to
me.
Cory Andrews
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rosli
SukriSent: Wednesday, November 01, 2006 10:16 AMTo:
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [SPAM
HEADER] - Re: [asterisk-users] Snom o
Hello.
In extensions.conf; in the context that is dialed by your internal
extensions, add this line.
include=>parkedcalls
This will include the extensions created by the extensions module, and
create your extensions 9006-9009.
Good luck,
Ejay Hire
From: [E
it is, the navigation button is exactly the same, also notice the
extreamly short handset cord
On 11/1/06, Cory Andrews <[EMAIL PROTECTED]> wrote:
That looks like a rebranded Snom 300 to me.
Cory Andrews
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
Thanks everyone for the input. After pricing everything we need out,
it's not worth trying to get our old system to work, so I've pitched
ditching everything and starting over. I'm very excited and hoping
they'll go for it.
Regardless, I'm going to throw a box together for my house, we have no
h
Erick Perez wrote:
Trying to compile asterisk-addons 1.2.5 on Centos 4.4 produces this:
Note: MySQL libraries are installed and the structure is as follows:
/usr/src/astsources/asterisk-1.2.13
/usr/src/astsources/asterisk-addons-1.2.5
in /usr/src/astsources/asterisk-addons-1.2.5 I do:
make clea
Hi all,
I have to buy some IP phones. Previously I have used Grandstream GXP-2000, Budgetone 101 and Linksys SPA-841. I always had problems with sound quality with all of them, and I was always of the opinion that it were the phones which were not good. In GXP-2000 deployment of about 50 phones,
I am trying to send commands to Asterisk manager via a telnet session. I am
able to lo in and receive event logs from AMI, but when I try to issue commands
I get an invalid/unknown command error. Here are some of the commands I am
trying to send.
Asterisk Call Manager/1.0
Action: login
Userna
Ken - take a look at using IAX protocol to route calls between your
Asterisk boxes.
Cory Andrews
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ken
Williams
Sent: Wednesday, November 01, 2006 10:58 AM
To: Asterisk Users Mailing List - Non-Commercial Di
Zeeshan Zakaria wrote:
Hi all,
I have to buy some IP phones. Previously I have used Grandstream
GXP-2000, Budgetone 101 and Linksys SPA-841. I always had problems with
sound quality with all of them, and I was always of the opinion that it
were the phones which were not good. In GXP-2000 dep
Hi,
I want to upgrade my older switch from 1.0.9 to 1.2.6 asterisk
version. What do I need to be aware of? I AM aware 1.2.6 is not the
newest version, but anything above .6, at this time, seems to have
stability issues (I've tried them on multiple machines)
_
Ok sorry for not being specific. I am having a problem when people
outside call in to my number which terminates at VoicePluse then The
send IAX to me and I do not get any tones. People press buttons but it
just goes to the next dialplan fall through. It happens 60-70% of the time.
extentions
I'd recommend any of the following, which are all in your
price range
Snom 300
Polycom IP430
Polycom IP501
Aastra 9112i
Linksys SPA-922
Grandstream GXP-2000
Cory Andrews
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Zeeshan
ZakariaSent: Wednesday, November 01, 2006
The quality issues you describe may not be the fault of the individual
phones! The quality of the PSTN connection, and the hardware through
which it's connected can play as big a part in this scenario. The
quality of the internal network with 50 IP Phones could also be part of
the problem.
At 03:53 AM 11/1/2006, you wrote:
exten => ,1,GotoIf($["${CALLERIDNUM}" = ""]?5)
This will tell it to jump to 5 if callerID if but how do i tell
it do jump based on length of callerID?
exten => ,1,GotoIf($[1${CALLERIDNUM} <= 1999]?5)
Ira
Writing this as a user of VoIP and not a reseller, (meaning off the record), we really love the Snom phones here as well, I wish the Snom 300's had a bit more functionality (like the Grandstream), and the Snom 320's and 360's were a little less confusing with their buttons (aka too many buttons on
I tend to stay away from the Grandstream phones for business use because they simply break to easily. I would suggest using Snom phones like the Snom 300 for around $99.2 Asterisk boxes in different locations? Sure, you can do that and its quite easily.
On 11/1/06, Ken Williams <[EMAIL PROTECTED]
Sorry, but I failed to mention that I am running Asterisk BE B 1-1
I am trying to send commands to Asterisk manager via a telnet session. I am
able to lo in and receive event logs from AMI, but when I try to issue commands
I get an invalid/unknown command error. Here are some of the commands
Jason,There are a couple things we can try to fix your problem.Your firmware shouldn't be an issue, but latest I've got now is: MP118_SIP_F4.80A.034.004.cmpLet's try some quick things first though:In your web interface, go to advanced config - channel settings / voice settingsThere are some options
Zeeshan Zakaria wrote:
Hi all,
I have to buy some IP phones. Previously I have used Grandstream GXP-2000,
Budgetone 101 and Linksys SPA-841. I always had problems with sound quality
with all of them, and I was always of the opinion that it were the phones
which were not good. In GXP-2000 deploym
Matt wrote:
Hi,
I want to upgrade my older switch from 1.0.9 to 1.2.6 asterisk
version. What do I need to be aware of? I AM aware 1.2.6 is not the
newest version, but anything above .6, at this time, seems to have
stability issues (I've tried them on multiple machines)
/path/to/src/asterisk/
snom 300 :">
CS
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Kristian
Kielhofner
Gesendet: Mittwoch, 1. November 2006 12:01
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: Re: [asterisk-users] Which IP phones have best voi
This e-mail, including attachments, contains privileged and confidential
information intended only for the use of the addressee(s) name above. If you
are not the intended recipient of this e-mail, or an authorized employee or
agent responsible for delivering it to the intended recipient, please
The problem is voicepulse, but they refuse to accept responsibility. From What phone are you pressing the DTMF?On 11/1/06, Jason Walker <
[EMAIL PROTECTED]> wrote:Ok sorry for not being specific. I am having a problem when people
outside call in to my number which terminates at VoicePluse then The
I have the Budgetone 101 and GXP2000 and thought the sound quality
was excellent. Even over the internet... I agree with Joe that
something else may be the factor...
Todd
Zeeshan Zakaria wrote:
Hi all,
I have to buy some IP phones. Previously I have used Grandstream
GXP-2000, Budget
I had the same problem trying to use an iaxy for an overhead paging system.
SIP has an option to set DTMF to inline, but iax does not.
There was nothing I could do to get the iaxy to play audible DTMF tones.
I had to use a SIP ATA for my paging system with the inline DTMF option.
Note: The DTMF
Ken,
Also stay away from Swissvoice phones
I have found several ways to do the second thing.
http://www.voip-info.org/wiki/view/Asterisk+Connect+2+servers
It works great.
Jason
Tom Vile wrote:
I tend to stay away from the Grandstream phones for
business use because they simply break to ea
Does anyone have a management tool for Polycom phones? For instance something to
view software and boot versions of all the phones? I am looking for a product to remotely mange
all phones in the environment without having to connect to each phones web config individually.
Thanks
Hello,
Whenever I put in a password/Shared Secret in my 7960 and try and get
it to register with asterisk on OS X setup, the phone fails to register.
Oct 31 20:03:46 NOTICE[989]: chan_sip.c:11045
handle_request_register: Registration from ''
failed for '67.121.71.120' - Wrong password
W
I am testing 1.4 branch on OSX (10.4.8) and although it's running and
passing calls ok, I am still not able to connect using asterisk -r.
When I do open a CLI using asterisk -r, it appears to start up
normally, but then is non responsive to commands (exit works though?).
I am currently runni
Dovid B wrote:
> Read the book Asterisk: The future of Telephony
> http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
>
> It will teach you a lot.
The trouble with this (I have it) is that it's dated.
I do wish we had a more structured and maintained documentation project.
voip-info
> "BT" == Brad Templeton <[EMAIL PROTECTED]> writes:
BT> The correct behaviour, as I see it is:
BT> a) Native bridge when connecting two external channels --
BT> everybody is on the real internet b) Native bridge when connecting
BT> two internal channels -- everybody is on the 192.168.* n
On 2006-11-01 08:28:28 -0800, Jason Walker <[EMAIL PROTECTED]> said:
Ok sorry for not being specific. I am having a problem when people
outside call in to my number which terminates at VoicePluse then The
send IAX to me and I do not get any tones. People press buttons but it
just goes to the
On Wed, Nov 01, 2006 at 09:08:43AM -0600, Ejay Hire wrote:
> This is incorrect. The data is still packetized and passed through IP which
> provides the same echo cancellation and distortion issues as a call that
> passed through an FXO/FXS card.
The issue here is an "implementation bug" of Zaptel
Martin Joseph wrote:
Good news!
I did an SVN update to my 1.4 branch again today, and 1.4-r46154 seems
to have resolved the asterisk hogging the whole CPU issue.
I still can't use the regular console though (asterisk -r) as that is
unresponsive.
Using asterisk -c to start it , works and
Thanks for the suggestions.. there is no such document in 1.2.6 in docs.
On 11/1/06, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote:
Matt wrote:
> Hi,
> I want to upgrade my older switch from 1.0.9 to 1.2.6 asterisk
> version. What do I need to be aware of? I AM aware 1.2.6 is not the
> new
I wanted to buy service from SellVoip, however I have NEVER been able to reach anyone via phone, and I never really got email responses from them either.I have recommened a few times ISPhone (
www.isphone.net) however they don't have nationwide DIDs.On 11/1/06, Brad Templeton <[EMAIL PROTECTED]
> w
Hello list partners
you know about a softphone made in java attachable
in a web page?
GNU!
Thaks in advance!
Visita www.tutopia.com
y comienza a navegar más rápido en Internet. Tutopia es Internet
para todos.
___
--Bandwidth and Colocation
Hello,
The problem was wrong contexts defined like Marco said, and is solved.
Now, i have another problem...of course :)
On incoming calls, i only can receive calls if i define a line like
the following, in extensions.conf:
exten => _.,n,Dial(SIP/500,30,tr) (all incoming calls are redirected
to
Has anyone noticed that Asterisk seems to always set the remote-party-id in a
SIP invite to be the same value as the From: field? In most cases that isn't a
problem. However, in the case of an attended transfer it IS a problem. The
remote-party-id should be the party who initially called and the
Hi Andrew, I can highly recommend using the Granstream GXP 2000.
Upgrade the firmware to ver. 1.1.1.14 and you won't have any problems.
The 4 line buttons are not actual lines they are calls queued up on an
extension so you can have as many incoming lines as you want. The first
call comes in on lin
I strongly recommend you upgarde to the latest firmware for the GXP 2000.
I have been using them for almost a year now and while the early firmware
was poor they are now very stable and working fine (from 1.1.1.9) onwards.
Henry L.Coleman CEO
*VoIP-PBX* 1-866-415-5355
Toronto Ontario
Canada
>
Khaled wrote:
Dear
How can I customize a2billing to have two groups
One have service to play its balance and the second group do not play
the balance.
This is not the a2billing support forum.
Jeremy McNamara
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Any potential testers eager to build imap storage support using proper
Debian packages:
Resonably up-to-date packages of c-client (uw-imap) 2004/2006 are by now
only availble from experimental:
http://packages.debian.org/experimental/mail/uw-imapd
On my test Etch system I simply downloaded the s
Martin Joseph wrote:
I am testing 1.4 branch on OSX (10.4.8) and although it's running and
passing calls ok, I am still not able to connect using asterisk -r.
When I do open a CLI using asterisk -r, it appears to start up
normally, but then is non responsive to commands (exit works though?).
On Wed, Nov 01, 2006 at 11:15:23AM -0700, Stephen Bosch wrote:
> Dovid B wrote:
> > Read the book Asterisk: The future of Telephony
> > http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
> >
> > It will teach you a lot.
>
> The trouble with this (I have it) is that it's dated.
>
> I
On Wed, Nov 01, 2006 at 08:29:32PM +0200, Tzafrir Cohen wrote:
> On Wed, Nov 01, 2006 at 09:08:43AM -0600, Ejay Hire wrote:
> > This is incorrect. The data is still packetized and passed through IP which
> > provides the same echo cancellation and distortion issues as a call that
> > passed throug
Neider, Clint wrote:
Does anyone have a management tool for Polycom phones? For instance
something to view software and boot versions of all the phones? I am
looking for a product to remotely mange all phones in the environment
without having to connect to each phones web config individually.
On Wed, Nov 01, 2006 at 06:16:25PM +0100, Benny Amorsen wrote:
> > "BT" == Brad Templeton <[EMAIL PROTECTED]> writes:
>
> BT> The correct behaviour, as I see it is:
>
> BT> a) Native bridge when connecting two external channels --
> BT> everybody is on the real internet b) Native bridge w
Neider, Clint wrote:
Does anyone have a management tool for Polycom phones? For instance
something to view software and boot versions of all the phones? I am
looking for a product to remotely mange all phones in the environment
without having to connect to each phones web config individually.
I think I will agree with folks here, it must be something else on the network, not the phones themselves. I am not going to replace all of the phones, its too expensive, but for trial, want to try something better. PoE is also important to me at this point. I am thinking of trying Linksys 942. I w
Hi Brad,I can confirm the service quality of unlimitel.Have you look at www.les.net they provide both US and Canada DID. I heard good feedback about them
On 10/31/06, Brad Templeton <
[EMAIL PROTECTED]> wrote:I've been losing patience with my current provider, a small company
called Sellvoip. Thei
Jason Walker wrote:
Ok sorry for not being specific. I am having a problem when people
outside call in to my number which terminates at VoicePluse then The
send IAX to me and I do not get any tones. People press buttons but it
just goes to the next dialplan fall through. It happens 60-70% of
Has anyone noticed that Asterisk seems to always set the remote-party-id in a
SIP invite to be the same value as the From: field? In most cases that isn't a
problem. However, in the case of an attended transfer it IS a problem. The
remote-party-id should be the party who initially called and the
Sorry, the file is located here:
[EMAIL PROTECTED] ~]# ls -l asterisk-1.2.6/UPGRADE.txt
-rw-r--r-- 1 1000 1000 8739 Dec 1 2005 asterisk-1.2.6/UPGRADE.txt
Matt wrote:
Thanks for the suggestions.. there is no such document in 1.2.6 in docs.
On 11/1/06, Eric ManxPower Wieling <[EMAIL PROTECTE
Sergio R. D'Ippolito wrote:
> Hi all, i have an * version: Asterisk SVN-branch-1.2-r45691, I need to
> register a linksys 922 phone thru internet and when I make sip debug
> command i see this debug information:
> */SIP/2.0 401 Unauthorized/*
>
> /Via: SIP/2.0/UDP x.x.x.x:1025;branch=z9hG4bK-43
Douglas Garstang wrote:
Has anyone noticed that Asterisk seems to always set the remote-party-id in a
SIP invite to be the same value as the From: field? In most cases that isn't a
problem. However, in the case of an attended transfer it IS a problem. The
remote-party-id should be the party wh
On 2006-11-01 10:42:12 -0800, Joshua Colp <[EMAIL PROTECTED]> said:
Martin Joseph wrote:
Good news!
I did an SVN update to my 1.4 branch again today, and 1.4-r46154 seems
to have resolved the asterisk hogging the whole CPU issue.
I still can't use the regular console though (asterisk -r)
On 2006-11-01 09:09:26 -0800, Martin Joseph <[EMAIL PROTECTED]> said:
I am testing 1.4 branch on OSX (10.4.8) and although it's running and
passing calls ok, I am still not able to connect using asterisk -r.
When I do open a CLI using asterisk -r, it appears to start up
normally, but then is
Hello list partners
you know about a softphone made in java attachable
in a web page?
GNU!
Thaks in
advance!
Visita www.tutopia.com
y comienza a navegar más rápido en Internet.Tutopia es Internet
para todos.
___
--Bandwidth and Colocation
On Wed, 1 Nov 2006, Pedro Silva wrote:
> Hello,
>
> The problem was wrong contexts defined like Marco said, and is solved.
> Now, i have another problem...of course :)
>
> On incoming calls, i only can receive calls if i define a line like
> the following, in extensions.conf:
> exten => _.,n,Dial
All the phones already have the latest firmware. They keep updating themselves automatically.
In my setup of Grandstream phones, all the computers of the network go through the phones, i.e. I am using the builtin phones as swithces. They all have 2 ethernet ports. Does this has to do anything wit
I have a config where I define a single peer and have possibly hundreds of register commands for that single peer.I'm not clear if I can do the register part via Asterisk Realtime (right now I updated a file and force a reload which re-registers all the users defined in the register directives).
I
Hi,
I really need some assistance in installing and configuring this card. I have already physically installed it into the computer which is running Mandriva 2006. I have compiled and installed asterisk 1.2.13 along with zaptel-1.2.10 and libpri-1.2.4. However I do not know what the next ste
On 21:47, Wed 01 Nov 06, Tzafrir Cohen wrote:
> Any potential testers eager to build imap storage support using proper
> Debian packages:
>
> Resonably up-to-date packages of c-client (uw-imap) 2004/2006 are by now
> only availble from experimental:
>
> http://packages.debian.org/experimental/mai
I am testing toll free and US DID inbound
as well as A-Z outbound with les.net at the moment. Both the quality and
support are quite good. Ping time to Vancouver is around 80ms.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Marcel Eric Loiselle
Sent: Wednesday
Can anyone confirm that SMS() works correctly under asterisk 1.2.12?
It used to work around version 1.2.7, but a few people have reported
that 1.2.8 and 1.2.9 were a bit dodgy and that all their problems went
away when they used app_sms from 1.2.7 in the later versions of
asterisk.
When a fixed l
Salve Salve Galera.
Tenho a seguinte situacao:
Uma placa MD3200 ligada em uma linha telefonica comum(PTSN) e
funcionando "belezinha"...
Tenho configurado um URA, onde ele atende a ligacao que chegou no
canal e solicita o numero do ramal de destino da ligação:
Acontece que ao discar o ramal de
Seems to me that you have a routing problem, asterisk should not know
how to send packets to an outside IP using the NATed network. Make
sure that the internal (NAT) interface doesn't have a gateway to it.
On 10/31/06, Brad Templeton <[EMAIL PROTECTED]> wrote:
I've read a lot of the description
Sorry for my previous post I misunderstood the problem.
You should set canreinvite=no to all sip peers that connect from outside.
On 10/31/06, C F <[EMAIL PROTECTED]> wrote:
Seems to me that you have a routing problem, asterisk should not know
how to send packets to an outside IP using the NATed
Hi,
I'm new to asterisk. I want asterisk to connect a external line with
an internal line: the PC dials a number and connects this call to a
internal telephone (telephone switchboard, based on ISDN, 4 analogue
telephones) of my office.
Can somebody here give me keyword how to search (e.g. wi
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