I have not found this to be so. As an end user, I have had excellent support
from Digium on TDM400p. They have been more responsive the several times I had
to call. Even cross shipping replacement cards (CC, required, of course).
Cannot fault their support at all.
joe a.
On 2/16/2008 at
Trying to setup SIP to register with a VOIP provider. I am behind a firewall
(IPCOP) with NAT.
Getting this, in CLI with SIP debug on.
Retransmitting #2 (no NAT) to aa.bbb.ccc.ddd:5060:
REGISTER sip:voip-xxx.com SIP/2.0
Via: SIP/2.0/UDP 192.168.0.xxx:5060;branch=z9hG4bK727a6144;rport
From:
-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joe Acquisto
Sent: Friday, December 14, 2007 2:44 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] SIP fails to register
Trying to setup SIP to register with a VOIP provider. I am behind a
firewall (IPCOP
On 12/11/2007 at 7:22 AM, Joe Acquisto wrote:
I have a working system with two fxo and two fxs channels. I recenlty got an
IAX2 account I would like to use also.
While I have gotten the IAX2 channel to register, it remains non
functional, as the incoming calls, go nowhere
On 12/12/2007 at 8:35 AM, Joe Acquisto wrote:
I have a working system with two fxo and two fxs channels. I recenlty got
an
IAX2 account I would like to use also. . . .
. . . the outgoing calls attempt
to go out over the ZAP channel. I can see this, via the CLI, with debugs
I have a working system with two fxo and two fxs channels. I recenlty got an
IAX2 account I would like to use also.
While I have gotten the IAX2 channel to register, it remains non functional,
as the incoming calls, go nowhere and the outgoing calls attempt to go out over
the ZAP channel. I
On 12/7/2007 at 2:33 PM, Doug [EMAIL PROTECTED] wrote:
At 10:58 12/7/2007, Joe Acquisto wrote:
I have an odd issue, where a polycom 601 stops ringing, or more
properly, maybe, stops *being* rung, when a call comes in. Other
phones/extensions, continue to work fine, they being run
I have an odd issue, where a polycom 601 stops ringing, or more properly,
maybe, stops *being* rung, when a call comes in. Other phones/extensions,
continue to work fine, they being run at the same time.
My dial plan works fine (?) seems it will ring properly, right after a reboot.
It works
Currently running two POTS lines into an asterisk system. Analog and SIP on
premises. Being in the sticks, the POTS service is abysmal for quality,
especially in the rain.
Recently, cable has become available with VOIP phone. The cost savings are
attractive as it can replace several
On 11/27/2007 at 12:26 PM, Ira [EMAIL PROTECTED] wrote:
At 06:01 AM 11/27/2007, you wrote:
I am hesitant to believe that I can simply plug my TDM400P
(2fxo/2fxs) into these (ATA ?) jacks and call it good.
Any insight? Am I better off ignoring their phone offering and
setting myself up with
My Polycom phones are displaying time, off by one hour. Seems they are on the
old DST rules. How do I fix this?
joe a.
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My thanks to all. Problem resolved with the assistance.
joe a.
On 11/1/2007 at 1:43 PM, Joe Acquisto [EMAIL PROTECTED] wrote:
My Polycom phones are displaying time, off by one hour. Seems they are on
the old DST rules. How do I fix this?
joe
On 11/1/2007 at 4:22 PM, Turbo Fredriksson [EMAIL PROTECTED] wrote:
Quoting Joe Acquisto [EMAIL PROTECTED]:
My thanks to all. Problem resolved with the assistance.
Would be nice if you posted HOW it was fixed to... I have this exact
same problem at home, but the work phones displays
I am having a problem detecting incoming FAX. TMD22p (tdm400p 2 fxo, 2fxs)
As I understand it, I must have faxdetect = incoming to enable detection of the
fax tone.
Then, I must have a [fax] context to pickup the line and send it to whatever
extension the FAX device is on.
In my case, I ask
This can be a partial never mind, I guess. I can see via the CLI that the call
is being handled by
some FAX related routines. Just not quite the solution I expected.
joe a.
On 9/29/2007 at 8:56 AM, Joe Acquisto [EMAIL PROTECTED] wrote:
I am having a problem detecting incoming FAX. TMD22p
On 9/29/2007 at 3:27 PM, Lee Howard [EMAIL PROTECTED] wrote:
Joe Acquisto wrote:
As I understand it, I must have faxdetect = incoming to enable detection of
the fax tone.
Then, I must have a [fax] context to pickup the line and send it to whatever
extension the FAX device is on.
It's
On 9/5/2007 at 10:56 AM, Jason Parker [EMAIL PROTECTED] wrote:
Joe Acquisto wrote:
I need to ask, to refresh, is the aux power connector on the TDM400P card
*only* to power the ringer on any
analog phones/devices on the system?
Can I still use this board, to terminate POTS lines
I need to ask, to refresh, is the aux power connector on the TDM400P card
*only* to power the ringer on any
analog phones/devices on the system?
Can I still use this board, to terminate POTS lines and use all SIP Phones?
Due to circumstances, I end up with a 1u server that has no aux power
On 9/5/2007 at 1:06 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote:
Thomas Kenyon wrote:
Joe Acquisto wrote:
I need to ask, to refresh, is the aux power connector on the TDM400P card
*only* to power the ringer on any
analog phones/devices on the system?
Can I still use this board
Is there more than one version of the Cisco 7960?
I see some items advertised as 7960 or 7960G, but searching on 7960 only brings
up 7960G info, or ambiguous stuff.
joe a.
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On 9/2/2007 at 9:32 AM, Joe Acquisto [EMAIL PROTECTED] wrote:
Is there more than one version of the Cisco 7960?
I see some items advertised as 7960 or 7960G, but searching on 7960 only
brings up 7960G info, or ambiguous stuff.
joe a.
A partial never mind, it appears they are two
. . .
I bought a nic card for my APC3000 UPS because I was led to believe it
could turn on an off all of the 8 power points independently but have
never been able to work out how to do this.
Anyone know how to do this?
Cheers,
Dean
I have an APC3000 and don't believe that is
On 9/1/2007 at 7:46 AM, Michiel van Baak [EMAIL PROTECTED] wrote:
On 18:59, Fri 31 Aug 07, Joe Acquisto wrote:
What is involved in getting SIP firmware into a Cisco 7960 with sccp
installed?
Expensive image from Cisco? Plated in unobtanium?
You'll need the firmware and an TFTP server
What is involved in getting SIP firmware into a Cisco 7960 with sccp installed?
Expensive image from Cisco? Plated in unobtanium?
joe a.
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Any great disadvantage to using polycom 300/500/600 vs the 301/501/601?
joe a.
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On 8/28/2007 at 9:30 AM, John Novack [EMAIL PROTECTED]
wrote:
Marc Patino Gómez wrote:
Hi Steve,
Thanks for your advice, I will order a Sangoma card and test the
box. A
part from this, you know any other point to recomend Sangoma cards
versus Digium cards?
Many thanks,
Marc
5
Any experiences putting and supporting IAXmodem on Fonality? They themselves
do not seem interested.
joe a.
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On 8/26/2007 at 10:41 AM, Patrick [EMAIL PROTECTED] wrote:
On Sun, 2007-08-26 at 09:36 -0400, Joe Acquisto wrote:
Any experiences putting and supporting IAXmodem on Fonality? They
themselves do not seem interested.
Do you mean Fonality or Trixbox? If Trixbox you can use the iaxmodem
SRPM
. . .
Personally I recommend SuSE Linux. OpenSuSE without the GUI installed
will do just fine. If you want to buy SLES that's fine, but I really
don't see the value in it.
The value would be live support and access to online updates. Courtesy
(for the price) of Novell.
There are, of
. . .
The value would be live support and access to online updates.
Courtesy (for the price) of Novell.
There are, of course, some differences between OpenSuse and SLES. I've run
Asterisk on SLES 9 and SLES 10 without problems.
Your View/Mileage May Vary.
joe a.
With OpenSuSE you
. . .
Besides naming a flavor and saying It is the best, can someone add a
few statements as to why, which will obviously have to compare the other
flavors.
Thanks,
Steve Totaro
I'd have to review the entire thread to see if anyone actually claimed any
flavor was best, but
can point
Excuse me if I recently posted on this, but I cannot find it, in my, or the
list archives.
Is it possible, when transferring a call, to set the user ID to that of the
outgoing number instead of the incoming number? I believe the answer is
(was) yes.
New twist, does it matter what the
I did post recently, under another subject line.
But would appreciate some response, as some are telling a client that this is
not possible.
joe a.
On 8/20/2007 at 1:57 PM, Joe acquisto [EMAIL PROTECTED] wrote:
Excuse me if I recently posted on this, but I cannot find it, in my
There was a discussion a while back about how to pass Calller ID, when
forwarding, as either the calling number, or the forwarding number.
Had something to do with scams IIRC, but could not find in browsing the
archives.
So, is it in the docs? Starting point or full tilt would be
. . .
The cost of the wire is not that much, without even shopping around or
going through my regular distributor I found this link.
http://www.wesbellwireandcable.com/Bare_Tinned_Copper.html?gclid=CNLa25jj6Y0
CFQ1zHgodBychsQ
1000ft = $169.
Again, it is not that big of a deal if
Telephone conversations that are being recorded, are supposed to beep
periodically, to alert/remind the
recorded person that the conversation is being recorded.
Perhaps that is what you are hearing?
joe a.
On 8/2/2007 at 8:47 AM, Matt [EMAIL PROTECTED] wrote:
Strange issue when I record
. . .
Even if you can find non-original-artist recordings of such music, the
*compositions* are registered with BMI and ASCAP, and you'll need
blanket licenses to play them. (Well, if you only wanted one or two
tracks, you might negotiate specific licenses, but I'm not sure it
would be
APC makes a two line unit. PTEL2. But it's two lines in one jack.
Another - www.ablecom.com is a bit more Pro
Just do a google and take your pick.
joe a.
On 7/14/2007 at 10:17 PM, Todd H [EMAIL PROTECTED] wrote:
I lost one channel on an FXO module on a Sangoma A200 card due to a
On 7/11/2007 at 11:04 AM, Joe acquisto [EMAIL PROTECTED] wrote:
Since the beginning (of my Asterisk life) I have an install that is,
supposedly, set up for call waiting.
Using a TDM400p, with FXO and FXS modules.
On the Analog phones, I can hear the Incoming call (call waiting) tone
Since the beginning (of my Asterisk life) I have an install that is,
supposedly, set up for call waiting.
Using a TDM400p, with FXO and FXS modules.
On the Analog phones, I can hear the Incoming call (call waiting) tone, but the
system does not respond to a hook flash, to place the current
. . .
We let you win, you were terrorists and England's never been good at
fighting terrorists. Now you're having the same problem !!!
One is stuck by the semi-irony. Those who do not learn from History are doomed
to repeat it. However, the current unpleasantness has dis-similar roots.
problem - occasional garbled calls, mostly remote users.
T1 connected to PSTN, SIP over local LAN and internet to remote users. NAT
at local firewall and at remotes. There is no traffic shaping in place, no QoS.
Most are Polycom phones, two Aastra's.
Start with QoS on LAN switches? No
QoS does nothing for you unless you're using MPLS between connections
to a degree. (re-stated...) If you're under the impression that you're
going to magically place some auto-qos of sorts and your traffic will be
magically shaped for high performance, you're semi-mistaken. While it
may shape
. . .
QOS across the internet is pointless and further more doesnt really
exist, I would suggest setting qualify=200 in sip.conf so that asterisk
will not send a call to the remote end if they are more than 200
milliseconds away.
Away, in what sense? Are you referring to packet
. . .
Away, in what sense? Are you referring to packet latency? How does
Asterisk measure this? Ping response?
No, it does NOT measure packet latency. qualify= measures the response
time of the remote device to a SIP OPTIONS packet. If the device is
busy doing something and does not
Contrary to the opinions of Anglo-Philes, we, here in the Colonies,
speak American, not English. In some places, 'Murican.
We get to do that, because, back in the late 1700's . . . we won.
It is only referred to as English out of a sense of compassion.
Oh, so anyway, who was guy Eng you named
This is a follow up to an earlier post.
Looking for a means to individualize incoming FAX, so as to distribute them
to the intended recipient.
While the PBX is based on Asterisk, it is not possible for me to enter the
box to modify things, to any great degree. I thank those who mentioned
. . .
With all due respect, this project should be handed over to whomever has
authorization to administer the Asterisk box. We can tell you how to do it
in Asterisk, but if you can't take our advice, our ability to help you will
be severely limited.
Thanks. Point taken. I'm, unfortunately,
. . .
It looks to me like you have two choices. The first you probably
can't do. That is, get a two port board in the Asterisk system with
the second T1 going into an Eicon Board in a Hylafax system. Then,
you can assign DIDs with whatever web interface you have on this
Asterisk system to
On 6/26/2007 at 3:04 PM, Lacy Moore - Aspendora [EMAIL PROTECTED] wrote:
On 6/26/07, Joe acquisto [EMAIL PROTECTED] wrote:
One idea is to utilize DID, and have Asterisk forward the calls to the
current FAX lines, preserving the DID as Caller ID. I am fairly sure
Asterisk itself can do
I have an existing Hylafax system using a mainpine 4 port board, 4 POTS lines.
Have a recently installed Asterisk system, with a dedicated T1 line. (Well,
it's really a fonality system).
What would I need to do, or where is the reading material, for what I need to
do, to convert the Hylafax
Having had various issues with local vendor (begins with V). am looking to
move to all wireless. Anyone know if current vendor can refuse to port the
current land line numbers to a wireless provider?
From what I've read, the Fed's seem to say no, they cannot refuse, or impede
this.
joe a.
Gergo Csibra [EMAIL PROTECTED] Wrote: 5/4/2007 8:12 AM:
Friday, May 4, 2007, 1:56:09 PM, Joe wrote:
Gergo Csibra [EMAIL PROTECTED] Wrote: 5/4/2007 6:34 AM:
Well this is a digium list, so here will be digium cards
recommendation. But You can use a linksys spa3102, that costs about
half
. . .
man tcpdump indicates that I should be able to use = syntax but it
doesn't
work as expected. Any further advice appreciated.
Cameron
When interested in packets, I usually use ethereal and a 4 port hub, plugging
the ethereal and asterisk boxs into the hub and uplink the hub to
I have dual posted this to the user and biz lists.
Has anyone ever heard of someone running an Asterisk based system, yet
abandoning SugarCRM, and opting to develop their own Visual FoxPro database/CRM?
Please don't dump on me now, this is not my idea, I am just asking for
comments, to see if
Paul [EMAIL PROTECTED] Wrote: 4/30/2007 8:53 AM:
Joe acquisto wrote:
I have dual posted this to the user and biz lists.
Has anyone ever heard of someone running an Asterisk based system, yet
Has abandoning SugarCRM, and opting to develop their own Visual FoxPro
Has database/CRM?
Please
Try this:
# strings -a /usr/sbin/asterisk | grep Digium
I get:
Asterisk 1.2.16, Copyright (C) 1999 - 2005, Digium, Inc. and others.
Asterisk 1.2.16, Copyright (C) 1999 - 2006 Digium, Inc. and others.
but if the version string has been removed from the sources, it's
A very simple question - what version is running?
A CLI - show version does not tell me, only shows info about who compiled, and
when. A google and other nefarious devices have not yielded the secret.
Sure, I could scour the docs and determine what is features do not exist in all
versions,
Gordon Henderson [EMAIL PROTECTED] Wrote: 4/14/2007 7:02 AM:
On Sat, 14 Apr 2007, Joe Acquisto wrote:
A very simple question - what version is running?
A CLI - show version does not tell me, only shows info about who
compiled, and when. A google and other nefarious devices have
Tzafrir Cohen [EMAIL PROTECTED] Wrote: 4/14/2007 7:41 AM:
On Sat, Apr 14, 2007 at 07:15:36AM -0400, Joe Acquisto wrote:
Gordon Henderson [EMAIL PROTECTED] Wrote: 4/14/2007 7:02 AM:
On Sat, 14 Apr 2007, Joe Acquisto wrote:
A very simple question - what version is running?
A CLI - show
Tzafrir Cohen [EMAIL PROTECTED] Wrote: 4/14/2007 9:59 AM:
On Sat, Apr 14, 2007 at 08:58:57AM -0400, Joe Acquisto wrote:
Tzafrir Cohen [EMAIL PROTECTED] Wrote: 4/14/2007 7:41 AM:
Next-best thing would be to sart looking for differences in messages and
such from recent commits
Stephen Bosch [EMAIL PROTECTED] Wrote: 4/9/2007 7:51 PM:
Joe Acquisto wrote:
Stephen Bosch [EMAIL PROTECTED] Wrote: 4/9/2007 7:12 PM:
Have you been able to test this yourself? (Three to four seconds seems
inordinately long. That's as bad as a satellite link.)
No, not tested by me, I only
Eric ManxPower Wieling [EMAIL PROTECTED] Wrote: 4/10/2007 3:53 PM:
Steve Edwards wrote:
On Tue, 10 Apr 2007, Joe Acquisto wrote:
My own perception of delay finds it acceptable. Could be
intermittent, tho, I suppose.
Anytime somebody complains of delay or lag, have them call a cell
? Or is someone quite well versed in SIP traffic who can read
the trace?
joe a.
Joe Acquisto [EMAIL PROTECTED] Wrote: 4/9/2007 1:42 PM:
Hi.
Is there a way to isolate what shows on CLI to just the conversation
with that extension? There appears to be a lot of stuff unrelated to
this extension
Steve Prior [EMAIL PROTECTED] Wrote: 4/6/2007 8:30 PM:
Steve Prior wrote:
I've seen in the wiki that it is possible to use a celldock device to
use a cell phone as a PSTN line to Asterisk, but I haven't seen any
comments as to how well this actually works. I was thinking about
hooking a
Looking for 404 in inbound-video (domain sip3701.ibsonecall.com)
list_route: hop: sip:[EMAIL PROTECTED]:5060;user=phone
Joe Acquisto wrote:
Steve Totaro [EMAIL PROTECTED] Wrote: 4/4/2007 8:44 PM:
Joe Acquisto wrote:
Attempts to do SIP thru firewall (IPCop) are unsuccessful. using
. . .
I have a Dock-n-Talk at home I use to connect my motorola V60i via a
cable so I can't comment on bluetooth. I needed it because for some
reason I can only get good cell reception in my bedroom. It works well
enough. You can certainly tell you are talking over a cell connection
. . .
Can't be worse than my POTS lines. The cable runs here are about 30
years old, and run underground, supposedly, where crossing a
government right of way. This run is ancient, as well.
Supposedly, during wet weather, this becomes a grounding problem.
Certainly the audio quality
the messages correctly.
Joss.
On 4/9/07, Joe Acquisto [EMAIL PROTECTED] wrote:
I never get this far, apparently. While the connection seems to be made,
and calls can be completed (rings, answers) there is no audio. On CLI, I
can see what appears to be call being made and connected. These are x
Stephen Bosch [EMAIL PROTECTED] Wrote on: 4/9/2007 2:16 PM:
Joe Acquisto wrote:
Sometimes it's just a matter of finding a clean pair in the cable. Have
you tried asking Verizon to fix the problem?
Don't get me started. That's how I know so much about the situation.
They seem disinclined
In a system connected to a verizon T1, Digium TE411P (quad T1 echo
cancellation), client is complaining it is too quiet.
The complaint regards calls over the T1, not in house SIP only calls.
Their description indicates they want some earpiece feedback of themselves
speaking. Also, they
Stephen Bosch [EMAIL PROTECTED] Wrote: 4/9/2007 7:12 PM:
Joe Acquisto wrote:
In a system connected to a verizon T1, Digium TE411P (quad T1 echo
cancellation), client is complaining it is too quiet.
The complaint regards calls over the T1, not in house SIP only calls.
Their description
Tzafrir Cohen [EMAIL PROTECTED] Wrote: 4/6/2007 9:53 PM:
On Fri, Apr 06, 2007 at 12:37:47PM -0700, Lee Howard wrote:
It's usually built and left in the zaptel source directory where you
extracted and built zaptel. If it doesn't get built for you from
zttest.c then check the Makefile that
Tzafrir Cohen [EMAIL PROTECTED] Wrote: 4/6/2007 9:52 PM:
On Fri, Apr 06, 2007 at 03:19:16PM -0400, Joe Acquisto wrote:
zttest does not exist on this system, Suse 10 based. IIRC, I never
found the file(s) needed to compile it.
Do you actually have a timing source?
head -c 0 /dev/zap
Tzafrir Cohen [EMAIL PROTECTED] Wrote: 4/7/2007 8:24 AM:
On Sat, Apr 07, 2007 at 07:47:12AM -0400, Joe Acquisto wrote:
Tzafrir Cohen [EMAIL PROTECTED] Wrote: 4/6/2007 9:52 PM:
On Fri, Apr 06, 2007 at 03:19:16PM -0400, Joe Acquisto wrote:
zttest does not exist on this system, Suse 10 based
. . .
So your timing source is basically working.
--
Tzafrir Cohen
And this means . . . any FAX-ing issues must be due to other problems?
joe a.
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:
Paste here the rules of your extensions.conf for outgoing calls.
Sds,
Gustavo
From: Joe Acquisto [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Analog phones, dial
There seem to have been many discussions about this, so sorry if this is boring.
Can one connect a standard fax machine (or fax modem) to an analog port on a
TDM400p (as if it were an analog phone, say) and expect it to work reliably?
For sending, that is. Detecting and routing the call is
are on the other side of your SIP switch.
Bryan Johns
Partner
Shelton | Johns
1805 Old Alabama Road
Suite 200
Roswell, GA 30076
USA
Office: 678.248.2637
FindMe: 678.229.1809
Email: [EMAIL PROTECTED]
On Apr 6, 2007, at 8:39 AM, Joe Acquisto wrote:
There seem to have been many discussions
While pondering several issues, poor quality PSTN POTS lines, potential cost
savings with multiple cell numbers, the FAX problems over TDM400p, etc, I
wondered about:
Cell phone Base stations to replace POTS lines. Devices to cradle cell
phones and connect to TDM400p, for instance, to mimic
Gordon Henderson [EMAIL PROTECTED] Wrote: 4/6/2007 9:51 AM:
On Fri, 6 Apr 2007, Joe Acquisto wrote:
AFAIK, the FAX targets are normal FAX machines, on the PSTN.
What happens is, there appears to be a dial out, and a FAX negotiation,
but it always fails.
I think what Bryan is asking
Lee Howard [EMAIL PROTECTED] Wrote: 4/6/2007 2:15 PM:
Joe Acquisto wrote:
AFAIK, the FAX targets are normal FAX machines, on the PSTN.
What happens is, there appears to be a dial out, and a FAX negotiation, but
What it always fails.
What does zttest say about your zap card configuration
I find the source. IIRC, the problem was there was some dependency
(terminology ?), or package that was missing and I got complaints when trying
to compile. I can't recall what it was, but it is something that is included
in most distro's but not SUSE by default.
Is there a way to compile
J. Oquendo [EMAIL PROTECTED] Wrote: 4/5/2007 6:47 AM:
Joe Acquisto wrote:
Thanks. And this might go where, in rc.d/rc.firewall.local ?
But I don't get it. Isn't this redundant? Since I have port forwarding
already. . .?
joe
Steve Totaro [EMAIL PROTECTED] Wrote: 4/4/2007 8:44 PM:
Joe Acquisto wrote:
Attempts to do SIP thru firewall (IPCop) are unsuccessful. using x-lite
softphones, for eval/testing. They do get registered, and can call each
other, but mostly get no audio, sometimes one way audio
I have a system with a TDM400p 2FXO, 2FXS. Analog phones work fine, on
incoming calls, ring, answer, talk, hangup.
However, not so good dialing out. Pickup handset, get dail tone. Cli shows
Starting simple switch on Zap/1-1.
Press a key and get Hungup Zap/1-1 and get the doot-doot-doot
Gordon Henderson [EMAIL PROTECTED] Wrote: 4/4/2007 3:32 AM:
On Tue, 3 Apr 2007, Joe Acquisto wrote:
Is it possible to install a stun server on asterisk?
You can install a stun server on the same PC that asterisk is running
on.
No need for it to be part of asterisk itself, it's a totally
Attempts to do SIP thru firewall (IPCop) are unsuccessful. using x-lite
softphones, for eval/testing. They do get registered, and can call each other,
but mostly get no audio, sometimes one way audio.
Suggestions/fixes?
joe a.
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New install, Asterisk, obviously, Baystack 450 swtiches, verizon T1, Digium 4
port T1 card
Some (few) users have had complaints from their clients that sound quality is
poor. I do not know if the calls were placed via asterisk, or received via
asterisk. If it matters.
I believe this is a
. . .
http://sourceforge.net/projects/stun/
Which is linked from:
http://www.vovida.org/applications/downloads/stun/
That's what I'm running.
Gordon
Thanks. Looking there, why would I need a stun client if the
device/softdevice already has STUN support?
All I should need is
Easiest method in a nutshell...
iptables -A INPUT -s YOUR_HOST -i ETHERNET_CARD -d PBX_SERVER -p TCP -j
ACCEPT
iptables -A INPUT -s YOUR_HOST -i ETHERNET_CARD -d PBX_SERVER -p UDP -j
ACCEPT
iptables -A INPUT -s PBX_SERVER -i ETHERNET_CARD -d YOUR_HOST -p TCP -j
REJECT
iptables -A
Joe Acquisto [EMAIL PROTECTED] Wrote: 4/4/2007 4:24 PM:
Easiest method in a nutshell...
iptables -A INPUT -s YOUR_HOST -i ETHERNET_CARD -d PBX_SERVER -p TCP -j
ACCEPT
iptables -A INPUT -s YOUR_HOST -i ETHERNET_CARD -d PBX_SERVER -p UDP -j
ACCEPT
iptables -A INPUT -s PBX_SERVER -i
J. Oquendo [EMAIL PROTECTED] Wrote: 4/4/2007 5:58 PM:
On Wed, 04 Apr 2007, Joe Acquisto wrote:
iptables -A INPUT -s YOUR_HOST -i ETHERNET_CARD -d PBX_SERVER -p TCP -j
ACCEPT
iptables -A INPUT -s YOUR_HOST -i ETHERNET_CARD -d PBX_SERVER -p UDP -j
ACCEPT
iptables -A INPUT -s
Is it possible to install a stun server on asterisk?
joe a.
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Getting no service display on aastra 480i. Sip debug shows an unathorized
blub when the aastra tries to register.
Some reading indicates that 1.4 firmware wants aastra.cfg and mac.cfg in
/tftpboot/. There are none.
Anyone have basic config files? Or can point me to a good link? All links I
Joe Acquisto [EMAIL PROTECTED] Wrote on: 4/2/2007 3:03 PM:
Getting no service display on aastra 480i. Sip debug shows an
unathorized blub when the aastra tries to register.
Some reading indicates that 1.4 firmware wants aastra.cfg and mac.cfg
in /tftpboot/. There are none.
Anyone have
Michelle Dupuis [EMAIL PROTECTED] Wrote on: 4/2/2007 3:23 PM:
You don't need the cfg files (or a tftp) to boot the phones or register.
There are some sample configs lying around, but Aastra's are very poorly
documented (and their firmware still has big bugs - so don't modify from
default too
What are peoples experience with the reliability of the TDM400p. Specifically
in the 2 FXO, 2 FXS configuration, which is the 022 (?) model.
Is this board prone to random failures?
joe a.
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New install, using TDM400p. wctdm is loaded, asterisk loads. Zaptel and
zapata.conf are from a working system, same model board, same module locations.
CLI command zap show status shows all OK, zap show channels shows nothing
defined.
Incoming calls show nothing on CLI, analog handsets have
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