For some reason something that seems like it should be simple is leaving me a
bit perplexed. I am receiving incoming CallerID ANI on my PRI, but on my VoIP
phones the display just shows asterisk when calls come in. I am receiving
the calls with DNIS and have the DNIS digits setup as
Hello
I am curious as to what hardware folks are using
successfully from HP or DELL. I will likely be running just a quad span T1
card with the system.
I appreciate your input.
Thanks,
Dave
___
--Bandwidth and Colocation
Can someone give me some direction on automon filenames? I
would like them to be the dialed number if possible. I saw a patch available
for changing this but havent quite figured out how to use it.
Can someone point me in the right direction?
Thanks,
Dave
-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike Fedyk
Sent: Sunday, January 08, 2006 10:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Non-PRI T1
David Sampson wrote:
Hello -
I have a non-PRI T1
[...]
How do I take incoming
Hello
I have a non-PRI T1 setup and have been making outgoing
calls for several months no problem. I have Zapata.conf setup for fxs_ks on
these channels. How do I take incoming calls on these same channels? Do I
need to change the signaling?
Any help is appreciated.
Thank you,
] On Behalf Of Dave Cotton
Sent: Friday, September 16, 2005 3:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] RxFax/TxFax - Compile Problem
On Thu, 2005-09-15 at 16:10 -0400, David Sampson wrote:
I’ve reduced my problem down to this:
[EMAIL
] On Behalf Of BJ Weschke
Sent: Wednesday, September 14,
2005 2:43 PM
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
RxFax/TxFax - Compile Problem
What version of
spandsp are you attempting to compile in to the 1.0.9 tree?
On 9/14/05, David Sampson
app_rxfax.c:264: error: structure has no
member named `verbose'
app_rxfax.c:325: warning: passing arg 1 of
`fax_release' from incompatible pointer type
make: *** [app_rxfax.so] Error 1
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David Sampson
Sent: Thursday
Anyone know how to fix this?
gcc -shared -Xlinker -x -o app_rxfax.so app_rxfax.c -lspandsp -ltiff
In file included from app_rxfax.c:14:
/usr/include/asterisk/lock.h: In function `ast_mutex_init':
/usr/include/asterisk/lock.h:302: error:
`PTHREAD_MUTEX_RECURSIVE' undeclared (first use
Hello
My single line extension users (connected via channel banks)
need to be able to hang up faster. If they just flash the hook it doesnt
disconnect right away. Any ideas on how to resolve this?
Thanks,
Dave
___
Asterisk-Users
Hello
I have a Premisys Slimline Channel Bank connected to a
Digium TE110P. I am not able to call the FXS extensions or get dialtone on
them. The channel bank is connected via a T1 crossover to the cable and lights
show green. I really need to get this functioning by end of day. If
I have a Mitel SX2000 with no voicemail. Im
wondering if it would be possible to use Asterisk to meet the need. This is a
hotel property with mostly analog extensions.
Dave
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Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
I have many old channel banks around that I would like to
use to generate analog extensions. Will most channel banks work with Asterisk?
Dave
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Asterisk-Users@lists.digium.com
I have 2 4-port Digium FXS cards in my system. I would like
to play a different recording based on which trunk rings. Any pointers?
Thanks
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How do I setup my T1 card as an EM tie line? Any
special configuration?
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To UNSUBSCRIBE or update options visit:
! Good luck!
-Matt
-Original Message-
From: David
Sampson [mailto:[EMAIL PROTECTED]
Sent: Monday, May 23, 2005 3:16 PM
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] EM
Tie Line
How do I setup my T1 card as an EM tie line? Any
special
Hello
I am looking at connecting 7 10 locations together
using Asterisk and possibly some VoIP gateway appliances. I need to
insure best voice quality as these trunks will be used primarily for customer
calls. I am considering implementing a full T1 frame relay circuit to
each
Does anyone else have info regarding the port speed matching the CIR?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
Wieling aka ManxPower
Sent: Thursday, May 19, 2005 11:55 AM
To: Andrew Latham; Asterisk Users Mailing List - Non-Commercial
I run a call center and would like agents to be able to
initiate call recording from a web page.
Can anyone help?
Thanks,
Dave
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Asterisk-Users@lists.digium.com
Hello
.
I currently have a generic modem card in my Asterisk system
and will be upgrading to a Digium card next week. What I need to be able to do
is immediate answer and play a greeting, then hang-up. Currently the system
answer after two rings how can I minimize this?
Thanks,
Dave
Hello
My PSTN line answers after two rings and plays a greeting
but I would like it to answer sooner. How can I make it answer immediately?
Thanks,
Dave
David J. Sampson
Information Technology Manager
InnSeason Resorts
212 Mid-Tech
Drive
West
Hello
I have an IP500 (my first). The phone is up and running and
I am able to make outgoing calls but I cant get the phone to register
and take incoming calls.
This is what my sip.conf looks like:
[8503]
type=user
username=dave
callerid=Dave Sampson 8503
secret=default
Hello
I have just setup my first Asterisk box and Im having
a great time.
I am having a little trouble getting incoming calls to
answer. This is what I see on the console:
Apr 25 17:01:07 NOTICE[3514]: chan_zap.c:5374 ss_thread: Got
event 2 (Ring/Answered)...
-- Executing
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