On 8 dec 2010, at 20.50, Michael Picher wrote:
> you do realize that odd numbers in second digit indicate development builds,
> right?
I most certainly do, but still, since this was something that seemed to work in
previous versions, I was trying to figure
out why it stopped or what I'm doing w
Thanks to all who replied. Problem is solved.
Deleted my custom dialing plan, and went back to the Local and Long Distance
dialing plans that are built in.
No firewall changes were made. I believe it was 100% voipvoip at this point.
Basically I set up the new trunk to voip.ms and went bac
When sipx sends to voip.ms, it does so on port 5060. When you register with
voip.ms you register on port 5080, and they send inbound calls to port 5080
(following the registration). You can view the ip:port your registered from
at their portal. if ti does not show the public ip of your firewall and
Doesn't apply to our setup...we're not using pix firewalls, and that command
set has been deprecated--like 3 years ago. There is no inspection of packets
on tcp/udp ports 5060 or 5080 going on.
I would expect some log file somewhere to be throwing some message that
indicates what's going.
N
On 12/8/10 5:26 PM, Tony Graziano wrote:
i think you still have to (on pix)
no fixup protocol sip 5060
no fixup protocol sip udp 5060
what he said:
because sipx wants to 'fixup' the protocol.
also, on voip.ms, LIE TO THEM. tell them you ARE NOT BEHIND NATTED
DEVICE (sipx wants to handle th
yes, page 4. also ports 5222/5269 (xmpp user/server) if you need that.
http://www.myitdepartment.net/support/Three_things_I_really_like_about_sipXecs_4.pdf
i think you still have to (on pix)
no fixup protocol sip 5060
no fixup protocol sip udp 5060
what firewall you running?
On Wed, Dec 8, 2010
On 12/8/10 5:07 PM, Victor Williams wrote:
ports/services inbound to that public ip address, of which there are
none defined currently.
then you need udp 3-31000, udp 5060, 5080, tcp 5060, 5080.. and
maybe 8443 and 80 if you want external users to access the gui.
--
Michael Scheidell, C
Switched over to voip.ms and it's the same issue. Disabled the previous
gateway and created another SIP trunk with the VOIP.MS template.
On on the voip.ms portal it shows my account registered and at which pop, and
from which public ip address. When I try calling my DID,
works...bi-direction
1-to-1 nat in the Cisco world is not the same in the rest of the world. 1-to-1
nat just means that only the SipXecs server can use the specific public IP
address to talk to the public internet. You still have to establish access
rules that allow specific ports/services inbound to that public i
On Wed, Dec 8, 2010 at 4:30 PM, Joe Micciche wrote:
>> My blackberry doesn't do that (trim). I can delete all or nothing.
>>
>> Tony Graziano, Manager
>> Telephone: 434.984.8430
>> Fax: 434.984.8431
>
> So the other x-hundred or thousand of us have to just deal with it
I need a new one and a new blackberry server.
I don't know what any of this has to do with cdr database errors.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
T
On Wed, 8 Dec 2010, Tony Graziano wrote:
> My blackberry doesn't do that (trim). I can delete all or nothing.
curious as mine does
mark, copy, delete all, and paste
just fine
- R
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> My blackberry doesn't do that (trim). I can delete all or nothing.
>
> Tony Graziano, Manager
> Telephone: 434.984.8430
> Fax: 434.984.8431
So the other x-hundred or thousand of us have to just deal with it? I'm
finding
1:1 nat leaves you pretty open though. Its always recommended to punch the
ports needed through.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.98
I was just about to ask what the "recommended" ITSP was here. I'll try them
and report back. Will hopefully know something here in the next couple hours.
FWIW, I have our Cisco firewall set to not block from this server outgoing on
any tcp or udp port, and have port randomization turned off.
After reading the voipvoip.com site, I'm not sure what format they are after
to send calls as relates to pai.
One can assume you are dialing/sending 1+10 digits now. Perhaps asking
voipvoip what they "don't like" about the call is in order?
Tony Graziano, Manager
Telep
On 12/8/10 4:10 PM, Victor Williams wrote:
Followed this, same result...fast busy. Also get no additional
activity in the sipxbridge.log file after your suggested changes.
try a known itsp, like www.voip.ms.
($25 deposit.. 1.49C for first did, 1.1c/min plus .99c per month)
if they work, its
Followed this, same result...fast busy. Also get no additional activity in the
sipxbridge.log file after your suggested changes.
--- On Wed, 12/8/10, Tony Graziano wrote:
From: Tony Graziano
Subject: Re: [sipx-users] Can't place outgoing calls - fast busy
To: "Discussion list for users of s
they may be sending the calls in on port 5060, which means the call is not
anchored. It also means transfers might not function, etc.
No initial "9" is needed.
Delete your custom rules. Disable the default local, long distance and toll
free rules. Create a custom rule
1+10 digits, send entire nu
Well, we can receive calls without issue. The only problem is dialing out.
Why would the 5080 issue come up if we're receiving calls fine?
I've removed all dial plans from the trunk and created another that requires 11
digits be passed after the initial 9. Same thing happens...fast busy.
My blackberry doesn't do that (trim). I can delete all or nothing.
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
Help
I don't see they support registering on a port other than 5060, which is a
problem. You should ask if they support you registering on port 5080 and
sending inbound calls to you on port 5080.
They want "11" digits sent to them, a "9" is not needed.
So yes, create a new dial plan accordingly.
On W
On Wed, 8 Dec 2010, Tony Graziano wrote:
> I think it can be done, but realize in Q2 there might be real openacd
> functionality in sipxecs.
>
> Openacd is a skills based acd system, and utilizes freeswitch. Its very
> scalable and highly functional too.
Is there a reason to carry over 350 lines
The ITSP is VoipVoip.com.
When you state "rule", what does that mean? Dial plan?
--- On Wed, 12/8/10, Tony Graziano wrote:
From: Tony Graziano
Subject: Re: [sipx-users] Can't place outgoing calls - fast busy
To: "Discussion list for users of sipXecs software"
Date: Wednesday, December 8,
I would disable that rule and replace it with a custom rule that send
"exactly" the number of digits to the itsp that they need. You might use "9"
if you are using an analog gateway programmed to accept it.
What format does your itsp want you to send? Who is the ITSP?
On Wed, Dec 8, 2010 at 2:21
Staffan,
you do realize that odd numbers in second digit indicate development builds,
right?
On Wed, Dec 8, 2010 at 10:58 AM, Staffan Kerker wrote:
> There is actually nothing written in the sipxconfig.log during the import
> procedure. Debug-level is set to 'debug'.
>
> //Staffan
>
>
> On 8 dec
Looking for a little direction from some more experienced telephony
people...this is my first dealings with setting up any type of phone system in
probably 10 years...
I've somewhat successfully set up our installation (4.2.1 / latest stable) and
can receive incoming calls. We set up with voi
On Wed, Dec 8, 2010 at 9:45 AM, Henry Dogger wrote:
> Unlike stated in issue: http://track.sipfoundry.org/browse/XX-8476 that a
> issue around these errors is fixed.
>
> I still see these lines in the logging: ERROR: duplicate key violates unique
> constraint "cdrs_call_id_unique"
>
> I’m using: s
There is actually nothing written in the sipxconfig.log during the import
procedure. Debug-level is set to 'debug'.
//Staffan
On 8 dec 2010, at 15.59, Tony Graziano wrote:
> Can you provide a log snippet from sipxconfig.log in the failed attempt?
--
Staffan Kerker
mail/sip/xmpp: staf...@kerk
(if you remember most of Mike's postings, they usually have to deal with a
particular ITSP, sounds like he's made some headway though!)
On Wed, Dec 8, 2010 at 10:19 AM, Nikolay Kondratyev wrote:
> LOL!
>
> Rgds,
> Nikolay.
>
> --
> *From:* sipx-users-boun...@list.si
On Wed, Dec 8, 2010 at 4:44 PM, Tony Graziano
wrote:
> I think it can be done, but realize in Q2 there might be real openacd
> functionality in sipxecs.
>
Right, last iteration we started OpenACD integration - to get a first
impression see http://wiki.sipfoundry.org/display/sipXecs/OpenACD+Setup
Can you provide a log snippet from sipxconfig.log in the failed attempt?
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
I'm running SipXecs snapshot (0.0.4.5.1-4199b40 2010-12-01T23:55:40 junco) and
I'm not able to upload any
CA certificates using the web gui anymore. I've tried with different root-CA
certificates from different issuers but the gui always
gives me "Certificate XYZ.crt" is not valid.
When I try t
Great news! Thanks for reporting this :)
I will take this into account.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony
Graziano
Sent: woensdag 8 december 2010 15:45
To: sipx-users@list.sipfoundry.org
Subject
Hi all,
Unlike stated in issue: http://track.sipfoundry.org/browse/XX-8476 that a issue
around these errors is fixed.
I still see these lines in the logging: ERROR: duplicate key violates unique
constraint "cdrs_call_id_unique"
I’m using: sipXconfig (4.2.1-018971.dhubler 2010-08-21T04:59:23
I think it can be done, but realize in Q2 there might be real openacd
functionality in sipxecs.
Openacd is a skills based acd system, and utilizes freeswitch. Its very
scalable and highly functional too.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Not a very comforting thought but thanks for the tip :)
We are also looking into a possibility to use the Queue functionality of
Asterisk 1.8 in combination with SipXecs, since sipXecs is better
scalable.
Do you know if this has been tried before?
Setup now is Asterisk 1.8 with SIP trunk to sipXbri
Again, sipxproc is your friend...
sipxproc --state
Is anything failed? What is the log level of cdr?
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone
I've never embraced the acd as it is, rather I avoid it completely. There
are technical limitations with it that really make it unuseable for a lot of
people.
When in doubt, remove acd queue's, remove role from server, add to other
server and recreate acd queue's.
Tony
I restarted the sipxconfig like you said, but still getting the same
errors in logging.
Getting CDR records seems to work now, but the errors tell me something
is wrong.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behal
I'm testing the ACD functionality, so not using it intensively but encountered
these errors in the logging.
Due to testing, my log levels are on debug level :)
But you say these errors can be ignored and are not due to moving ACD from
primary server to a secondary server?
-Original Message--
Not really any indication how you are using ACD (or not). If you are
not using ACD it is best to disable the role and remove the related
cron job (or deactivate it). If you are using ACD and having issues,
please report them.
I'm not a huge fan of some of the logging messages, depending on the
log
Ok thanks, I will have to live with it than as long as there is no
upgrade :)
How about the ACD errors? Any ideas?
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony
Graziano
Sent: woensdag 8 december 2010 14:49
I think in 4.2.1 cdr activity causes some cpu issues and makes sipxconfig
unuseable.
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:
Ok thanks,
Is this a common problem?
I also saw errors concerning my ACD, while browsing my log files:
"2010-12-08T09:48:08.219656Z":140:SIP:WARNING:servername:SipSubscribeCli
ent-35:440DA940:sipxacd:"LinePresenceMonitor::handleNotifyMessage
presence processed, contact 'sip:50...@servername:5130
Restart sipxconfig
sipxproc -r ConfigServer
Learn the sipxproc command, it is your friend.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.84
Hi all,
When looking at the Call Detail Records, the sipxconfig ui crashed, and
I see following in the logging:
ash-3.2$ tail -f sipxcallresolver.log
"2010-12-08T12:36:14.330202Z":ERR:Loss of connection to database
# - retrying to connect after
sleep
"2010-12-08T12:36:24.332959Z":ERR:Loss of
Hi
I'm trying to implement a HA sipx system. I managed to solve trunking
with a script runing on router that will basicaly switch port forwarding
between servers based on their availability, but need to get bridge
working on secondary server. I tried to follow manual sipx config but
the files
Hi Kris,
I have the same problem..
If you don't have remote workers, you can set the IP specified by ITSP as
the public ip in the configuration of the server..
It works for me.. But i need also remote workers to work.. So it's not an
option for me..
On Wed, Dec 8, 2010 at 2:52 AM, Kris Amy wrot
Hi Paul..
I tried your idea.. but it didn't helped.. :(
to make it more clear.. here is my voip network scheme..
http://img573.imageshack.us/img573/999/myscheme.jpg
so what happens..
the local ip of my server is 10.4.62.2.. the public ip from the router is
89.202.222.55 (dynamic)
when the serve
Am 08.12.2010 01:13, schrieb m...@grounded.net:
> Seems every link I find for repo's on sipx are dead.
> Can someone tell me either where I can get the repo info or just post it.
>
> I would like to update my test system to start becoming familiar with the new
> features coming, including incoming
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