Hi from now a day anyone successfully using hylafax in sipXecs?
how about "Faxvoip"?
Tony Graziano wrote:
The t.38 support for hylafax is highly experimental. I don't know of anyone actually using it successfully via a siptrunk. The t.38 modem software is a nice idea, but noone has really
On Mon, 2010-03-15 at 18:26 -0500, Andres Jaramillo wrote:
I just upgrade the SIPX SERVER from 3.10 to 4.0.4 But The AudioCodes
still receiving this kind of messages, now in this form
6d:21h:22m:35s SUBSCRIBE sip:6346...@192.168.0.122 SIP/2.0
Record-Route:
Via: SIP/2.0/TCP
Ok, thanks for your help. I configured Option 66 to go to the primary
server. On that server, I configured the Primary Registration Server with
the domain name. That did the trick.
Now we've got another issue. The phones cannot call each other using the
alias. I have 10 digit DIDs for the user
On Tue, Mar 16, 2010 at 10:23 AM, Ken Fulmer
kenful...@icstechnologysolutions.com wrote:
Ok, thanks for your help. I configured Option 66 to go to the primary
server. On that server, I configured the Primary Registration Server with
the domain name. That did the trick.
Now we’ve got
Yes, I've got faxing working fine but it's not on sipx, it's a stand alone
server with hylafax/avantfax.
I have pools of did's on the mediant which are routed to different pbx
machines.
I had found documentation that seemed to imply that the mediant could route
based on incoming fax but it
Would you be kind enough to post how you got hylafax to work? I get
IAXmodem to work but hylafax just doesn't seem to want to do anything.
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 3/16/2010 9:35 AM, m...@grounded.net wrote:
Yes, I've got faxing
The simplest way is to start with a pbx.
For example, set up a stand alone Elastix box, don't enable any of the extra
things, just use the asterisk portion.
Once you've got the pbx working with at least one extension, you'll know it's
ready for the next step which is faxing.
# Install
Can the call forward busy and call forward no answer settings be change
on the SipX??
I know that it can be done on the phones themselves but with server
based call forwarding, I much prefer to do it on the server. I done some
search but everything always points to the call forwarding but
Totally agree!
Andrew
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
Goran Donev
Sent: Monday, March 15, 2010 11:43 PM
To: 'Todd Hodgen'
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users]
We've been talking about internal extensions as our default dial plan. I'll
convert the phones and see what happens.
Thanks,
Ken
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Tuesday, March 16, 2010 9:31 AM
To: Ken Fulmer
Cc: sipx-users@list.sipfoundry.org
Subject:
On Tue, Mar 16, 2010 at 11:53 AM, Ken Fulmer
kenful...@icstechnologysolutions.com wrote:
We’ve been talking about internal extensions as our default dial plan.
I’ll convert the phones and see what happens.
Whatever you do make sure your internal dialplan matches. half solutions
dont work
I was hoping you would have some insight on setting up hylafax manually.
as in no GUI.
I do all my asterisk work without GUI (text files only).
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 3/16/2010 10:06 AM, m...@grounded.net wrote:
The simplest way
The setup is all from CLI but once installed, you could do your configs from
CLI or GUI is my guess since it's all in the config files of course.
On Tue, 16 Mar 2010 11:06:06 -0500, Josh Patten wrote:
I was hoping you would have some insight on setting up hylafax manually. as
in no GUI.
I
/sipXregistry/doc/service-tokens.txt was where it was kept. Don't know now.
It's now in meta/system-sip-identities.
From the SIP point of view, you subscribe to sip:[extensi...@[domain] for
dialog events to find the status of user [extension].
You can also subscribe to
Hello all,
It's possible to play MoH (or other .wav file) to the caller while the call
ring on the hunt group phones?
Thanks,
Bruno
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On Tue, Mar 16, 2010 at 1:59 PM, Bruno Macedo bruno.mac...@identity.ptwrote:
Hello all,
It's possible to play MoH (or other .wav file) to the caller while the call
ring on the hunt group phones?
Thanks,
Bruno
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On Tue, Mar 16, 2010 at 1:52 PM, Jeff Gilmore j...@thegilmores.net wrote:
On my server running 4.0.4 ISO, which as been stable for several months, I
suddenly could not call offsite (through my trunks on sipxBridge). Looking
in the sipxbridge log gave me this message:
You may have hit on a port that was being used by some other process.
What does sipxrelay.log look like?
Ranga
In the time frame of this error I see:
2010-03-16T17:18:20.273000Z:
5289:JAVA:INFO:choicevoip.ev.ithaca.ny.us:SslListe
ner0-3::symmitron:setDestination : controllerHande
I'm using polycom soundpoint ip 450's and most of my calling features are
working properly.
I have a setup with audiocodes analog lines inbound and broadvox outbound.
If I try to conference via the polycom phone I can call the first leg and call
the second leg, but when I press conference. It
On Tue, Mar 16, 2010 at 4:16 PM, Lara Johnson lcr...@ciscorp.biz wrote:
I’m using polycom soundpoint ip 450’s and most of my calling features are
working properly.
I have a setup with audiocodes analog lines inbound and broadvox outbound.
If I try to conference via the polycom phone I can
On Tue, Mar 16, 2010 at 4:16 PM, Lara Johnson lcr...@ciscorp.biz wrote:
I’m using polycom soundpoint ip 450’s and most of my calling features are
working properly.
I have a setup with audiocodes analog lines inbound and broadvox outbound.
If I try to conference via the polycom phone I can
Running sipx 4.0.4
Bootrom 4.1.2
Sipversion 3.1
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Tuesday, March 16, 2010 4:23 PM
To: Lara Johnson
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] One sided conference
On Tue, Mar 16, 2010 at 4:16 PM, Lara Johnson
On Tue, Mar 16, 2010 at 4:26 PM, Lara Johnson lcr...@ciscorp.biz wrote:
Running sipx 4.0.4
Bootrom 4.1.2
Sipversion 3.1
3.1.3RevC? If so, all good.
This is just the conferencing function to join two calls on the phone, it
should work. A call trace might provide useful since the
I have a remote user that uses a VPN connection to connect to the courthouse.
They are using x-lite to make calls. I recently changed their dial-plan to
route external calls via an ITSP. The call connects correctly, but after 30
seconds, the call disconnects.
As a temporary work around, I'm
It is running 3.1.3RC
I've got a similar implementation with that firmware and it does seem to work
fine there.
The ingate has all sip signaling set to go to the SipXecs, the audiocodes is
set to the sipxecs as the proxy but has the normal internet gateway as it's
default gateway.
I can make
I have a remote user that uses a VPN connection to connect to
the courthouse. They are using x-lite to make calls. I
recently changed their dial-plan to route external calls via
an ITSP. The call connects correctly, but after 30 seconds,
the call disconnects.
As a temporary
I'm using polycom soundpoint ip 450's and most of my calling
features are working properly.
I have a setup with audiocodes analog lines inbound and
broadvox outbound.
If I try to conference via the polycom phone I can call the
first leg and call the second leg, but when I press
All good questions...
I'm slightly embarrassed to mention this...however I've just realized that the
ITSP that I thought I've been routing thru, is actually not the one I've been
routing thru. On that note, I'm going to make some configuration changes and
re-test. I've inadvertantly been
Hi.
Have some questions regarding sipx.
1. Can it be configured for multi-tenant? Right now we provision one server
for every client, and are thinking we can consolidate some of them that don't
use much BW and resources.
2. Is there a billing engine that can integrate with sipx? I think
On Tue, Mar 16, 2010 at 4:00 PM, Jeff Gilmore j...@thegilmores.net wrote:
You may have hit on a port that was being used by some other process.
What does sipxrelay.log look like?
Ranga
In the time frame of this error I see:
It appears that polycom only situations work as expected.
Although a external polycom polycom will exhibit the same issues (where you can
only talk to the lg of the conference that you were on when you hit
conference). So if you're a polycom call polycom and try to conference in
external, when
Can you try enabling Single key press conference under Call Handling
then send profiles and try again?
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 3/16/2010 3:58 PM, Lara Johnson wrote:
It appears that polycom only situations work as expected.
Okay - I've now officially tested the remote thru the sip
trunk I wanted to use. The same scenario is happening.
I placed a call from X-Lite on my computer (inside the
courthouse) with no issues - so I'm assuming there is
something odd with the VPN...
Appears to be so. Start by
On Tue, Mar 16, 2010 at 4:56 PM, Nathaniel Watkins
nwatk...@garrettcounty.org wrote:
Okay - I've now officially tested the remote thru the sip trunk I wanted to
use. The same scenario is happening.
I placed a call from X-Lite on my computer (inside the courthouse) with no
issues - so I'm
Okay - I'll start there. Thanks!
-Original Message-
From: JOLY, ROBERT (ROBERT) [mailto:rj...@avaya.com]
Sent: Tuesday, March 16, 2010 5:04 PM
To: Nathaniel Watkins; sipx-users@list.sipfoundry.org
Subject: RE: [sipx-users] issues from remote caller vpn into office - using sip
trunks
It appears that polycom only situations work as expected.
Although a external polycom polycom will exhibit the same
issues (where you can only talk to the lg of the conference
that you were on when you hit conference). So if you're a
polycom call polycom and try to conference in
1. no and as far as I know there is no plans for this.
2. Billing as in cut the system off when they run out of money, or just
keep track of it and send them a bill? You could probably write
something that pulls the CDR records out of the DB and bill that way.
You could probably also have
What are the SIP standards for providing caller-id to an itsp? What does sipx
use? I'm attempting to test with freepbx (which I think is basically
bandwidth.com - without good support) and my outbound calls just show Unknown
Number...I'm guessing I'm SOL?
It very much depends on the carrier.
I can send whatever reasonably formatted caller id to bandwidth.com with
sipx using sipxbridge or an ingate.
If I use a pri I can send it to centurylink IF they allow me to send it, but
I am not able to send ANY caller I'd to that provider.. You need to query
I'm under the assumption that FreePBX would act similar to bandwidth.com - the
other IP Trunks I've played with are allowing me to send my actual DID. I've
got an email into their tech support asking if they can help...I'm not going to
hold my breath...
-Original Message-
From: Tony
Dunno freepbx. If its free maybe it aint that good. Chuckle.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
Fax: 434.984.8427
Helpdesk
wait a minute...
I've always had two opinions... (Opinion #1) The guy who created the nagios
config file formate must have had the same father as the guy responsible for
the * configuration stuff. Both are like inbred cousins. I can't understand
that stuff. (Opinion #2) I'd like to take that guy
Bandwidth.com pointed me in their direction - it's not free ($25/month) - but
it doesn't require a 12 month contract. The bill on my credit card actually
says bandwidth.com. I think it is just their solution for the open source
community...
At any rate - if I don't get anything useful from
They also (bandwidth.com) have a demo trunk, which you can use with sipx
to test with or consider it a monthly thing as I recall.
On Tue, Mar 16, 2010 at 6:06 PM, Nathaniel Watkins
nwatk...@garrettcounty.org wrote:
Bandwidth.com pointed me in their direction – it’s not free ($25/month) –
but
Does anyone know where the branding/skinning of sipxConfig is described.
I have read the page
http://sipx-wiki.calivia.com/index.php/SipX_ConfigServer_Customize_Color
s,_Layout_and_Logo
which gives some good hints, but I am looking for something that
explains the CSS files.
Regards,
Sven
I'll have to smack my bandwidth.com reps around a little bit - that's what I
asked for - they pushed me off to these other guys
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Tuesday, March 16, 2010 6:20 PM
To: Nathaniel Watkins
Cc: sipx-users@list.sipfoundry.org
Subject:
Problem: calls to a specific number in rural MN (507-375-) never ring when
using voip.ms via a sipxecs trunk though voip.ms.
What I expect: outbound call to ring and be answered
What happens: called number doesn't ring, 60 seconds later call is terminated
by voip.ms
Environment: Polycom 650,
Nope. You got it. The interconnect fees in some places are horrible. I have
that same problem with customers calling MN also. Its not just voip.ms. The
clec's and itsp's underbuild because of outrageous costs.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax:
FWIW I just called the problem number on a POTS line (from MN, about 100 miles
away in Qwest territory) and it had *terrible* voip style lag and call quality.
I'm guessing you're right, the CLEC has some capacity problems.
The customer at that end told me their mom+pop has changed hands 4 times
And now centurylink. Embarq was sprint was sprint-centel and was centel and
was central telephone. They merged with century link to tey and survive. Try
6 years of same or less income and higher costs to keep your lec status and
it happens. As I happen to be in former embarq territory.
___
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of John
[jcl...@sbsproactive.com]
Finally I set up an account at Vitelity and within 10
minutes I had a functioning phone system. I think the
problem may just
sorry for reply late, because i from malaysia.
Means you done in hylafax(astrisk)?but i never touch this hylafax and
astrisk before
Can i have some guide for this?
Since last month i replace all of the normal analog line to PRI (under
m1k gateway), and now i pure use sipxecs to my phone
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