How would you categorize a Yellow Alarm sensed by the Asterisk side in a Legacy PBX integration?We have a Mitel SX200 connected to an Asterisk(1.2.4) with a TE110P.Twice today (first time in over a month) we received a Yellow Alarm on the TE110P. I have been able to clear it easily by restarting
On 2/23/06, C F [EMAIL PROTECTED] wrote:
So my question is, is anybody else using 7.5 firmware?I haven't had the same issue you describe so I can't help, sorry. But I have had other issues with 7.5 firmware. If asterisk restarts then the phone needs to be rebooted to re-register.
I haven't seen
On 2/23/06, Geoff Manning [EMAIL PROTECTED] wrote:
How would you categorize a Yellow Alarm sensed by the Asterisk side in a Legacy PBX integration?We have a Mitel SX200 connected to an Asterisk(1.2.4) with a TE110P.Twice today (first time in over a month) we received a Yellow Alarm on the TE110P
On 2/28/06, Chris Miller [EMAIL PROTECTED] wrote:
I'm chasing down a pop/click type of disturbance on a PBX system.Strangely, the disturbance is only heard by the outside caller, theinternal recipient hears the caller crystal clear. This seems to havecrept up when upgrading the zaptel driver to
Sergio Chersovani wrote:
I know it's hard to find out infos at the cisco site.
Maybe you can open a TAC case
Sergio
I did find this info:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20phone%20cisco%2079xx
comments_threshold=0comments_offset=0comments_sort_mode=commentDate_desc
[EMAIL PROTECTED] wrote:
Has anyone attempted or have had any issues getting Asterisk and
Digium cards working on a Dell SC420 server?
I have it up and running on 2 SC420's with TE110P cards. Haven't had any
problems as of yet (knock on wood).
Dell SC420
TE110P
2x40 GB SATA drive in a RAID 1
Matthew Boehm wrote:
wassim darwish wrote:
Jul 14 17:30:58 WARNING[14196]: codec_g729.c:180
g729tolin_framein: Out of G.729 Decoder Licenses!
Well, if that isn't the most self-explanatory error in the entire
asterisk code, I don't know what is.
++
Skype uses wideband-ilbc.
I don't think thats right. I think it just uses iLBC over it's own
proprietary Voip protocol.
http://www.skype.com/help/faq/technical.html
How much bandwidth does Skype use while I'm in a call?
Skype automatically selects the best codec depending on the
Adam Dobrin wrote:
Rate: 0.0189 for calling Taiwan via NuFone
Duration: 930 seconds
Lets vote for the answers:0.7269 or 0.2929 ???
I get .31$. Where did you all go to school? Is there a connection
charge?
???
how about .29 (rounded)
.29295 = (930 / 60) * 0.0189
Mike Clark wrote:
I am interested. We have just started using Nagios, so this could be a
nice add-on.
Mike Clark
Jeremy Melanson wrote:
If anyone is interested, I can send the script (it's in Perl) and an
example of how to check the PRI status in Nagios.
I'd love to hear from other
wassim darwish wrote:
what is the most stable linux that we can build
business on it, i mean the best linux a linux without
problems .
You're probably asking the wrong people, well the *right* people but the
*wrong* forum. A lot of people here will have opinions of their favorite
linux
-Original Message-From: Tim King
[mailto:[EMAIL PROTECTED]Sent: Tuesday, August 02, 2005 11:29
AMTo: 'Asterisk Users Mailing List - Non-Commercial
Discussion'Subject: [Asterisk-Users] WHat does it
take
How many times do you ask for help here before getting a
respone? Every single
I am having quality problems on SIP bound calls made over the Zap channels.
All Sip only calls (Cisco phone through Asterisk to another Sip device sound
fine).
Our setup looks like this:
User -- Executone PBX -- Asterisk Server -- Router -- Internet
The user is using a legacy handset that works
Every so often, and it seems that it happens only when a call is in
progress, all 24 Zap channels get reset. All channels are opened and then
timeout. This causes the in-progress calls to terminate.
There are no corresponding Red/Yellow alarms on wither the PBX or Asterisk
although we do receive
How much of an impact can/does local network traffic have on call quality?
Would opening large files on local servers affect call quality? We are
running QoS on the router but that will only prioritize traffic in/out of
the network.
___
Asterisk-Users
Michael Graves wrote:
Sure it can. If you have a network segment that's fully saturated and
you're also pushing VOIP data over that segment you'll have problems.
In practice most networks are not that busy, but it can happen. If
your phones, switch and NICs are VLAN capable you can setup a
Michael Graves wrote:
Oh, yes! That's a good possibility as well, expecially with some Cisco
gear.
One problem that I had was related to saturating a segment during an
automated backup procedure. When a server in the UK started its backup
processes at an apparently idel time callers in the
Eric Wieling aka ManxPower wrote:
Are your phones on shared links to the switch?
i.e.
PC - Phone - Switch?
Actually it is a legacy PBX - Asterisk integration
Legacy Handset -- Mitel SX 200 -- Asterisk -- Switch -- Router
The calls come inbound over the internet as SIP to Asterisk and are
Eric Wieling aka ManxPower wrote:
In my experience, for local LAN audio issues, duplex problems are the
problem, not LAN traffic.
Rock on!
I am in half duplex mode:
serv01:~# ethtool eth0
Settings for eth0:
Supported ports: [ MII ]
Supported link modes: 10baseT/Half
Edwin Lam wrote:
hi folks.
i'm planning to connect * to 120 POTS line. i've done some research
on FXO cards but unfortunately most manufacturers only make 4
ports/card. the most i've found is 12 ports. so do i have to get 10
of these cards and setup 3 Asterisk servers (assuming each have 4
Julio Arruda wrote:
Half duplex by itself doesn't hurt (depends in number of calls and etc
really, but anyway...)
What is a killer for VOIP is duplex mismatch.
If you have autonegotiation enabled, and your peer (the switch ?) has
autoneg off, and 100/Full-duplex hard coded, you WILL have a
We are having line noise issues in our Asterisk to legacy PBX integration.
All SIP calls originating from IP phones sound crystal clear. All calls that
originate from the legacy PBX (Isoetec 228) and route through the Asterisk
and out SIP have a lot of line noise.
I believe I have it pinned down
Kevin P. Fleming wrote:
Geoff Manning wrote:
The TE110P card in the Asterisk server is set as the sync source:
span=1,1,0,d4,ami
em=1-24
That is incorrect. You have your span configured to recover timing
from the T1 and use that as the source for the card. If you want this
span
Peter Svensson wrote:
A blue alarm sounds really strange. That indicates that the remote end
(asterisk) in this case does not want to play at all. On a T1 it is
sent as a continous series of unframed 1:s. I am not sure if asterisk
ever sends a blue alarm (Alarm Indication Signal).
Receiving
Kevin P. Fleming wrote:
Geoff Manning wrote:
The TE110P card in the Asterisk server is set as the sync source:
span=1,1,0,d4,ami
em=1-24
That is incorrect. You have your span configured to recover timing
from the T1 and use that as the source for the card. If you want this
span
Peter Svensson wrote:
Which side shows the slips?
The slips are seen on the legacy PBX side (Isoetec 228)
I am not that familiar with T1, Are you sure the signalling between
the pbx and asterisk is set the same on both?
Unfortunately I am not aware of the signalling set on the Isoetec
Peter Svensson wrote:
I am not that familiar with T1, Are you sure the signalling between
the pbx and asterisk is set the same on both?
I have unearthed some documentation on the programming side of the legacy
PBX. I can set the following on the PBX for each line on the T1 card:
Line Type:
Bruce Ferrell wrote:
You need to be looking at a lower level
Like hw/cabling errors?? If so, that's what I was afraid of for cost
reasons.
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Asterisk-Users@lists.digium.com
Jon Pounder wrote:
Geoff Manning wrote:
Bruce Ferrell wrote:
You need to be looking at a lower level
Like hw/cabling errors?? If so, that's what I was afraid of for cost
reasons.
just out of curiousity - what are you paying for a T1 cable that you
are worried about cost ? you do
Jon Pounder wrote:
Bruce Ferrell wrote:
You need to be looking at a lower level
Like hw/cabling errors?? If so, that's what I was afraid of for cost
reasons.
no - the stuff you found relates to configuring one 64k channel of
the T1, you need to find the settings to configure the
A red alarm means I don't see any signal. A blue alarm means I
see a signal and something downstream (repeater etc) is saying they
don't see
a signal.
I used this as my reference:
http://www.fratec.com/FAQ/NFO/NFO_WAN_009.HTML
Slips sometimes cause an LOF condition, sometimes they
Now that I've looked back over my work for the past few days I realize that
I was trying to play with the txgain/rxgain to adjust the levels and hope to
smooth out the line noise. Well, any integer other than zero for either of
those values causes BLUE alarms and all the channels to reset in
Patrick Fortin wrote:
At first it was on scsi drives but we re-installed using a IDE drive
We deactivated the two onboard nic and tried two different brand.
We have deactivated hyper-treading
We have deactivated USB
We have deactivated SATA
We have tried a noise-cancelling power-bar
We
I am trying to test the T1 card in our legacy PBX but the connector to the
card is a 15 pin serial cable. I would like to make it myself so I can try
this test today. Does anyone have a pinout for it?
I just made a T1 RJ-45 loop back to test my TE110P and it tested out fine.
I'm trying to resolve
I just installed Xorcom Rapid and I'm trying to connect with Xlite.
In my SIP Proxy I have set the Domain/Realm and SIP Proxy as the IP Address
of the new install. I can ping that box.
When I try to connect I get hung on the Awaiting Proxy login information
and the log reads:
In my SIP Proxy I have set the Domain/Realm and SIP Proxy as the IP Address
of the new install. I can ping that box.
When I try to connect I get hung on the Awaiting Proxy login information
and the log reads:
© 2004 Xten
I have been successful in setting up asterisk and making workstation to
workstation SIP calls. But I am lost when it comes to anything past that.
We are trying to integrate this asterisk server into with our Executone
(432?) PBX to allow us to make outbound SIP calls between our disparate
Correction:
The hardware is a Wildcard T100P (not a TE110P)
Thanks!
-Original Message-
From: Geoff Manning [mailto:[EMAIL PROTECTED]
Sent: Wednesday, May 18, 2005 9:07 AM
To: Asterisk Users (E-mail)
Subject: [Asterisk-Users] Integrating Asterisk into our Legacy PBX
--Newb
I
I haven't been able to find much support for the RDNIS or DNID variables
online.
I am trying to prove a concept of call routing before we move towards
development of a production system. I need to have calls routed coming into
a call center based on DNIS. What type of syntax is needed in the
I was wondering how the Dell SC420 will perform under normal office to
office communications. We would equip each server with a T1 card to make
office to office SIP calls. They will integrate into our existing PBX
systems.
Does anyone on this list use this hardware currently
Thanks!
will
need to be either careful (spare parts stocks, failover
servers, etc.) or
lucky to avoid a day or two of downtime each year.
William
Signate
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Geoff Manning
Sent: Friday, May 20, 2005 12:01 PM
Well, if you are only making office to office calls, save the $500 per
T1 card and just use NICs.
The T1 card is only required if you are using a voice T1. If you are
doing IAX to IAX for example between offices, then Asterisk is your
friend.
Avoid SIP altogether as it is not needed and
Those individuals that are responding to the OP should probably note
what type of digium cards they are working fine with. One of the
obvious issues is the T1 card vs TDM card, since the TDM seems to be
far more critical then any other digium card.
Thanks.
We will be looking to
I am about to perform this same installation Scott so I'll be wathcing this
closely.
What type of T1 card do you have in the Mitel? Does it take a 15 pin serial
or an RJ 48x? The one I need to install to has the RJ 48x and we are trying
to figure out if it needs to be straight through or
Is it possible to route calls based on the number called when the inbound
call is SIP based?
Here is what we are trying to do:
1) Someone dials one of the companies 5 long standing, published phone
numbers which have been forwarded to ONE Voip telephone number by the telco.
2) The SER server
Is this even possible or am I better off getting a voip number for each of
the existing numbers I want to forward.
Thanks!
-Original Message-
From: Geoff Manning [mailto:[EMAIL PROTECTED]
Sent: Friday, June 03, 2005 4:53 PM
To: Asterisk Users (E-mail)
Subject: [Asterisk-Users] Call
sipgetheader(or_To=To)
Cut(or_To,or_To,:,2)
Cut(or_To,or_To,@,1)
That works! Thanks!
Correction to the cut command below, replaced , with = :
sipgetheader(or_To=To)
Cut(or_To=or_To,:,2)
Cut(or_To=or_To,@,1)
___
Asterisk-Users mailing list
When I start Asterisk, I receive these errors:
Jun 13 16:26:05 WARNING[2870] chan_iax2.c: Unable to support trunking on
user 'gv_trunk' without zaptel timing
Jun 13 16:26:05 WARNING[2870] chan_iax2.c: Unable to support trunking on
peer 'gv_trunk' without zaptel timing
Jun 13 16:26:05
Every now and again we are receiving this error in our logs:
Jun 14 15:54:42 WARNING[11137] chan_sip.c: Maximum retries exceeded on call
[EMAIL PROTECTED] for seqno 101
(Non-critical Response)
We resolved a previous issue of this type but that was a Critical Response
and occurred due to bad
The best type of error possible, intermittent.
We have PSTN numbers being switched to SIP then forwarded to our Asterisk
server which sits inside our LAN
Every once and a while (maybe 1 out of every 20 calls) goes like this:
-- Executing Answer(SIP/213.199.36.50-0818e3e8, ) in new stack
the more restrictive localnet the better results
at handling sip
devices behind NAT devices.
Gene
19. Call being answered, but no audio on either end
(Intermittent) (Geoff Manning)
--
Message: 19
Date: Tue, 14 Jun 2005 17:30:31 -0400
From
Hello,
I am not too familiar with the settings in our PIX (learning though).
Here is the only access-list setting that we have in place for Asterisk:
access-list acl-prod permit udp any host EXTERNAL_*_IP_HERE eq 5060
In rtp.conf we are allowing ports 1 - 2.
We are not using SIP Fixup
as to port forwarding the correct ports through our PIX and
if that has an effect on my audio issues? We have old firmaware that has a
bug in the SIP Fixup so it has been turned off. Could that be the issue?
-Original Message-
From: Geoff Manning
To: Asterisk Users (E-mail)
Sent: 6/17/05 1
I apologize in advance for posting this yet again (3rd time actually). But I
have a little more data to share this time so bear with me.
I have Asterisk running on an internal IP address behind a Cisco Pix 515
with firmware version 5.2(3)
Here is the setup
Mitel SX200 PBX --- Asterisk ---
I have a second-hand 7960 which I am attempting to upgrade to
use a SIP
image.
The phone currently has a firmware release which doesn't seem
to be listed
in Cisco docs - P003AM30. On reboot, it finds the tftp server
Here's how I performed the upgrade:
Downgrade from the stock P003AM30
I trying to obtain some information relation to implement
Zoom x5v 5565 and Asterisk
What exactly are you trying to do? Are you trying to use Asterisk with the
Global Village service? I assume you have the X5v up and running and
providing internet access.
Could someone help
me out with how I can rotate asterisk's
log's without killing the process?
Does restarting the syslog service help?
# service syslog restart
or
# /etc/init.d/syslog restart
___
Asterisk-Users mailing list
Asterisk doesn't use the syslog daemon tho does it? I thought it
did internal logging to a file.
My mistake, you are correct (both of you actually!)
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
We are runnign into problems where our legacy PBX reaches a frame loss threshold and takes it's T1 card offline (the T1 card that interfaces with the Asterisk servers TE110P). During this time, the Asterisk server senses a Yellow alarm.
We've noticed that if we quit asterisk, stop zaptel, start
On 3/15/06, Geoff Manning [EMAIL PROTECTED] wrote:
We are runnign into problems where our legacy PBX reaches a frame loss
threshold and takes it's T1 card offline (the T1 card that interfaces with
the Asterisk servers TE110P). During this time, the Asterisk server senses a
Yellow alarm
On 4/21/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
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,
We had a user report that they were on a SIP --- PSTN call for about 4.5 minutes before the call went to on-way audio. The user called the person back and they reported being able to hear my user, but my user couldn't hear them. The audio condition persisted for about 15 seconds before the user
I have a Dell PE SC420 (a no-no with a TE110P) connected to a Mitel SC-200. The Mitel gets Slip and Frame errors that cause the T1 card in the Mitel to go offline and this causes a service interruption. Could the SC-420/TE110P be causing these errors? I know it is listed on the incompatibility
Here are the settings that I have for the Mitel and the Asterisk server, as well as logs of errors, etc. We've been chasing this issue for months now and it's getting frustrating. Any help would be appreciated!Thanks!
On 4/27/06, Geoff Manning [EMAIL PROTECTED] wrote:
I have a Dell PE SC420
My Setup looks like this:
Mitel 200 SX (1st T1) Bell South
(2nd T1)
|
|
|
Digium TE110P
Asterisk
MITEL CONFIGURATION
Primary Timing Source: 1st T1 Card
Secondary Timing Source: 2nd T1 Card
ASTERISK CONFIGURATION
span=1,1,0,d4,ami (Look to the Span for timing)
We
We are looking to acquire E1 service in Fleet right outside of London. I am
in the States so I am not aware of the key players. We currently get ADSL
from Eclipse but were interested in a quote for E1.
What is a typical E1 line go for nowadays and who can I get it from?
Thanks,
Geoff
We are using Cisco 7960 and 7940 phones to connect to a remote Asterisk
server over DSL. They work fine from boot up but after half a day they stop
registering. They are set to register every 60 seconds. They need to be
rebooted in order to register again. When we look at the log on the modem we
Braz wrote:
Hi.
Which Cisco firmware are you using? There's a known problem with lost
of registration with SIP 7.5 firmware and you need to reboot the
phone in order to re-register. If you're using this one, try to
downgrade to 7.4.
Ahhh, the phones are running 7.5!! I have another
Braz wrote:
Hi.
Which Cisco firmware are you using? There's a known problem with lost
of registration with SIP 7.5 firmware and you need to reboot the
phone in order to re-register. If you're using this one, try to
downgrade to 7.4.
Braz
Is there any more documented information about
Paul wrote:
-Original Message-
What information do you need on the 7960?
Paul
Info relating to the 7.5 firmware version and it failing to register. Thus
needing a reboot to fix:
Hi.
Which Cisco firmware are you using? There's a known problem with lost
of registration with
[EMAIL PROTECTED] wrote:
Also - an outside chance - make sure Tip and Ring
are correct. You could be getting ground loops - depends on the noise.
I am having noise and slip errors between my TE110P and a legacy PBX T1
card. Could this be the same symptom? The connection is made using a 15 pin
[EMAIL PROTECTED] wrote:
Probably not Geoff. It is still digital at that point I think.
It should be coming to you as a four wire balanced circuit.
It depends on which legacy PBX you are using (tho it is pretty
standard) And if it wasn't right - it probably wouldn't work at all.
Brett
gincantalupo wrote:
Hi,
is there anyone trying a power over ethernet solution to feed IP
phones? I'd like to buy a good but cheap hub/switch but I don't
know which. Can anybody help me??
We are testing out the 3Com 2226-PWR Plus ($800US roughly). We haven't made
it too far but the phones
We are drafting a plan for a new office setup. The users will be using Cisco
7940 phones registered to a remote Asterisk server. We were thinking of
using two ADSL lines coming into a Multi-WAN router to allow for load
balancing. As opposed to setting up half the users on one ADSL line, half on
One of the command line parameters is the -c flag which is supposed to allow
you to run custom commands. Does that mean commands that I can write into
the check_asterisk perl script?? It doesn't allow you to run the asterisk
manager commands that I can see. Anyone using check_asterisk for anything
Say I had a crossover cable that connected a Mitel SX200 to a TE110P and the
pinout was done as such:
1 - 4
2 - 5
5 - 1
4 - 2
(the 5 and 4 are transposed on the left side)
Instead of the proper way of:
1 - 4
2 - 5
4 - 1
5 - 2
What would the results be? We have had the former as our
We have integrated an Asterisk (TE110P) and a Mitel SX200. We usually get
over 500 frame errors and over a 500 slip errors per hour. When the errors
reach 1000 per hour the Mitel will take it's T1 card offline. At that point
no calls can be routed from the Asterisk server to the Mitel and the
Eric ManxPower Wieling wrote:
span=1,1,0,d4,ami
em=1-24
Looks like you have told Asterisk to get it's timing from the Mitel.
I'll bet the Mitel is trying to get it's timing from Asterisk.
Try span=1,0,0,d4,ami and run ztcfg -vvv
I just set this back. It was originally set to your
Dennis Walker wrote:
But I did find that down in the t1 parameter settings you can set the
limits higher. I maxed them out and the problem went away, it would
reset the count in a rolling 24 hours luckily the slip count just
stayed below the limit.
Do you by chance now what the max
Eric ManxPower Wieling wrote:
span=1,1,0,d4,ami
em=1-24
Looks like you have told Asterisk to get it's timing from the Mitel.
I'll bet the Mitel is trying to get it's timing from Asterisk.
Try span=1,0,0,d4,ami and run ztcfg -vvv
We turned on the zaptel debugging and noticed the
We are trying to debug a connection between Asterisk and a legacy PBX (Mitel
SX200). We turned on the Zaptel debugging and we get the following message
quite frequently:
Oct 12 07:14:09 localhost kernel: T1: Lost our place, resyncing ( 28 )
Oct 12 07:14:09 localhost last message repeated 3 times
Kerry Garrison wrote:
The SC430 will experience unusable call quality with a TDM400P due to
IRQ Sharing problems. If you have some magic to get around this,
please share because everyone I know that has tried using an SC430
has given up and switched to other platforms. -Kerry
I experience
? All the
voice packets would become fragmented so it sounds logical.
And simply changing the MTU on the modem, will that fix it, I can't find a
way to change it at the Cisco phone level.
Thanks,
Geoff Manning
___
--Bandwidth and Colocation provided
Rich Adamson wrote:
There will be a delay associated with any sip-to-sip call, but it
should not be all that noticable unless both the talker and listener
are in the same room.
Are you sure this is a delay problem, or might it be a half-duplex
problem?
If any of the hardware mentioned
Matt Riddell (IT) wrote:
Geoff Manning wrote:
Our users are experiencing some unacceptable delay when trying to
have a conversation. The delay is so noticeable that they keep
stepping on each others words and resort to calling the customers
via cell phone.
We've had some pretty bad delay
Rich Adamson wrote:
Be carefull with vlan assumptions. If two or more vlans exist across
multiple switches, how do you know if another vlan hasn't consumed all
available resources leaving little (or none) for your phone vlan?
Hint: look for discarded packets in or out on the physical ports
Rusty Dekema wrote:
How far (physically) is the Asterisk server location from the
location of the phones? Have you tried pinging the Asterisk server
from the network to which the phones are connected?
As a rule of thumb, If the two sites are within 2500 miles of each
other and the network
Rich Adamson wrote:
Absolutely not. The MTU is the Maximum Transmission Unit, and sip
packets are about 214 bytes in size (including all pkt headers). Way
smaller then the MTU.
If the only thing on my network are these Cisco Phones, would lowering the
MTU encourage more efficient transfer of
Rich Adamson wrote:
No. The reason is that if the phones are the only thing on this, the
size of the sip packets will never be greater then 214 bytes.
Given your table below, there are other devices on your network and
6% of those are sending packets of in the 512 to 1023 byte range.
Andrew Kohlsmith wrote:
My suspect is the SDSL modem; what is it? We use ADC Megabit modems
here and they work fairly well. We've had some issue with the old
Flowpoint 5250s.
It is a Speedtouch 610s. Seems like a pretty robust small biz class modem
but it could be the issue. We are just
Pete Barnwell wrote:
Are you sure about that? Most ADSL in the UK is on PPPoA (BT supplied
- it may be different for LLU providers), not PPPoE so I wouldn't
think this has actually changed.
Correction, you are right. The old ADSL we were running was indeed PPPoA.
That has not changed.
As an update and back to the original response from Rich re: duplexing
The topology looks like this:
8 Cisco IP Phones
|
|
3COM 226 PWR-Plus
|
|
Speedtouch 610s
|
|
Rich Adamson wrote:
So if I leave it as is (both set to Auto) then Flow Control is
Disabled on the 3COM switch
If I configure it so the Flow Control is Enabled then the 3COM
defaults to Half Duplex.
Is there a way for you to use ethereal to see what's coming through
the dsl circuit?
Rich Adamson wrote:
What happens if you set the 3com AND Speedtouch to full duplex?
Setting both to Full Duplex (10 or 100) shuts the port on the 3COM switch
down!
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing
We use
SetVar(TIMEOUT(digit)=8)
In our dialplan to make sure that the user is done dialing before Asterisk
executes the call. I just recently came across the piece I've copied below.
It says for new incoming ZAP connections, the default digit timeout is 3
seconds and can only be configured in
Prezydent Kaczynski wrote:
It there avalible quality and delay test for sip connections for
asterisk. Something like to clients making a call with different
codecs and measuring delay , jitter ? I know there is a Astertest but
in that you need 2 asterisk mashines (which is usually hard to
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