Hi again. We have installed sipxecs-4.0.4-017289 on Centos 5.
This morning i saw a very strange records in my sipregister logs.
It looks that somebody is trying (or it registered successfully)
register and make calls through our system with one of our extension.
I checked on Call Details Records
Hi,
from time to time some of our hardware phones loose connection to
sipxecs. Sometime when a call is hanged up, sometimes it just stop to
register for a hour or more. I cannot see errors in sipregister log, but
i saw today this in sipxsupervisor.log.
I am not sure if these errors have some
that looks like a proxy log. the call is being initiated from a user line,
but the user is xxx'd out by you.
Since the call is using TCP, my guess is that it is a remote user or a user
with softphone, and the user has been hacked.
Since the user has to pass through the proxy credentials in order
Thanks for reply and good explanation.
On Tue, 2010-02-23 at 05:07 -0500, Tony Graziano wrote:
that looks like a proxy log. the call is being initiated from a user
line, but the user is xxx'd out by you.
Since the call is using TCP, my guess is that it is a remote user or a
user with
On Mon, 2010-02-22 at 14:45 -0800, jnolen wrote:
Greetings,
Customer Geomagic is running sipxpbx 3.6.0. The ssl certificate expired
and voice mail cannot be reached.
A new self-signed certificate was generated and the system passes a
configtest, however there are Java errors (see
On Tue, 2010-02-23 at 11:11 +0800, Winson (Elabram) wrote:
Hi, I am experiencing some call transfer issues when using the E1
gateway m1000 pass a call to SipXecs
It can auto pass to my Extension DID (1303)
Example : outside person (0127788328) call this number (170089XXX)
when my gateway
Currently sipx does not support multiple ITSP accounts with the same
domain too well. i.e. for outbound dialling there is not enough
information to pick the specific ITSP account to use. In trying to
address this limitation this we defined a line ID in issue xx-4785 to
be associated with an a
I do not immediately see this need, but my need is very similar.
Our sipX in UK has SIP trunk to UK (no problem), but also to other
countries. When calls arrive at the local break outs in those countries,
the caller ID that arrives from the ITSP to sipX, does not have country
code. The ITSPs are
I could see this in an instance where there are several offices in dispersed
geographic regions, even different countries.
While all three might use the same ITSP, they might also have their own
account. I would imagine that each office could set the other accounts up in
their system to handle
On Tue, Feb 23, 2010 at 9:58 AM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
I could see this in an instance where there are several offices in dispersed
geographic regions, even different countries.
While all three might use the same ITSP, they might also have their own
account. I would
On 2/19/10 1:39 PM, Michael Scheidell wrote:
*76 for intercom
*77(extension) for paging groups
*78 (extension) for directed call pickup
*4(extension) for call park
anyone with a list of them?
accidentally found two more.
*88 presence sign in
*86 presence sign out
--
Michael Scheidell, CTO
On Tue, Feb 23, 2010 at 10:03 AM, M. Ranganathan mra...@gmail.com wrote:
On Tue, Feb 23, 2010 at 9:58 AM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
I could see this in an instance where there are several offices in
dispersed
geographic regions, even different countries.
While all
Hello,
I use sipXecs together with Polycom Sound Point IP670 Phones. Provisioning
worked flawlessly so far.
Recently we added a LDAP Directory Server to our network, so that the
Polycom phones can access the contacts.
In order to do that I had to edit (enable feature, enter LDAP information,
For custom configuration follow:
http://wiki.sipfoundry.org/display/xecsuser/Polycom+Phone+Customization
edit */etc/sipxpbx/polycom/polycom_sip.cfg* , add the config info in,
and your change will be applied to all phones that have the feature.
Josh Patten
Assistant Network Administrator
Almost forgot, you have to send profiles to the phones once you make
that change.
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 2/23/2010 10:03 AM, Josh Patten wrote:
For custom configuration follow:
ok i've ran packet capture.
below are the details:
12:09:50.969517 IP 66.xx.xx.7.5080 67.216.35.162.5060: UDP, length 564
12:09:51.972186 IP 66.xx.xx.7.5080 67.216.35.162.5060: UDP, length 564
12:09:53.990353 IP 66.xx.xx.7.5080 67.216.35.162.5060: UDP, length 564
I don't understand why the
I am looking to install sipx on a machine that has two nics. One nic
is facing wan. The other nic is facing lan. I would like to enable
connections to sipx from both networks. Is this feasible? If yes, is
there a walkthrough on how to set it up?
Thanks in advance
No. Don't even try it.
sipX is a one-NIC kind of software.
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 2/23/2010 11:33 AM, Roman Gelfand wrote:
I am looking to install sipx on a machine that has two nics. One nic
is facing wan. The other nic is
I have an interesting question; Can I create an extension that (currently) has
no phone attached to it, but allow calls placed to that extension to be
forwarded to an offsite number? I tried it, and the call instead goes to
voicemail.
I have a neighborhood phone service where I make
It should work unless your phantom user does not have the necessary
calling privileges to make that call. It shouldn't matter if voicemail is
enabled or not, but if noone will check it why enable it also?
Tony Graziano, Manager
Telephone: 434.984.8430
Fax:
Hi there, can anyone recommend an ATA similar to the Audio Codes MediaPack
118 analog VoIP gateway, 4FXS, 4FXO that has built in NAT support? Key is
NAT support.
Bob
___
sipx-users mailing list sipx-users@list.sipfoundry.org
List Archive:
It sure would be cool if there was a way to dump the internal state that makes
up the dial plan after its all been set up, its always hit or miss to figure
out what its going to do.
-Eric
On Feb 23, 2010, at 9:10 AM, Michael Scheidell wrote:
On 2/19/10 1:39 PM, Michael Scheidell wrote:
I would think this would be needed for the following scenario:
Assume ½ of all calls are long distance - the other ½ are local
Using Bandwidth.com as an example:
5 unlimited trunks/5 metered trunks
Long distance calls would want to use the unlimited trunks
Local would want to use the metered
On 2/23/10 3:37 PM, Eric Varsanyi wrote:
It sure would be cool if there was a way to dump the internal state
that makes up the dial plan after its all been set up, its always hit
or miss to figure out what its going to do.
kind of adds some excitement to Monday morning after updating the
However, you can accomplish the same thing by having two separate accounts
with Bandwidth.com, each have the same 5 trunks, and set your permissions
for long distance on one, and local on the other.
Seems a configuration change to the system to accommodate that scenario is
better than making a
But 2 accounts with the same domain name is what is being addressed.
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
Two accounts with the same domain name is not an issue, if you can terminate
on different IP addresses rather than the Domain name. I suspect that
Bandwidth.com has many termination points on their network to chose from?
For instance, with Broadvox, they have dozens of termination points to
I am not sure if it has been implemented yet, does the new stable
software allow for incoming call routing based on target phone number?
Also, how stable is unstable version?
Thanks in advance
___
sipx-users mailing list sipx-users@list.sipfoundry.org
There is a branch identifier in 4.1.6 (dev-unstable).
Right now incoming calls are routed by target (dialed) number if your
carrier supports DID.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
On Tue, 2010-02-23 at 16:21 -0500, Roman Gelfand wrote:
Also, how stable is unstable version?
Much of the development team has been using it for a few months now
(with frequent updates). The stability is getting pretty good now in
that the services mostly work reliably. There _are_ still
Any issues with using sipstation for a sip trunk?
Nathaniel
This message and any files transmitted with it are intended only for the
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entity named you are hereby notified that any disclosure, copying, distribution
or
I need a reliable sip trunk that I can test with without a term commitment. I
actually called bandwidth.com and they pointed me to sipstation because it
doesn't require a 1 year commitment.
Once I am satisfied with the quality/etc., I will go over to the 'real deal' so
we can have some
Forgot the reply all.
This seems to be just a reseller of various sip trunks. For example, if you
order from them their Bandwidth.com option, you are essentially ordering
Bandwidth.com, which is definitely supported. However, there are several
users on this forum that sell Bandwidth.com as
When I need to test or prove something I order a test trunk from
bandwidth.com.
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:
I called them and was very clear that I needed a test trunk - they said
buying directly from them required a 1 year commitment. Their sales guy
pointed me to sipstation.
Nathaniel Watkins
IT Director
Garrett County Government
316 East Alder Street, Suite 2
Oakland, MD 21550
Telephone:
If I am not mistaken, in the past, cd installation, amongst others
things, took care of installing and configuring the dns, http, and
sipx server. For some reason, at the end of my installation 4.0.4,
the only thing that was installed was centos 5.2. When I tried to
install bind using yum, I
It sounds like you need to run the installation script, it somehow was
aborted at the end maybe?
/usr/bin/sipxecs-setup-system
And away you should go.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
Thanks again. Will do.
BTW... can I install sipx using source on debian? Is the source
package redhat specific?
On Tue, Feb 23, 2010 at 5:44 PM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
It sounds like you need to run the installation script, it somehow was
aborted at the end maybe?
I don't know that there's a debian source anymore. Of course one can always
build their own...I think back in version prehistoric int was on Gentoo.
Several people have asked about Debian, but maintaining it is a bit of a
problem because noone seems to want to bite the bullet. There are 32 64
I'm testing with voip.ms because everything is pay as you go. Its about 0.01
per minute and 0.99 per DID number/month. I ported a PhonePower number in (for
$25) and that went smoothly. You just prepay your account (I think the minimum
is $10) and it debits until you run out or refill. No
I'm using them as well - I just want to be certain I have the best quality
possible before having other people experiencing the proposed solution.
Nathaniel Watkins
IT Director
Garrett County Government
316 East Alder Street, Suite 2
Oakland, MD 21550
Telephone: 301-334-5001
Fax: 301-334-5021
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