Thank you for confirming that I'm here!!! :).
On Thu, 19 Nov 2009 18:30:50 -0800, Todd Hodgen wrote:
> yes
>
> -Original Message-
> From: sipx-users-boun...@list.sipfoundry.org
> [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
> m...@grounded.net
> Sent: Thursday, November
For anyone who gives a hoot, I'm changing my list name to m...@grounded.net
because my
address looks too spam like :).
Thanks for the good times, I'll truly miss them and BRB!
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> When you say feature codes, do you mean in the TELCO world, STAR CODES (*67
> to block outbound caller id or do a forward calls function)?
Right, *xx codes. They are called Feature Codes in asterisk and in the google
searches
I've been doing at least.
> If so, I wouldn't know how to do this
> What exists that might be on such a list depends on what you thing
> 'standard feature codes' are and what they do.
Well, most phones seem to have similar, but sometimes, a bit different feature
code
settings. Asterisk for example has a number of feature codes pre-set so have
always
followed
I know the end devices have their own feature codes but for testing of the
system itself,
this is what I am not able to find much on.
On Thu, 19 Nov 2009 11:42:27 -0600, li...@grounded.net wrote:
> Is there a list of standard feature codes for sipx? For example, doing an
> echo te
Is there a list of standard feature codes for sipx? For example, doing an echo
test and
such things?
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Someone was trying to reach me on this matter but I don't seem to be able to
reply. Also
not sure why my email address might not be showing up in the list but I can be
reached
directly as well at li...@grounded.net.
Thanks!
On Wed, 18 Nov 2009 14:59:19 -0600, li...@grounded.net
I've asked this several times but it was burred in a long thread so thought I
would ask
again.
I have a used Media Gateway 3200 (Mediant 2000) I purchased which I would have
loved to
use with sipx. However, I have read that there are possible problems with older
software
versions and more imp
On Tue, 17 Nov 2009 13:07:40 -0500, Tony Graziano wrote:
> NAT can be disabled if noone needs to reach it from the outside. RULES
> needs to stay enable or it will break access via the LAN.
Wait now, yes, no NAT, no one can reach it from outside, but, can you expand on
what you
mean about the ru
> NAT can be disabled if noone needs to reach it from the outside. RULES
> needs to stay enable or it will break access via the LAN.
Can you expand on this, that's interesting, I was not aware of this.
For example, I need to give access to sipx from another web site, which simply
has a
wrapper
Weird, another person gave it a try and they can get in also. Something isn't
right.
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On Tue, 17 Nov 2009 12:47:46 -0500, Tony Graziano wrote:
> When I did this from outside of your network to the public IP it worked, as
> in, port 80 redirected me to 8443 and loaded the login screen.
No one else has been able to get that screen. Going to the IP did it uh?
>As long as both ports a
> I also tried from the outside when you originally did this and port 80 was
> redirected to 8443. So I think it is either an internal route or a proxy
> config that you might be using inside.
Thanks for the input and I'm sure you're right. I wondered also but it looks ok.
# route
Kernel IP routi
Bit stumped here.
Using Tony's pfsense config file, sip/rtp traffic gets in just fine but 80/8443
does not.
I think it's a sipx setting and not pfsense because I also tried giving remote
access to
sipx over my main firewall and sipx never wants to reply to anything outside of
the lan.
I've be
> software from vendors that can adhere to problems escalations and rapid
> software fixes as needed. The burdens for this in the open source world,
> due to the legal needs to always evolving e911 regulatory requirements, are
> pretty harsh.
I was hoping that the system was fairly standardized by
How are some of you dealing with the 911 requirement when it comes to remote
phones?
I have agreements in place which are pretty much just our updating a text file
and firing
it off to our providers which I'm not very comfortable with.
The providers like to say 'it is extremely expensive to set
I didn't want to start another thread but have been asking if anyone using
these might
know if the 4.x firmware running on the used mediant 2k I bought would work
with sipx.
Mike
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> when you ran opensbc IT was the NAT traversal device instead of pfsense
> is all.
One very big difference is that now I'm using pfsense, which is a firewall. The
only
option I seem to have is pfsense in multi wan setup since sipx can only have
one default
gateway. Unless of course, I might b
On Wed, 11 Nov 2009 09:37:04 -0500, Tony Graziano wrote:
> You have to decide what you want to do with the multiple servers.
I know what I'd like to accomplish, fail over on the front end (WAN sides), and
reliability, high availability on the back end (sipx).
I'm asking questions to understand c
On Wed, 11 Nov 2009 03:46:18 -0500, Tony Graziano wrote:
> psense can be run in clustered mode OR it can be run with dual wan
> support.CARP (harware failover) is supported and state table sync (pfsync,
Yes, I was reading about that last night and have it in my browser tabs
actually. I was
just
I've been looking high and low to try and get a handle on this.
I was using opensbc in front of the sipx box. The beauty of this was that the
sipx box
didn't need NAT traversal so it was easy to add entry points, load balance/fail
over using
DNS.
Since I am now using PfSense firewalls, I need
Any chance you could give a new Mediant 2K user a hand? :)
On Sat, 31 Oct 2009 18:59:38 -0500, Kgee Gee wrote:
> AudioCodes is supported and really easy to setup with sipx.
>
> On Sat, Oct 31, 2009 at 6:54 PM, li...@grounded.net
> wrote:
>> So if I get rid of this blasted ve
> service that delays the transport of media in favor of other well known
> protocols to give them priority. Welcome to the Internet.
Yup, no kidding :). This I am aware of very well as I suspect one of our own
providers and
I've also used PacketShaper hardware myself at one time.
> If your ex
Today's testing was interesting. We are seeing some cases where audio is just
one way or
can't connect at all. Since it works pretty much all of the time otherwise,
have to
believe it's related to the hot spot.
Still, that is a problem, it needs to be 100% of the time.
Mike
BTW, finally got into the box but am not sure it can be used with sipx as it's
running
what seems to be a very old 4.x version of software, is used, and has no
support.
From what I've read, needs to be at least 5.x to work with sipx?
Versions;
Version ID: 4.20.354.6420
DSP Type: 48624
DSP S
> Nice article about load balancing with "sticky" connections in pfsense,
> very important with sip.
Yes, I noticed that in the Advanced section of the firewall and am meaning to
read that
but your link speeds that up, thanks :)
Mike
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>I don't think this thread is applicable to the subtleties of this.
Yes, this thread was/is more about building a system which can be reached
remotely,
consistently.
> http://sipx-wiki.calivia.com/index.php/High-Availability_Installation
Yes, I've seen this.
> Mike Picher's book has a great c
> or maybe both at priority 1 and see if the registrations balance. There's
> been a lot of talk on how to do this kind of thing on the list lately.
> Your two sipx installs perhaps need to be in HA so that if one fails, it
> will continue to allow registrations and calls.
This is the part I'm n
After testing yesterday, everything went well consistently, to the point where
I decided
to rebuild the sbc servers into pfsense servers instead.
I then had both setups working until both stopped working. I suspect too many
firewalls,
too many gateways, routes etc. I turned off the hardware of
On Mon, 9 Nov 2009 16:58:06 -0500, Tony Graziano wrote:
> It's a firewall. If you want to allow SSH or anything else through, you can
Right, I understand. I had read that sipx actually uses a couple of
authentication means
when someone connects. I am wondering what else I might need in terms of
Ok, so aside from the DNS input I'm hoping for, pfsense seems to work.
So, the next step would be security right. How safe is it to leave sipx
somewhat exposed
behind the firewall like this? I mean, there is no preliminary authentication
such as vpn
or something else for example. I would prefe
> those records tell whoever gets them that your system can use either TCP
> or UDP (but not TLS, because there's no record that says it can).
>
> Many SIP implementations (including most components of sipXecs) will
> just assume that those two both work if there are no NAPTR records.
I have some
On Mon, 9 Nov 2009 12:12:15 -0500, Tony Graziano wrote:
> Yes. It registered just fine (using sipdomain.com, not host.sipdomain.com).
This is the public side DNS setup.
sipdomain.com. IN NS ns1.mydomain.com.
sipdomain.com. IN NS ns2.mydomain.com.
$ORIGIN sipdomain.com
hst1.sipdomain
On Mon, 9 Nov 2009 12:12:15 -0500, Tony Graziano wrote:
> Yes. It registered just fine (using xxx.com, not xxx.xxx.com).
Thank you kindly for the input, that is helpful since we don't have a remote
handy right
now. Sending someone off to various locations to test right now and will update.
I'm
On Mon, 9 Nov 2009 11:38:49 -0500, Tony Graziano wrote:
> Ok then. It might be that filter was redirected, but it should hit it.
> Always works for me.
My bad. I was using one of those free Internet web proxy servers. When it sees
a
redirected connection, it blocks it. Maybe that prevents hacker
> That's intentional. Port 80 on sipx auto redirects to
> https://sipx.yourdomain.com:8443/sipxconfig/app for the user.
Right, as it does locally. Thing is, when I try connecting to port 80 from
remote, I don't
get redirected, I don't even see an incoming connection.
> sipx it should work unl
> OK. I see whats happening. Your NAT filters have the ip as 192.168.10.100
> for sipx. You'll want to change those to 192.168.10.80.
Yup, that was it alright. I used a 'compare' editor and mixed up what I wanted
for dhcp
and sipx addresses.
It's working now so we'll do some testing and see wha
Just spent hours trying to figure this out and one thing finally popped up. Our
connections are coming from outside, they are remote web browsers and sip
phones. Every
connection to the pfsense firewall is given a dhcp address and never routes to
the phone
system.
tcp In 69.19.14.31:9623 1
Got it, I'll take a look at that next.
> Like I said, make sue they both have the same route for your private
> network. If you have something else, like a L3 switch or another router for
> the private network that sipx is using, you should add a static route in
> pfsense for the same private netw
On Sun, 8 Nov 2009 10:22:54 -0500, Tony Graziano wrote:
> I would login to the cli of pfsense and make sure it can "ping or
> traceroute" to the sipx server. If the pfsense box is not the default route
> for sipx you should add a static route in pfsense for the local network I
> think.
I thought I
> Int Src SrcPrt Dst DstPrt NAT-Address NAT-Port Static-Port Desc
> WAN192.168.0.0/16 * * * * * YES
>
> Based on the last input I sent, do I need to change this to 192.168.10.0/24
> then?
I tried that change by connecting to port 80 fro a remote proxy server, no
difference.
tcp In 66
> Then go to FIREWALL>RULES>OUTBOUND
> WAN
> 10.255.252.0/22
> YES
> Auto created rule for LAN
> Make sure your subnet is stated there and the "MANUAL" box is checked.
Sorry for the multiple replies, just waking up.
Under Firewall/NAT/Outbound, I have Manual checked.
Int Src SrcPrt Dst DstPr
I think it might just be a subnet issue.
I'm using a /16 mask on 192.168.0.0 network because I recently installed
multiple layer 2
switches to move things across as many VLANs. I've not completed the wiring
(physically
connecting servers to their respective switches) so for now, everything is
Tony, do you want me to send you the xlm file? Maybe you'll notice something I
didn't
catch?
On Sat, 7 Nov 2009 22:31:11 -0500, Tony Graziano wrote:
> Go through your firewall rules and nat, make sure the ip's look right. Click
>
> through all the pfsense screens and make sure you network is t
On firewall, I can;
Traceroute output:
1 192.168.10.80 (192.168.10.80) 0.833 ms 0.307 ms 0.224 ms
Using the pfsense Ping tool, it just goes right back to the ping scree, nothing
shows up.
From shell, ping to both IP and name resolve.
Routing shows;
Destination Gateway Flags Refs Use Mt
Don't see anything that should not be there, but then, it's been a hell of a
long day so
far, maybe I'm missing some small thing.
On the sipx box, I'm not seeing any traffic from the firewall box either.
On Sat, 7 Nov 2009 22:31:11 -0500, Tony Graziano wrote:
> Go through your firewall rules
Everything is completed. The firewall is pfsense with Tony's xml file modified
to my
network. There must be something I have overlooked because connections don't
seem to be
making it past the firewall.
Calls work fine from inside phones, consistent two way audio, calls to/from
pstn over
pri/
Ok, so I now have everything installed, ready to go. I've also made notes about
the whole
process and will continue doing so. Perhaps I can pass that on to someone who
would like
to turn it into something others can use.
So now I have my new sipx box, ready to config.
I have my new pfsense box
> 1. Delete the siproxd package it has.
Done.
> 2. Make sure AON is enabled (the sipx wiki has some details about static
> NAT and how to enable it in sipx.
Done on pfsense but am not sure if you mean there's something else to be done
on sipx?
>I also posted a sample config for pfsense (xml d
r thirty minutes. Your deployment is a little different
> with vlans, but I think this will help you in any case.
>
>
> blog.myitdepartment.net.
>
> On Sat, Nov 7, 2009 at 10:16 AM, li...@grounded.net
> wrote:
>>> Each box can have their own SIP domain and for the
>Each box can have their own SIP domain and for the purposes of testing and
>characterization, remote users could have a line on either (or even both).
My hope was to have fail over ahead of the sipx box. The DNS load balancing
with opensbc
is pretty cool, I can watch connections flip flopping
On Thu, 5 Nov 2009 19:28:10 -0500, Tony Graziano wrote:
> You're so cute when you argue.
At 50 years of age, I tend to express thoughts but try not to argue :).
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n
>> that benefit the product long term.
>>
>> Wondering how many beatings I'll take on this email...
>>
>> -Original Message-
>> From: sipx-users-boun...@list.sipfoundry.org
>> [mailto:sipx-users-boun...@list.sipfoundry.org] On
res, fixes, and solution
> that benefit the product long term.
>
> Wondering how many beatings I'll take on this email...
>
> -Original Message-
> From: sipx-users-boun...@list.sipfoundry.org
> [mailto:sipx-users-boun...@list.sipfoundry.org] On Be
r than working on features, fixes, and solution
> that benefit the product long term.
>
> Wondering how many beatings I'll take on this email...
>
> -Original Message-
> From: sipx-users-boun...@list.sipfoundry.org
> [mailto:sipx-users-boun...@li
According to a guide, the following is used to reset this device.
1. Disconnect the gateway from the power and network cables.
2. Reconnect the power cable; the gateway is powered up. After approximately 45
seconds,
the ACT LED blinks for about 4 seconds.
3. While the ACT LED is blinking, pres
Ok, here is the reply I've received from my tester;
---
Well, here you go, but they look the same to me (mostly)
xxx...@eee1:~$ dig -t srv
_sip._tcp.mydomain.com_sip._udp.mydomain.com
; <<>> DiG 9.6.1-P1 <<>> -t srv
_sip._tcp.mydomain.com_sip._udp.mydomain.com
;; global options: +cmd
;; Got ans
> That command is asking the wrong question (it's asking for A records,
> not SRV records).
Oops, didn't even catch that. Too much going on, starting to miss details again.
I'll ask him to run the test again.
>Both of the answers you got were no-answers; if there was an A record,
>it would have a
> I'm not sure I understand the question. Did you run the dig or
> nslookup queries on both the private and public dns servers to see if the
> results were correct in both instances?
Ok, here is the result from the remote tester. The tester is using two
different DNS
servers and using dig to see
On Mon, 2 Nov 2009 19:56:26 -0500 (EST), Francis Tinio wrote:
> Doesn't g729 use less bandwidth than g711? So it should have better
> performance and resource management.
Yes, it's somewhere around the 8k mark, give or take headroom.
But this got me thinking...
So, my SIP gateway has g729 availa
> Audiocodes? I know due to their support "arrangements" the reseller has
> to supply the support.
Guessing that a support contract on these is pretty pricey. Are there any
other options for firmware updates on the 2k?
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> (4 supplies blown in one data center)
> Can't get replacement power supplies because
> audiocodes didn't have any in stocker. Wonder why?
Wow, ya, that's not coincidence. So they have either run out of replacement
supplies or have stopped selling what they know will blow up anyhow.
That's no
> I think the post he had about power problems and some of the replies he
> received indicated he had taken more than adequate precautions.
I didn't see the thread, I was just adding my own experiences in.
> Most likely a quality control or manufacturing defect and the product
Yup, happens.
Mi
> For the user having issues with their
> power supply, couldn't you keep a cold-spare of the power supply on hand
> or, even better, have dual power supplies?
Some other thoughts on that post. If a piece of hardware keeps blowing power
supplies, it could be something other than the power supplie
Anyone using a Jasomi PeerPoint c100 SBC. I'm trying to find the manual so
that I can reset the password.
Mike
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>> -NI2 support seems to be somewhat lacking but improving. This is
My provider PRI is NI2, is the audiocodes a problem?
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On Sat, 31 Oct 2009 19:51:01 -0400, Tony Graziano wrote:
> Its not asterisk. You don't really expect all software packages to have the
> exact same functionality.
That wasn't what I'm implying, just that users I would be move from asterisk
are used to having the emails. They would not be able to
So if I get rid of this blasted vegastream, how hard is it to get a mediant
2000 online real fast since I've never used one yet. In other words, would
there be anyone here that might be able to help me get it online to replace
this thing?
Thanks.
Mike
I understand what you're saying but I can't agree to all of what you said :).
Outgoing server name, URL's, reply to address, look of such a form/email,
those things should be a given.
> functionality that should be there it is built into the system after a
> review and "polls" or "votes" throug
Mike
On Fri, 30 Oct 2009 21:59:43 -0400, Andy Spitzer wrote:
> Woof!
>
> On Fri, 30 Oct 2009 18:32:41 -0400, li...@grounded.net
> wrote:
>
>> I very badly need to modify the email that is sent to the users to
>> notify them
>> about a voice mail to change the d
at can be added to the
>> mod before its too late.
>>
>>
>> On Fri, Oct 30, 2009 at 6:26 PM, li...@grounded.net
>> wrote:
>>> On Fri, 30 Oct 2009 18:17:31 -0400, Tony Graziano wrote:
>>>> you should post the question on the sipx-dev list though. it m
On Fri, 30 Oct 2009 18:17:31 -0400, Tony Graziano wrote:
> you should post the question on the sipx-dev list though. it might be that
> you have to open a jar file, change the text, and repackage the jar file.
I think I'll do that. I know what file most of the original files are in when
I do a re
So, editing the xml files aren't the trick either as it's too easy to break
the system.
There's got to be someone on this list who knows what file/s can be edited to
change this information.
Mike
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On Fri, 30 Oct 2009 17:18:43 -0400, Tony Graziano wrote:
> perhaps, but this might revert upon reboot, restart of services and/or
> upgrade. Amything you do in the sense of what you are trying to achieve is
> subject to that.
Right, no problem, I'm aware of that. My biggest challenge all day so fa
Gee, I don't suppose it's as easy as messing with the
/etc/sipxpbx/voicemail.xml file is it?
On Fri, 30 Oct 2009 13:46:43 -0500, li...@grounded.net wrote:
> So, I've been using grep all morning so far, looking for the hard coded
>
> notification message, does anyone kno
So, I've been using grep all morning so far, looking for the hard coded
notification message, does anyone know what file/s that might be in? I just
can't find it.
Mike
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ald | Systems Administrator | Brammo, Inc.
> Email: jmcdon...@brammo.com | www.brammo.com
> 550 Clover Lane | Ashland, OR 97520
> Phone: 541.482.9555 ext. 339 | Mobile: 541.531.0005 | Fax: 541.552.0414
> MCSA 2003
>
> -Original Message-
> From: sipx-users-boun...@list.sipfoundry.
Ok, so it's the same as 4.0.1, have to change all instances of 5.2 to 5.
Mike
On Fri, 30 Oct 2009 11:47:42 -0500, li...@grounded.net wrote:
> Looking on the wiki, trying to find the proper repo to use with 4.0.2.
>
> I recall that with 4.0.1, we had to edit the repo file, figu
Looking on the wiki, trying to find the proper repo to use with 4.0.2.
I recall that with 4.0.1, we had to edit the repo file, figured this would be
fixed in 4.0.2 but doesn't seem to be.
So, anyone know where I can get the proper repo file for the 4.0.2 ISO
install?
Thanks.
Mike
__
sers-boun...@list.sipfoundry.org
> [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
> li...@grounded.net
> Sent: Friday, October 30, 2009 1:52 AM
> To: sipx-users
> Subject: Re: [sipx-users] Msg Notice Modifications
>
> Lordy I wish this list was in fact forums, then
Lordy I wish this list was in fact forums, then I could simply edit my post
rather than have to keep adding messages to my thread :).
Anyhow, obviously, this doesn't seem to apply to version 4.0.2. There is not
General section un System. Is it somewhere else and I'm just not catching it?
http:/
> I would like to change the from address as it currently says from
> postmaster.
> Then the look and links, etc.
> HTTP ERROR: 404
Forget this, just half asleep here :).
What I need aside from changing the postmaster address is to change the URL
that is sent to the user. What I need is to chan
Is there a way of modifying the emails sent to users about messages left for
them, which can be retained through sipx updates etc.
Whether or not they can be retained is not as important as knowing where to
make the necessary changes.
I would like to change the from address as it currently says
> I have a TDM400 and a Rhino. Will these work with Sipx and if so where
> can I find info on how to configure them.
It is designed for external gateways but... if you look around on the net, you
could turn an asterisk box into a SIP gateway if you are familiar enough with
it. I say this because
If you mean the older FBII's for example, 1200's, 2500's, etc... those are
what I was using when I started doing voip and had to get rid of them as they
didn't support voip. Even doing a voip policy was not enough so had to
upgrade. I ended up going with juniper and in the long run, I'm now ending
Hi Dave,
Thanks for the input. Interestingly enough, we were just talking about that,
perhaps installing openvpn onto the vyatta/opensbc servers that we're using.
We have been toiling with the remote/mobile user for months and have not found
the right solution which works to date. Everything i
We have been struggling with remote users since day one. Lan users aren't a
problem at all and for remote users, we've installed vyatta/opensbc but to
date, cannot get consistent results.
I decided to try a vpn approach next, if not just for testing and started to
wonder if there are folks who
On Mon, 26 Oct 2009 12:09:33 -0400, Tony Graziano wrote:
> I've looked through the posts that I have here with vegastream. Noone
> reported success with BRI or PRI. Some problems, then they stopped posting.
In my case, it's working fine, we're using it on a daily basis. BUT... we've
only had at m
On Mon, 26 Oct 2009 11:39:51 -0400, Tony Graziano wrote:
> Sorry cant be more helpful. The reason I get Patton is direct factory
> support is included.
Ya well, like I said, lesson learned, the hard way... this time :).
Still, I do appreciate the input! Thanks.
Mike
s).
>
>
> On Mon, Oct 26, 2009 at 11:08 AM, li...@grounded.net
> wrote:
>> The config seems pretty straight forward, at least to get the PRI up and
>> running. Maybe there is some fine tuning which could be done on the
>> device but
>> that's where I need
;t know vegastream products or pretend to know them. You are using a
> device that might not have had a firmware update in over 5 years, so I'd be
> somewhat suspect if it is up to the task.
>
>
> I hope you find the answer you are looking for, but in the off-chance this
&g
Are those the screen shots you were wanting?
> What I was asking about is the setup option in the Vegastream. Do you have
> screen shots of how it is setup today so I can see what options they give
> you, an what options you are setup for?
___
sipx-us
On Sun, 25 Oct 2009 14:51:30 -0400, Tony Graziano wrote:
> ANY provider will provide you with what your timing source, framing, coding,
> etc. should be.
> When you are asked what the setup should be, it helps to answer those
> questions.
As I mentioned in another message in this thread, this was
> needs to be configured with repeaters that support it. It used to be more
> expensive for a B8ZS, so some people would order them without it.
Interesting, didn't know that. Thought that the options were when using E1
perhaps and/or other types of circuits. Haven't spent a lot of time learning
On Sun, 25 Oct 2009 14:26:11 -0400, Tony Graziano wrote:
> I never had a problem with 4.0.2 but haven't tried 4.0.3 yet.
Man, the list was jammed with folks having problems? :).
Anyhow, I'll go with 4.0.2 then since you sound confident.
Thanks.
___
si
I have to build another system, is it ok to use ISO 4.0.3 or better to use
4.0.2? I saw an awful lot of threads about problems with 4.0.2 so thought I'd
ask.
___
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On Sat, 24 Oct 2009 21:56:58 -0700, Todd Hodgen wrote:
> I suspect unchecked Clock Master puts it in slave clock, which is correct,
> as the Carrier circuit is running as master, and your device slaves off of
> it.
So leaving it unchecked still allows the timing to come from the provider
then?
On Sat, 24 Oct 2009 21:30:27 -0700, Todd Hodgen wrote:
> Is the Vega no longer made?
These older devices aren't even supported anymore from what I've found out.
It's not clear to me if they are doing much else, I think they (Lucent/Nortel)
is liquidating after a bk. I can't even find firmware up
On Sat, 24 Oct 2009 20:59:41 -0700, Todd Hodgen wrote:
> Short haul is probably only adjusting for the receive level, it has nothing
> to do with the distance of the t1 from its origination.
Yup, that's what it's doing. Of course, I should not need this, but I do.
Considering that we even had an
On Sat, 24 Oct 2009 20:33:36 -0700, Todd Hodgen wrote:
> There is no guessing with synchronous timing. There is a hierarchy that
> needs to be followed, and must be followed for it to work.
And that... is why I want to share findings because things are not working as
they should be, never have.
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