Displaying 20 results from an estimated 6000 matches similar to: "ASterisk@home + Broadvoice = Almost working installation..."
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 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 Aug 10
0
tdm400p / outbound zap prob
I'm having trouble getting outbound calls going with aah 1.3 and a tdm400p
w/ 4 FXO. Incoming calls work fine, outbound I get this:
-- Executing SetVar("SIP/231-af2b", "OUTNUM=6643955") in new stack
-- Executing Cut("SIP/231-af2b", "custom=OUT_1|:|1") in new stack
-- Executing GotoIf("SIP/231-af2b", "0?19") in new stack
2004 Dec 21
5
AMP - Fax Detections
Does anyone know of any obscur reference for detecting an incoming fax.
I currently have AMP running and everything else is working great.
Installed the spandsp patches and software... using the default AMP
extensions.conf, I start sending a fax, I hear it pick up and transfer
to voicemail after 20s.
Fax is set for system... Here is the detail from the extensions.conf
[global]
FAX_RX = system
2005 Jan 18
0
AMP and Asterisk PSTN extension config
Hi,
I have configured an Asterisk server with TDM01P (1FXO) for testing purpose. The interface I'm using is AMP. I want to configure my extension so that when I dial from my mobile phone to the asterisk line, I want it to transfer the call to any extension, say 3042 and after a particular number of rings, transfer the call to voice mail so that I can record my message.
My Zaptel.conf is as
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 Feb 28
2
Fax Failing
Hello All,
I am trying to set up faxing using Asterisk@home 0.6. I have followed
the instructions to the best of my knowledge. When a fax comes in, the
system seems to detect OK but does ot manage to make the fax to pdf to
email leap. Here is what I saw in the CLI when I tested. Any help
would be appreciated.
Thanks!
Wiley
-- Starting simple switch on 'Zap/2-1'
-- Executing
2006 Jun 21
2
FW: syntax error
(Try again from the proper email address)
--Rob
-----Original Message-----
From: Rob Thomas
Sent: Thursday, 22 June 2006 12:22 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] syntax error
That's freePBX or AMP code that we've since fixed - The replacement line is
exten =>
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
2005 Feb 11
2
transferring a IAX call into a conference
When I make a call out on the Faktortel number I am then able to
transfer to call to my asterisk meetme room of 801 by hitting 'transfer'
then '801' then 'send' on my grandstream phone.
This connects my faktortel trunk (and who ever is on the other end) to
my conference room I can then make another call using my local pstn
service and set up a 3 way (or whatever number
2006 Feb 28
1
FW: Re: Delay on Phone ringing
Skipped content of type multipart/alternative-------------- next part --------------
asterisk1*CLI> soft hangup Zap/1-1
Requested Hangup on channel 'Zap/1-1'
== Spawn extension (macro-exten-vm, s-BUSY, 2) exited non-zero on 'Zap/1-1' in macro 'exten-vm'
== Spawn extension (ext-local, 220, 1) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
--
2006 Feb 06
12
Cisco 2620 as PRI gateway
I just inherited a Cisco 2621 with a VWIC-1MFT-T1 card in it. Can I make
this thing into MGCP gateway or even a SIP gateway for asterisk? Seems like
it should bee useful for something!
I'm perfectly happy to do my homework, but also don't feel thee need to
reinvent the wheel! So, links with relevant info would be appreciated. If
there is a config for a 2621 being used as a gateway
2005 Jun 18
2
Unable to make outbound calls
Hi All,
I am a new bee to *. I just installed Asterisk@home on
FC3. I hv a FXO card. I hv configured two extensions
one x-lite and other iaxComm. I configured * using
AMP. The following setup works
- x-lite (x 200) to iaxComm (x 201)
- PSTN to x-lite
- PSTN to iaxComm
Voice mail, weather etc work fine.
When i try to make an external call i am getting
message "All routes are busy". In
2005 Mar 22
2
asterisk@home print incoming fax
*@home has this for it's incoming fax macro
--- start snip ---
[ext-fax]
exten => in_fax,1,GotoIf($[${FAX_RX} = system]?2:analog_fax,1)
exten => in_fax,2,Macro(faxreceive)
exten => in_fax,3,system(tiff2ps -2eaz -w 8.5 -h 11 ${FAXFILE} | ps2pdf
${FAXFILE}.pdf)
exten => in_fax,4,system(mime-construct --to ${EMAILADDR} --subject "Fax
from ${CALLERIDNUM}
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 28
0
Trying to cut out the paper work...
Hello everyone,
Ok. I am at a bit of a loss.... and would like someone to point me in
the right direction...(btw www.google.co.za did not give me ANY solutions).
The issue at hand is simple, I get asterisk (1.0.9) to answer the
incoming call with no problems... it does the fax detection thing with
app "Answer" and well it goes to the perfectly right context and sets
the varibles
2005 Aug 21
0
Problem with auto-attendant config, I think..
I never heard anything on the AMP list, so figured maybe someone here might
be able to help me sort this one out..
I was making some updates to my attendant config, which is really very
basic, and now incoming call processing stopped. Not sure exactly what the
heck happened, but figured maybe someone could help me with a clue as to what
broke. Now incoming calls are not being answered at
2011 Mar 28
1
DTMF input while waiting in queue...
Hey all!
I'm trying to figure out how to have a queue accept an inbound caller's key
press to action on. At first I'm just trying to implement a "Press 1 to
leave a voice mail" announced and at any time in the queue, the user can
press 1 and go to the queue's voicemail. Later I'd like to have it accept
"Press 1 if this is an x issue, press 2 if this a y
2006 Jan 27
5
External IAX2 phone defined as internal behaving as from PSTN
Have asterisk@home 1.2.1
The server is on an internal network eg 10.10.10.10
It is NAT'd 1:1 via Checkpoint firewall to external public IP eg
50.50.50.50
The remote IAX2 phone (ATCOM320) is configured to call 50.50.50.50 on
extension 1055.
Outbound calls to 1055 work perfectly.
Inbound calls from 1055 get picked up as if it were an external call
(see below) and goes straight to the ring
2005 Sep 14
1
TE110P - Asterisk@Home Install Problems
I am having problems sending and receiving calls over the T1. They never seem to connect - outbound keeps ringing, inbound gets busy. The T1 looks ok - no errors on the line. Any ideas on what is wrong? I have tried a variety of fxsks and fxoks configurations without avail. This is a single asterisk@home system with a single T1 card. Robbed Bit T1 ami, d4.
------------------inbound call