I finally have the solution, so thought I would post back to the list for
completeness.
It ended up being a series of changes. First, on the gateway, set Disconnect
on Broken Connection to false. Then, for the Polycom phones, set
voIpProt.SIP.allowTransferOnProceeding to 1 in the sip.cfg.
I just put in a Audiocodes Mediant 1000, which seems to be working well except
for one annoyance. I am using Polycom 501's and 601',s and if I do a
supervised transfer of a PSTN call where I complete the transfer before the 3rd
party has answered, the PSTN party hears dead air until the call
Hi listers,
I am looking for people who have used Power over Ethernet switches,
primarily in conjunction with Polycom IP 501's. I've been looking at the
Linksys SRW224P, since I've had good luck with the SRW224 in our office.
However, Nortel, Cisco, Adtran, etc. all have an offering, all
Thanks for the link. The ultimate solution was to change from fxs_ls to
fxs_ks. Now it works great!
Thanks,
James
Dr. Michael J. Chudobiak wrote:
[EMAIL PROTECTED] wrote:
If so, is there a way to detect the hangup?
Check out
Hello list,
I have recently deployed Asterisk as the phone system for my office. So
far, everything has been going really well, except for one little thorn in my
side. I have a set of 6 analog lines that are connected to a TE411P via a
Rhino FXO Channel bank. If I call the analog
I currently have Asterisk deployed in my office with a TE411P. On the first
port of this card is a T1 from the telco setup for D4 AMI. Unfortunately, I'm
not receiving caller ID on inbound calls from this line. The caller ID
information is arriving in the form *ANI*DNIS*. In zapata.conf, I
It appears that I'm getting **DNIS*, as my setup in extensions.conf uses the
DNIS, and that all works fine. Is there some debug logging I can turn on to
see if that is in fact what I'm getting?
Thanks,
James
From: Matt Florell [EMAIL PROTECTED]
Date: 2006/03/17 Fri AM 11:08:14 EST
To:
Okay, so as it turns out, this is *sort* of working. If I call from Extension
A to Extension B, and B places A on hold, A hears the hold music. However, if
A places B on hold, B does not hear the hold music. Is there a way to specify
the hold music if the originating party places a call on
I have a requirement to play different hold messages depending upon the
extension that originated the call. I noticed a musicclass setting in
sip.conf, but it appears this is global. I tried setting this on all of my
individual extensions, but it didn't have any affect. Is there a way to
So, I'm still having this problem with outbound calls not working when using a
channel bank. I've purchased a Rhino FXO channel bank from VoIPSupply.com to
make sure it wasn't an equipment problem. I am using a Digium TE411P card, and
have simplified it down to just 1 port plugged into the
Thanks for the reply. I have tried adding anywhere between 1 and 6 w's to the
dial string, but still no luck. I hooked up and listened on the line when the
call went out, and never heard any DTMF's. I'm sure this must be something
simple, I just can't seem to figure out for the life of me
The output from the CLI when I put in an inbound call is the following:
-- Starting simple switch on 'Zap/25-1'
-- Executing GotoIf(Zap/25-1, 1?from-pstn-reghours|s|1:) in new stack
-- Goto (from-pstn-reghours,s,1)
-- Executing GotoIf(Zap/25-1, 0?from-pstn-reghours-nofax|s|1:2) in
No, it's an Access Bank II SNMP.
Thanks,
James
C F wrote:
Is this an Adit 600?
On 2/1/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
The output from the CLI when I put in an inbound call is the following:
-- Starting simple switch on 'Zap/25-1'
-- Executing GotoIf(Zap/25-1,
I appreciate all of your help. I'm starting to think this is a setup/hardware
issue with the channel bank myself. I bought the FXO card off of EBay, so who
knows what kind of shape it's in.
Does anyone else out here on the forums know how to configure a CAC Access Bank
II SNMP channel bank
I am experimenting with an asterisk setup in my office. The last bit I have to
test is working with analog lines. I have TE411p digium card, with an ISDN
line plugged into the first, a channel bank plugged into the second port, and
the last two ports empty. I have the following setup in my
I'm curious if anyone has this working with [EMAIL PROTECTED] I just installed
the 2.0 Beta, which loads up * v1.2.0. I edited my features.conf to put in the
following:
[featuremap]
automon = *1
I place a call to my cell phone, and from my polycom put in *1, but nothing
happens. If I use
I am doing some experimenting with Asterisk 1.0.9 and Polycom IP501's. I have
the extensions setup, and everything is working well up to this point. Now, I
want to setup my system so that a user at an extension can start a recording on
demand. I have tried various Google searches, but am
I am getting ready to purchase my first Digium card to start experimenting with
Asterisk. Before I make my purchase, I wanted to make sure I'm not going to
have issues with these cards (need to see what the specs are on my box, 5V or
3.3V PCI ). I will be using Asterisk @ Home, so will be
Hi Matt,
Thanks for the response. I wasn't aware you could upgrade the card. In
that case, I think I'll make my boss happy by saving money and going with the
regular card first.
I'm basically checking out how well Asterisk works before putting into
production for our office. I figure
I've read several places that say you cannot have more than two 4 port Digium
T1/E1 cards, as it would overload the PCI bus. Is that true? If not, what is
the maximum number of cards people are putting into boxes? If two cards is the
limit, am I right in understanding that the preferred way
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