Displaying 20 results from an estimated 600 matches similar to: "tdm400p / outbound zap prob"
2005 Jul 11
1
ASterisk@home + Broadvoice = Almost working installation...
Hello Guys,
I'm somewhat of a newbie and am desperately seeking for some help...
I've managed to get asterisk up and running on my server, and signed up with a
broadvoice account...
I'm having no problem dialing and communicating between extensions, but whenever
anyone tries to call my broadvoice account, they are greeted by no ring or
anything, but rather simply a direct to
2010 Jun 16
0
H323 Trunk Problem calling from Asterisk to Avaya PBX
On Wed, Jun 16, 2010 at 4:35 PM, Shina Owolabi <shinacalypse at gmail.com>wrote:
> Hi!
> I've installed Asterisk 1.4.32 with freepbx-2.6.0 in an attempt to provide
> a conference bridge for an existing Avaya PBX. I have no control over the
> Avaya system, but I am able to speak with the admin in charge when I need
> stuff done. I am running all this in a VirtualBox
2006 Feb 09
0
re: voipjet -- Workaround if needed
Same thing here. I had this problem awhile ago and made this
workaround.
Going to another trunk does not work because they are answering and not
sending a error code. If you are using AAH code then this waits 10
seconds on your Voip then times out and goes to PSTN. You can modify
for your needs
The pertinent line is 14 in macro-dialout-trunk
I am going to clean it up and repost under my
2006 Feb 10
0
Half Solved - Fail over to Pri on VoIP connection failure
I want to say thanks to everyone for the help so far. I figured out a
way to modify some AAH code that worked for me (well sort of). The line
I modified is s,14 in macro-dialout-trunk. Then I just added a variable
and passed it from 9_outside.
I just have one last problem. This waits for an answer not ringing. So
if the called party has a long ring to voice mail the call is dropped
and goes
2005 Mar 08
2
GotoIf with Authenticate
Quick question...Im authenticate all exten except this one(2006). If I call from ext 2006 I still have to authenticate. If I call form any other ext I have to authenticate. Any suggestions?
Thanks
extex => s,1,GotoIf($[${EXTEN} = "2006"]?3)
exten => s,2,Authenticate(731)
exten => s,3,GotoIf($[foo${ECID${CALLERIDNUM}} = foo]?4)
exten =>
2005 Aug 02
12
WHat does it take
How many times do you ask for help here before getting a respone? Every
single thing I do No matter what I get busy extensions. I am willing to pay
someone to help here. Anybody got a clue?
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2005 Aug 24
0
SIP trunk rollover problem
Hello,
I've got an Asterisk system with 3 SIP trunks configured. Each SIP
trunk is actually a 4 port Mediatrix PSTN gateway. The current outbound
call routing (via AMP 1.10.007a) uses the 3 trunks in descending order,
all set with max channels to 4. Unfortunately, when the first trunk
reports a "480 Service Unavailable" (all ports in use), Asterisk reports
congestion without
2005 Sep 14
2
PRI to PRI passthrough with DID intact
I currently have: Telco-PRI ---- Panasonic DBS576 PBX ---- E&M wink
T1 ---- Asterisk.
I have configured the Panasonic to forward my Asterisk DIDs to the Asterisk
extensions over the T1.
I do not get DID nor CID on the Asterisk, so I want to use PRI between the
PBXs.
I do not want to pay for another PRI card for the Panasonic. (T1 and PRI are
different cards)
I see this as my least
2005 Sep 16
0
Unable to create ZAP channel - All circuits are busy
Hello,
I have *@Home 1.5 installed and all is working fine for incoming calls and
sometimes outgoing calls. Installed in the box is a digium TDM04B (4xFXO
Ports)
setup as ZAP1 to ZAP4. I have incoming calls coming in on lines 1-4 in that
order and outgoing calls prefering ZAP4 then ZAP3 then ZAP2.
When i try to dial out to the PSTN from a SIP phone it sometimes works
(normally after a reboot)
2005 Jul 05
1
Newbie question reg. Asterisk and Channel Access Bank I and TE110p
Hi,
I have some problem to get this setup working. I have a CAC Channel
Banl I, with FXO and an Asterisk box ( I am using Asterisk@Home 1.2)
and I have a TE110p installed in this box.
What I want to do is, just to be able to dial one of those lines that
already are connected to the channel bank, and transfer that call
through TE110p and Asterisk to a user agent somewhere through
Internet.
2005 Feb 15
0
oh323 question
I'm trying to connect an asterisk server via oh323 to a Lucent iMerge.
I patched the code due so that Lucent can handle asterisk's ver4 h323
http://www.voip-info.org/wiki-Asterisk+Lucent+iMerge+Configuration
I can now successfully dial in to our company over multiple lines.
The issue is when I dial out
The first outgoing call connects to an outside user A
The second call drops the first
2006 Mar 17
0
Call transfer problems, SOLVED
Hi All, in regards to my previous queries about call transfers not working from inside, several days of searching turned up this posting:
I got this to work by editing the line
exten => s,14,Dial(${OUT_${ARG1}}/${OUTNUM})
to say
exten => s,14,Dial(${OUT_${ARG1}}/${OUTNUM},,Tt)
in extensions.conf
seems like many people have had this issue in the past, I guess it's AMP related, as
2005 Jan 26
1
Inbound analog Telco line not answered
I have an X100P clone hocked up to an analog line of my PRI. I can use it
to dial out.
but when I call the extension it answers and says "GOODBY"
I have a Livevoip DID which successfuly rings to ext 202
I am using asterisk@home and through the AMP inface the line should ring to
ext 202
Below are Asterisk Messages, Extensions.conf and Extensions_additional.conf
Extensions.conf
2005 May 26
0
capi dial in/out configuration
Hi all,
I've recentrly starting to play around with *, when all I wanted is to
configure an fritz ISDN card with A@H.
Currently I'm stuck at the phase of what do I do with capi after
everything is installed.
I'm trying to understand how to setup incoming and outgoing calls at A@H
since I'm getting a bit lost with the default dial plan.
It seems that * answers but disconnect
2015 Aug 18
2
Asterisk 13 chan_sip trunk appending @string to dialled number
Hello,
So, I found this line under macro-dialout-trunk, in extensions_additional.conf (FreePBX, so it controls the conf files mostly);
exten => s,n,Dial(${OUT_${DIAL_TRUNK}}/${OUTNUM}${OUT_${DIAL_TRUNK}_SUFFIX},${TRUNK_RING_TIMER},${DIAL_TRUNK_OPTIONS})
If I grep for OUT_3_SUFFIX in all files in /etc/asterisk I get nothing..
Here's a paste of a few things out of the two files that I
2005 Mar 22
1
Call file misbehaviour
Greetings *`s,
I am manually creating call files and dropping them into
/var/spool/asterisk/outgoing to be picked up by *.
Presently, when I use local/internal parameters using SIP it works..ie I
make an internal call from device to device.
However, when I try dial an outside number which I have set up in a
custom conf file, it bombs out with the following message :
2012 Sep 24
0
stop on rows where !is.na(mydata$ti_all)
Dear R experts,
I got help to build a loop but there is a bug inside it that causes
one part of the mechanism to fail.
It should grow once, but if keep growing on rows where $ti_all is not NA.
Here is a wall of code that very crudely demonstrates the problem,
there is a couple of dim() outputs at the end where you can see how it
the second time around keeps adds (2) rows, but this does not
2013 Sep 09
0
DNS Administration Strangeness
I'm running Samba 4.0.9 as a domain controller in a Windows 2008 R2 functional level single domain forest with a Windows 2012 domain controller. I am using Bind 9.9.3-P2 as the DNS backend with dlz_bind9_9.so driver.
When I add my Samba DC to the DNS Manager snap-in, my main domain.com zone has a red X and I cannot expand it. The _msdcs.domain.com zone seems to be ok. Also, one of my reverse
2006 Dec 18
3
Inform callers on recorded/monitored number.
Hi,
How could I possibly inform incoming callers that the number they'd dialed is monitored and recorded.
I wanted that when a call-in or call-out is made, a playback will be played to inform caller & callee that thier line is monitored prior to start conversation.
Thanks.
Angel
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2005 Sep 04
0
FW: OH323 with Asterisk@home - seems incomplete
Thank you (for spamming) - it was the clue I needed to push this through.
Sorry it took me a while (and a google :-) ) to realize you'd addressed my
initial query - basically, my loss.
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jason Becker
Sent: Wednesday, August 24, 2005 00:36
To: Asterisk Users