There are three steps that you will need to do for this -
Under Devices, SBC, select your sipxbridge, select the SIP configuration,
and remove the Operator under incoming call duration.
Next - add the alias under the user you want (sounds like you have done
this.
Next - go to System, Dial Plans,
What did you set the alias as?
James Johnson wrote:
> I have a system set up with an auto attendant to take the calls, but I want
> to set up a DID so that when a person calls that number it goes directly to
> that persons extension and bypasses the main auto attendant.
>
> I have added the use
What you're asking for will be somewhat available in 4.2 with IMAP
synchronization. When a users voicemail is marked "read" on the IMAP
server then it is marked read on the voicemail server.
Max Clark wrote:
> Hello,
>
> Is it possible to mark a voicemail as read or delete it after it is
> email
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 receive this number (170089XXX) , 1st i will Stripped
9 Dig
Not at present. In 4.2 there is an IMAP integration option. If the user
deletes the email, it also deletes from sipxconfig.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems
Hello,
Is it possible to mark a voicemail as read or delete it after it is
emailed to the user?
Thanks,
Max
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You are probably using sipXBridge, the default setting there is that all
incoming calls go to "operator" (who thought that was a good idea!?).
Just remove that from the SBC and off you go.
Sven
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list
I have a system set up with an auto attendant to take the calls, but I want to
set up a DID so that when a person calls that number it goes directly to that
persons extension and bypasses the main auto attendant.
I have added the users extension
Added the Alias under that user
Added the gatew
John,
Good news. Thanks Tony -- good memory :-)
jim
> Tony's suggestion led me down the path of recreating the certificate
> (again!), but this time, I entered foo.bar.com as the CA, but left
> everything else default or 'standard' entry for OU name, etc. And it
> works. VM is back.
>
>
>
Tony,
Yes, 3.6.0 still exists..
caName = ca.sip.geomagic.com
sipDomainName = sip.geomagic.com
The server name, however, is intrepid.sip.geomagic.com.
Thanks!
--
John,
Could you regenerate the certificate making sure to follow Tony's
instructio
Crap that's old!
did you do it like this?
mkdir $HOME/sslkeys
cd $HOME/sslkeys
/usr/bin/ssl-cert/gen-ssl-keys.sh
where "CA Common Name" is anything BUT NOT the DNS name of your server, "SIP
domain name" is the domain name of your installation, and "Full DNS name for
the server" is the fully qual
since fsense can do its own pcap file from the webgui, why don't you create
one on a call with failed audio so this will identify "why' the audio is
failing.
On Mon, Feb 22, 2010 at 5:42 PM, Tony Graziano wrote:
> Your instance is different in that you have multiple public IP's.
>
> Specifically
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 below) and no
registrations are showing in the GUI -- even tho
Your instance is different in that you have multiple public IP's.
Specifically I do not know if your phone is on a different subnet than your
DATA network.
With mulitiple public IP's, I would suspect your firewall LAN rule would
have to include one that pointed the all the same protocols/ports to
Robert,
Why this last reset of the ATA made a difference I do not yet know -
but as of now it is working as we would expect.
Thank you for your time and help.
-Max
On Mon, Feb 22, 2010 at 1:15 PM, Robert Joly wrote:
>> Here's the registration for a Phone and the ATA. I'm going to
>> peak in th
What do you mean by source rules? Are you referring to the LAN side of
firewall rules? I have the default setting of allowing Lan subnet access to
any destination. I checked this against the xml you have in your blog and I
think I have the same Lan rules. Also, I assume that the incoming cal
It should have reformatted the drive, only you can confirm that though. If
it did reformat the drive, the error message would make one think there is
something missing from the build itself as it relates to the "branch"
function for the postgres db.
On Mon, Feb 22, 2010 at 5:01 PM, Pizza Napoletan
Doesn't the ISO install wipe out everything from before?
On Feb 22, 2010, at 1:56 PM, Huijun Yang wrote:
> The sql error indicates that your system has old database data prior to
> 4.1.6 install. Run upgrade should solve your problem.
>
> Huijun
>
> -Original Message-
> From: sipx-user
Yes, saw that too. I saw "both' were installed (the prior one). I didn't
raise an issue because this build is probably outdated. I deleted the old
version manually. Will keep an eye on it as new builds come out though.
On Mon, Feb 22, 2010 at 4:52 PM, Pizza Napoletana wrote:
> Thanks.
>
> Yes, I
Thanks.
Yes, I did an in-place upgrade on another machine last night and that went OK.
The ISO install on another test machine is where I am having this issue.
I should probably install the old version and do an in-place upgrade to get
around this.
btw, if anyone wants to look into it...
The in-
Looks like a sql issue with branch:
one.xxx.yyy.com:main::JDBCExceptionReporter
:"ERROR: relation \"branch\" does not exist"
"2010-02
I didn't have thhis same issue with an inplace upgrade but have not
installed a 4.1.8 iso yet.
Tony Graziano, Manager
Telephon
Yes, a couple of times.
Thanks
On Feb 22, 2010, at 1:22 PM, Tony Graziano wrote:
> Can one assume you rebooted after the install?
>
> On Mon, Feb 22, 2010 at 4:00 PM, Pizza Napoletana wrote:
> Has anyone successfully run the 4.1.6.018058 ISO?
>
> After install, when I visit the web GUI, I get:
Can one assume you rebooted after the install?
On Mon, Feb 22, 2010 at 4:00 PM, Pizza Napoletana wrote:
> Has anyone successfully run the 4.1.6.018058 ISO?
>
> After install, when I visit the web GUI, I get:
> HTTP ERROR: 404
>
> /sipxconfig Not Found
>
> RequestURI=/sipxconfig
>
> *Powered by J
> Here's the registration for a Phone and the ATA. I'm going to
> peak in the Linksys configuration for SIP configuration.
>
> sip:2...@foo.com 2214
So, this device (presumably the phone) , if a remote device, has some kind of
STUN or ALG in the picture.
> sip:2...@foo.com
>
Here's the registration for a Phone and the ATA. I'm going to peak in
the Linksys configuration for SIP configuration.
sip:2...@foo.com 2214
sip:2...@foo.com
771
Thanks,
Max
On Mon, Feb 22, 2010 at 1:05 PM, Robert Joly wrote:
>> Robert,
>>
>> I removed the STUN configu
> Robert,
>
> I removed the STUN configuration from the ATA and reset the device.
> The behavior is the same.
What about the Linksys router, any SIP ALG lurking in there? Can you go in
sipXconfig under diagnostics->Registrations and tell me what the registration
contact looks like for your ATA
Has anyone successfully run the 4.1.6.018058 ISO?
After install, when I visit the web GUI, I get:
HTTP ERROR: 404
/sipxconfig Not Found
RequestURI=/sipxconfig
Powered by Jetty://
sipxconfig.log says the following:
"2010-02-22T20:51:11.619000Z":1:JAVA:INFO:pbx.phone.xxx.yyy.com:main::D
Robert,
I removed the STUN configuration from the ATA and reset the device.
The behavior is the same.
-Max
On Mon, Feb 22, 2010 at 12:36 PM, Robert Joly wrote:
>> Hello,
>>
>> I have a sipx server running on a statically assigned public
>> ip address. I need to support far end NAT for remote us
Internet Calling is disabled on the sipx server.
The IP address/subnet for the NAT is unique and not listed in the
intranet subnets (I deleted the defaults for good measure).
No change in behavior.
-Max
On Mon, Feb 22, 2010 at 12:32 PM, Tony Graziano
wrote:
> ensure internet calling is disabled
> ensure internet calling is disabled at the sipx server.
> ensure the Ip address/subnet is unique and not listed in your
> intranet subnets in sipx.
Not that it has anything to do with the problem being reported here but
remote private networks and the local private network (the a private
sipXe
> Hello,
>
> I have a sipx server running on a statically assigned public
> ip address. I need to support far end NAT for remote users
> (ATAs behind a home firewall). I have a Grandstream HT 286
> that we are experimenting with. Initially I connected the ATA
> to our office DHCP network, auto
ensure internet calling is disabled at the sipx server. ensure the Ip
address/subnet is unique and not listed in your intranet subnets in sipx.
On Mon, Feb 22, 2010 at 3:30 PM, Max Clark wrote:
> Hello,
>
> I have a sipx server running on a statically assigned public ip
> address. I need to supp
Hello,
I have a sipx server running on a statically assigned public ip
address. I need to support far end NAT for remote users (ATAs behind a
home firewall). I have a Grandstream HT 286 that we are experimenting
with. Initially I connected the ATA to our office DHCP network,
autoconfigured the dev
Looks like the outbound caller id on the conference 'invite' is the
extension number of the conference, not the extention number and not the
caller id of the user who owns the conference.
since a conference does NOT need to be owned by any user, I suggest that
the conference needs a caller id,
I really hope the sipX team never implements a CoS matrix. I remember
demoing an Alcatel OmniPCX and telling them to come pick that garbage up
because the thing was practically impossible to configure without
extensive training classes. The CoS matrix was a nightmare and nothing
made sense. It
On 2/22/10 9:59 AM, Josh Patten wrote:
http://track.sipfoundry.org/browse/XX-7704
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
PBX's that I used in the past always had a class of service (permission)
bit: Precede to be call forwarded externally.
(ther
http://track.sipfoundry.org/browse/XX-7704
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 2/22/2010 6:58 AM, Scott Lawrence wrote:
> On Sun, 2010-02-21 at 09:57 -0600, Josh Patten wrote:
>
>> We're talking about users that barely know how to check their
Because you are using multiple public ip's, you need to creat source rules
too.
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: 434.9
May work in some cases but not in ours, we already have account codes
through our LD provider and the toll fraud employees just use them to
commit the fraud.
I wish it was as simple as "fire that theif" but unfortunately that's
not the case.
Josh Patten
Assistant Network Administrator
Brazos C
I reinstalled sipx since I replaced our subnet and firewall from endian to
pfsense. With pfsense, I cannot get any incoming calls. From my cell, I don't
hear anything when I try to dial my incoming DID number. Not even the operator
IVR.
I have setup Pfsense with my public IPs as virtual Prox
Here's a somewhat related idea that may be a better solution... and I
believe that a feature request is already in the tracker... what about
implementing accounting codes for all "toll " calls?
I've worked in shops where some or all calls required a billing code
before being allowed to proceed
Good day,
The SIPX version we have is 4.0.2.
We are having some issues with the CDR records. Successful inbound/outbound
PSTN calls show up as failed in the cdr historic report.
The call does not show up under active calls during the call. It is placed in
the historic table as "failed" with ze
On Sun, 2010-02-21 at 09:57 -0600, Josh Patten wrote:
> We're talking about users that barely know how to check their email
> here. The personal auto attendant concept gets me blank stares every
> time I talk about it to end users. If they're savvy enough to use that
> then more power to them :-P
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