Thanks Tony. It was the ca bundle. I had imported it via the web interface and
it was displayed properly but it was never saved eventhough I clicked on keep.
I had to drop the ca bundle into /etc/sipxpbx/ssl/authorities and run the
ca_rehash script manually. After the service restarted all is
Do you really want this external service sitting between whoever is
chatting and your xmpp server?
This is my issue with things like this and imo.im
I'm just sayin...
Thanks,
Mike
On Mon, Mar 26, 2012 at 5:36 PM, Tony Graziano tgrazi...@myitdepartment.net
wrote:
The widget is tied to your
About the prompt that only play a part, the English directory prompt play
something like Enter the name of the person that you want to reah press 7
for Q and 9 for Z and press star to cancel in french she only say Appuyer
sur l'étoile pour canceler press star to cancel so all the
beginning is
My seting are like that at the moment
Custom Bilingual Main Menu(choose your language) Custom French prompt
for the choice(3 extension + Directory) -- Directory in french missing
a part
Custom Bilingual Main Menu(choose your language) - Custom English
prompt for the
I tried with the solution Dragan gave me but the transfer doesn't work it's
like if I am in a parallel universe there is no hold music nothing.
I created a Dialplan auto attend and gave him the default auto attendant in
English
Then in my Main menu i tried with those input
Hi,
I installed a 4.4 64-bit test system from 287 iso, yum updated it, set
repo to staging and yum updated again.
Sipxproxy is now at release 374.g2acd4. But unfortunately it does not work
(pcap attached). Does this release contain the patch posted here before?
Do I have to set some option in the
Hi Guys,
I have a Sipxecs primary and two redundant servers, when users are
connected to one of the specific HA server all calls made by users are not
logged to CDR table view_cdrs on primary server.
When I go into CDR database of this HA server and check by doing request
below I'm able to see
On Redundant server where I have my problem the Recorded call events are
older than 7 days even if in Sipxecs GUI Purge age for CSEs is set to
7, I have checked on the second redundant server and older CSE are 7 days
older so it looks to be ok on this server, same case on primary it's ok.
So
This is where one swallows one's pride The way I was entering data
caused the drop-down to not be displayed.
To keep this short:
1. When you first select Add new gatewaySip Trunk, the template drop
down is not visible. I was not aware this was the case until
yesterday. I just thought
To keep this shorter the behavior is different in 4.4. Updating would keep
us all from having to use a way back machine.
On Mar 27, 2012 12:05 PM, Stiles Watson wat...@datatek-net.com wrote:
This is where one swallows one's pride The way I was entering data
caused the drop-down to not be
Correct.
On Mar 27, 2012 12:05 PM, Stiles Watson wat...@datatek-net.com wrote:
This is where one swallows one's pride The way I was entering data
caused the drop-down to not be displayed.
To keep this short:
1. When you first select Add new gatewaySip Trunk, the template drop
Sounds like your Nat rules for port 5080 is being sent to your pbx on port
5060.
On Mar 27, 2012 12:05 PM, Stiles Watson wat...@datatek-net.com wrote:
This is where one swallows one's pride The way I was entering data
caused the drop-down to not be displayed.
To keep this short:
1.
On 3/27/2012 12:03 PM, Stiles Watson wrote:
This is where one swallows one's pride The way I was entering data
caused the drop-down to not be displayed.
To keep this short:
1. When you first select Add new gatewaySip Trunk, the template drop
down is not visible. I was not aware this
Hi Guys,
After a postgresql restart calls made from this server are correctly sent
to primary.
Do you know how to restore data from call_state_events table from redundant
server which have not been written to primary server into cdrs table ?
Because my CDR report are wrong for calls made during
I can do that. I thought I saw an email a while back about Polycom
phones and 4.4 and firmware. Still looking for it, but while I search.
Is there an issue with using Polycom 335 rev 3.2.1.0078 with sipx 4.4?
Stiles
On 03/27/2012 12:16 PM, Tony Graziano wrote:
To keep this shorter the
Stiles, are you by chance putting port numbers in fields that are blank?
There are usually some fields that call for a port number that are blank, as
they default to 5060, sometimes people fill those fields out. If you have
changed a field from its default to something different, you will notice
I meant when I make an incoming call, not outgoing. And, yes, I did use
the template this time. Still on 4.2 at this point. Will not be able to
reinstall today.
Also, my SIP Trunk to callwithus is as you say:
Src IP Src Port Dst IP Dst Port Protocol Src Iface Dst Iface Flow
Type IPS
Todd,
No, I've not changed any port info. In the SBC SIP settings Public
Port is empty, and External Port is the default of 5080. Also, in the
voip.ms gateway, the only thing I changed from the default template is
Username, Authentication Username, Password, and ITSP server address
(changed
In the voip.ms template you would put in atlanta in two places.
beyond that, your first description was wrong on the call flow.
how sipx uses ports on itsp calls...
FROM port 5080 (sipx) to ITSP on port 5060 (ITSP) for outbound calls.
FROM port 5060 (ITSP) to sipx on port 5080 (sipx) for calls
Tony, I'm always thankful for the help I get here and I've always tried
to be verbose in describing what I'm doing and what problems I'm having.
I'll make sure I'm more so in the future.
At one time I was hearing sipx MOH (classical guitar, correct?), but I'm
not hearing it any more.
I'll
You really ought to confirm your phones are set to use the MOH uri and that
noone has touched that.
You really ought to ensure the itsp account is set to not use MOH and that
they are set to use G711 codecs only.
A siptrace would tell a whole lot from a failed call flow though.
The fact that
Content-Type: text/plain;
charset=utf-8
Content-Transfer-Encoding: 8bit
Organization: SipXecs Forum
X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 67015
Message-ID: 105c7.4f723...@forum.sipfoundry.org
I'm getting an intermittent outbound issue and it appears to
be a problem between the proxy
When dialing an extension, it will ring through to voicemail and then
two seconds later, disconnect you. Not sure where to even begin on this
one, I just built this server and it has been running fine until today
and no changes have been made.
Regards,
Aaron
Here is what the sip traffic
Here is the log info from the sipxivr.log:
2012-03-27T22:22:20.227000Z:383:sipXivr:DEBUG:midcscmph.esgw.org:Thread-7::sipxivr:::awaitLiveEvent
event: Content-Disposition: disconnect
On 3/27/2012 6:12 PM, Josh Kennedy wrote:
I'm getting an intermittent outbound issue and it appears to
be a problem between the proxy and the bridge during an
invite. Below is an excerpt from the trace. The first invite
was from a successful call, the second from a failed call.
Both were to
You are only showing one side of this. Put the proxy and bridge mechanisms
to debug and get a sip trace and attach it. You emailer is interfering with
the inline text.
You also need to describe the call flow and mechanisms (firewall, its,
etc.).
On Mar 27, 2012 6:12 PM, Josh Kennedy
Ahem---siptrace. If it were me I'd make sure it was updated to latest
stable 4.4 and my intranet sub nets was correct while using firmware 3.2.6
on the phones...
On Mar 27, 2012 6:16 PM, Aaron Pursell aar...@esgw.org wrote:
When dialing an extension, it will ring through to voicemail and then
Is this a new install? Describe the install method. Weird message that
showed before disconnected...
Can you call IVR at all internally? Was this a restore? Dis you send the
server its profiles?
On Mar 27, 2012 6:25 PM, Aaron Pursell aar...@esgw.org wrote:
Here is the log info from the
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