Displaying 20 results from an estimated 300 matches similar to: "use AT320 international call"
2005 Oct 07
3
TDM02B card difficulties
Hi all,
I just installed an TDM02B. My system is a dell pc with
linux 2.6.12-1.1456_FC4
asterisk-1.2.0-beta1
zaptel-1.2.0-beta1
libpri-1.2.0-beta1
in /etc/zaptel.conf I have (all others are default):
fxsks=3-4 <--- I saw light in the ports
channels=1-2 <--- change it to 3-4 has same result
but...
[root@nmsd0 asterisk]# /etc/rc.d/init.d/zaptel
2006 Jan 18
0
Atcom AT320: SIP or IAX?
Hi,
I have a few of Atcom AT320 phones. I can choose if install SIP or IAX2
firmware.
Any suggestion?
Mimmus
2006 Oct 26
10
ECHO Cancellation in SIP Calls
Hi,
i am from Germany, so excuse my School English.
I use Asteriks 1.2.12.1, zaptel 1.2.9 and mISDN Rc25 - since my update
of Asterisk 2 wooks ago, Echos accure in my SIP Calls.
I use SNOM 360, sometimes there is no echo (for example if i call myself
via SIP->Asterisk->SIPProvider->TELEKOM->ISDN)
but if i call other people there occures Echo many times. The Routing is
always the
2008 Feb 24
2
DUNDi with two servers
Hi,
I'm having difficulties with using DUNDi between two servers. If it were
three I think I could control looping by limiting TTL, but with two I'm not
sure how to prevent a loop causing bad things to happen. I've tried ttl=1
but things still blow up.
The DUNDi configurations are pretty simple and work just fine in both
directions as long as only one of them is using the switch
2009 Nov 16
1
can't call through voip provider
Hello.
Sorry to repost this message but, I don't have the original message in my inbox nor in my sent box.
Well, last week I posted a problem I am having trying to use an asterisk server use a voip provider and a pstn. Pstn works fine but, I cant even connect to my provider's server. I don't know what I'm doing wrong.
I tried using a soft phone and I'm able to register and
2005 Jul 16
3
Sip registration question
Hi everyone,
I have a number of SIP registrations going fine, but am trying to get a new
provider going, and they have no sample Asterisk SIP config. They have been
helpful, but keep falling back to the way they "think" packets should be
flowing,
and I've been trying to figure out how the Asterisk config should look like
to get the SIP packet to look correct.
Now, they say that
2008 Oct 09
2
Asterisk 1.6.0 CDR billsec and duration not working from h extension
Can someone tell me what I am doing wrong? Why doesn't CDR(duration)
or CDR(billsec) return the correct values?
cdr.conf
endbeforehexten=yes
extensions.conf
[macro-Dial]
; ${ARG1} - Dial String
exten => s,1,Dial(${ARG1},,M(post-dial))
exten => h,1,NoOp(Call was hung up - ${CDR(duration)} seconds long,
billed for ${CDR(billsec)} seconds)
The log shows:
-- Executing [h
2006 Nov 09
2
Alcatel trunk with asterisk problem on dialing digit-by-digit
Hi guys,
I have an Alcatel 4200 with a ISDN board pluged in the asterisk server with
TE110P.
Input calls
VOIP Proider ---> Asterisk ---> Alcatel
Output Calls
VOIP Proider <--- Asterisk <--- Alcatel
In alcatel phones, users should dial 2 for take a line tone and can dial. At
this point start my problems:
1. When users dial 2 on phone (alcatel) they don't received a dial tone,
2009 May 20
3
...is circuit busy message
Hi,
I am attempting to make about ten calls simultaneously and intermittently
get 'SIP/voipprovider is circuit-busy' followed by 'everyone is
busy/congested at this time"
I am not sure if this is related to my bandwidth to my voip provider, a
configuration issue or something else.
Has anyone seen this before and have any suggestions. Thanks in advance.
--------------
2005 Feb 14
1
[Bug 2187] rsync large file getting verification failed using -z
https://bugzilla.samba.org/show_bug.cgi?id=2187
wayned@samba.org changed:
What |Removed |Added
----------------------------------------------------------------------------
Status|ASSIGNED |RESOLVED
Resolution| |FIXED
------- Additional Comments From wayned@samba.org 2005-02-14 01:35
2013 Jul 20
1
rejected because extension not found in context 'introutingB'
Dear All,
I am trying to recieve call from inbound proxy then route to internal peer
(localhost) and then route to outgoing sip proxy but it failing with
subject error. Can any one please correct me what i am doing wrong in below
config.
SIP.conf
[Inbound]
type=peer
context=introuting
host=184.107.XXX.XXX
disallow=all
allow=all
[astinside]
type=peer
context=introutingB
host=localhost
2006 Oct 30
3
Grandstream ATA 286 tdm400 and Asterisk 1.2-13
Hi people,
I would like to read your suggestions as to where the issue might be.
ATA286=Grandstream Budgetone ATA 286 in SIP mode. One Lan Port and one FXS port.
TDM04B= 4 FXO signal fxls
There is a 8FXO-to-SIP unit in this scenario that works perfectly so i
will not make mention of it.
PSTN----VOIPprovider---Internet---ATA286------tdm04b---Asterisk1.2.-13
Asterisk is being used as a meetme
2003 Jun 05
2
sendmail: clientmqueue directory storing massive messages, not being cleared
I just got done sending 42,000 e-mail messages that were rotting away in
/var/spool/clientmqueue. I figured out the command to do so was
"sendmail -L sm-msp-queue -Ac -q30m".
I never even knew that directory existed. I thought everything queued was
kept in /var/spool/mqueue (which, I might add is being taken care of on a
regular basis.. just seems the problem lies in the
2006 Nov 06
1
Asterisk servers being greedy and not letting go of the media path. (using IAX2 channels)
Evening everyone (obviously depends on when you're readin this, but hey).
I'm trying to set up a multi * server situation, and am falling over at the
second server, and after a day of google etc, have come up against somewhat
of a brick wall.
I can make calls each way between the two servers no problem, and can
include the required extension at the remote * server as part of my main
2005 Aug 01
4
IAX Devices Recommendation
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Does anyone have any recommendations on an IAX Desktop Telephone or ATA
Device. I currently have 2 of the SIPURA-841's on my local network and
now I am wanting to try an IAX Device at my remote office since I think
that it would be easier to configure through various routers than a SIP
Device. I just started to look at the Digium IAXy Single FXS
2006 Jan 20
1
IAX and call transfer
Hi,
I flashed my ATCom AT320 phone (PA1888S based) with IAX firmware instead of
SIP but now call transfer doesn't work neither using phone buttons nor using
Asterisk features.
I heard that it can be a real problem.
Any help?
Mimmus
2007 Sep 25
1
Backuping VoIP provider with PRI
Hi list,
My Asterisk config for outgoing calls is the following:
exten => s,1,Dial(SIP/${MACRO_EXTEN}@voipprovider,60,g)
exten => s,n,GotoIf($[\"${ANSWEREDTIME}\" = \"\"]?pri:hang)
exten => s,n(pri),NoOp(Problems with voip provider trying PRI)
exten => s,n,Dial(Zap/g2/${MACRO_EXTEN},60,g)
exten => s,n(hang),HangUp
in most cases it works well but, if my
2006 Oct 17
1
how to activate recording (automon)
Hi all,
If I activate recording for an extension everything is OK.
but If I activate call recording on demand i am non able to start recording
In principle I should have to press *1, as indictaed in features.conf
(I am using almost last asterisk code, updated 2 days ago from svn, version
SVN-branch-1.2-r39379M )
Actually it produce no effect at all
I am using FreePBX interface, and I saw
2009 Sep 20
1
A in ACL of sip show peers.
Hello.
>> ubuntu*CLI> sip show peers
>> Name/username Host Dyn Nat ACL Port Status
>> voipprovider xxx.xxx.xxx.xxx A 5060 Unmonitored
I've ben trying to connect an asterisk server to a voip provider, and
I'm currently wondering what the 'A' in the ACL field of the 'sip show
peers' command might
2008 Mar 21
1
----www.cdsportal.net---- wholesale voipprovider --starting at 1.1 cent per min
Piling on...
InterNIC says the domain was created almost a week ago, and expires in a year. The registrar is GoDaddy. The owner of the site is located in the Dominican Republic:
C/1ra #15
Costa Criolla, Km9 Carr. Sanchez
Santo Domingo, New York 00000
Dominican Republic
Registered through: GoDaddy.com, Inc. (http://www.godaddy.com)
Domain Name: CDSPORTAL.NET
Created on: 14-Mar-08
Expires on: