Hi Todd,
what did you mean with G5 fax machine? Is it a T.38 IP fax machine?
Is it better to turn off the T.38 redundancy too?
Jan
2009/6/11 Todd Hodgen :
> Are you sure the issue is with the Gateway itself and not the fax machines.
> It's important on the faxes to turn off Error Correction. An
Are you sure the issue is with the Gateway itself and not the fax machines.
It's important on the faxes to turn off Error Correction. And, it gets
trickey, because some machines allow you to turn them off, some require the
manufacturer to turn them off, and some turn back on by themselves when the
Is there a list of sipxecs tested/certified ATAs that work for faxing? I
was able to get a SPA-2102 to send/receive yesterday. Tests using a
AudioCodes MP-112 (version 5 firmware) have failed.
FYI - I just discovered HP has a very handy tool available. Call
1-888-hpfaxme (1-888-473-2963) and it wi
I have installed sipx live cd on hp 140 g1. I had since learned that CentOS
5.1 used by live cd doesn't play well with my server. I was told to upgrade to
CentOS 5.3.
Would the CentOS 5.3 upgrade interfere with sipx?
BTW... even with CentOS 5.3 there are know acpi issues.
If I was to in
On Wed, 2009-06-10 at 20:37 -0400, Matt Keys wrote:
> When a call came in for the
> hunt group it simply couldn't reach the extension it needed to. You'd
> think it would just continue on down the list if the first extension was
> unavailable though...
It should. The call to the first extension g
It was right around 1.10 or so. The phones would register fine but
couldn't dial outbound on the PRI (Mediant 1000) if they were registered
to the secondary. That's when I noticed the problem and forced them all
to the primary. By chance/luck it also cleared up the hunt group issue.
Both primary a
On Wed, 2009-06-10 at 15:08 -0400, Matt Keys wrote:
> I think I figured it out, thanks. Fix action was to take our secondary
> second server out and force all the phones to re-register to the
> primary only. The hunt group has been ringing perfectly ever since.
The secondary *should* work as well
I think I figured it out, thanks. Fix action was to take our secondary second
server out and force all the phones to re-register to the primary only. The
hunt group has been ringing perfectly ever since.
-Original Message-
From: Dale Worley [mailto:dwor...@nortel.com]
Sent: Wed 6/10/20
Dear Mr. Graziano,
I confirmed with VoicePulse that they do in fact require sending
'1757' even when dialing local numbers. I followed your instructions below,
and they worked perfectly! Thank you so much for your reply.
Best wishes,
Andreas
Systems Engineer
Around the Clock Informatio
I have a customer asking about Cisco and/or Linksys phones.
It's been a long time since I've worked with a cisco phone and never worked
with Linksys.
Anybody have experience with them? I know BLF and intercom are big deals for
the customer, so those would be a must.
Thanks,
Matt
__
The 320/330 is an entry level phone. The difference between them is POE. If you
need a need a desktop phone with really good hands free/speakerphone audio, get
a 650/670. You won't really be able to change the audio characterstics "much"
given the quality of the components in an entry level phon
> I think even if you call Polycom to Polycom the audio volume is low.
> We
> usually do as Tony suggested and turn on the volume persistence.
>
> If you crank up the volume on your gateway you risk introducing noise
> and / or echo...
>
> Mike
Thanks for the feedback and advice. Since you think
I think even if you call Polycom to Polycom the audio volume is low. We
usually do as Tony suggested and turn on the volume persistence.
If you crank up the volume on your gateway you risk introducing noise
and / or echo...
Mike
> -Original Message-
> From: sipx-users-boun...@list.sipfo
2009/6/10 Лебедев Сергей :
> Hello Ranga,
> Thank you very much for the help. The problem has been resolved.
>
> Best Regards,
> Sergey.
>
Can you please share your settings ( i.e. what goes into
/etc/sipxpbx/sipxbridge.xml ) without sharing your username or
password. I can update the list of int
On Wed, Jun 10, 2009 at 8:54 AM, Bryan Hiller wrote:
>
> I have a bandwidth.com account
> static IP and I am running 4.0
> my sip Trunk systems has been working, but today I got a system failed
> message. here is the message.
>
> Messages for SIP Trunking
> Standard error
> SipXbridge : Exception
I also got a message saying I could not write to
/var/log/sipxpbs/sipxconfig.log with user sipxchange
thanks,
Bryan
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I have a bandwidth.com account
static IP and I am running 4.0
my sip Trunk systems has been working, but today I got a system failed message.
here is the message.
Messages for SIP Trunking
Standard error
SipXbridge : Exception caught while running
log4j:WARN No appenders could be found for log
On Wed, 2009-06-10 at 17:30 +0700, Boy Aidil Sjam wrote:
> Scott,
> I've setup lab environment before go live to public network (internet), and I
> like to see how it works.
> There's no NAT involve in this lab testing.
> The scenarios:
> 1. I installed 2 sipXecs server, each server in their own d
Hello Ranga,
Thank you very much for the help. The problem has been resolved.
Best Regards,
Sergey.
08.06.09, 16:44, "M. Ranganathan" :
> 2009/6/8 Лебедев Сергей :
> > Hi,
> > I've been testing SIP trunk with sipnet.ru on SipX 4.1.0 and met problems
> > with calls setup.
> > It seems that th
Scott,
I've setup lab environment before go live to public network (internet), and I
like to see how it works.
There's no NAT involve in this lab testing.
The scenarios:
1. I installed 2 sipXecs server, each server in their own domain.
2. Each server use the same dial-plan (I set up ext. 200 - 201
Thanks for the good explanation.
I tried the first situation with real user.
First i changed in sipxbridge "inbound calls destionation"
mysipxu...@mysipdomain.tld and then when i did inbound call, i got the
call in my softphone. And i was hearing the voice from calling.
I tried the other way. I
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