You see this a lot in CDR. When a call is transferred to Voicemail and the
drop off, or when they drop out of the Auto-Attendant you see failed calls.
I'm unclear if it is because the call length was too short, or what, but it
seems to be a normal behavior today with sipXecs.
If someone knows
Hi all,
I have purchased a dedicated circuit from my telephony vendor that
will be used exclusively for SIP trunking. It will have a different
IP address than my normal site internet connection.
My question is, will this cause me problems with sipx? In particular,
when users outsid
I have a lab machine that was running the latest stable version 4.0.4 RPM.
I upgraded it to the latest RPM development release so that I could test out
g722 support in the voicemail server.
I have now removed all sipx packages and reinstalled 4.0.4.
When it starts up, I get the following EXCEPTION
Will there be any integration of the Openfire component with Outlook or
SharePoint - like LCS/OCS in the future?
I thought this was an interesting piece - basically there are already projects
or work out there that tries to mimic LCS or OCS from Microsoft using XMPP /
Jabber / Openfire.
http:/
On Thu, Dec 10, 2009 at 4:52 PM, Burden, Mike wrote:
> I have a Trunk Hunt group set up at extension 99 that rings all of our
> “hard” phones simultaneously.
>
> The destination for inbound calls is set up as 99 in the sipXbridge.
Other than the CDR does the system actually work the way you would
I'll do some testing. If it is still intermittent on the system that "worked"
after
clearing asserted identity i'll be sure to send some traces.
Thank you
//Ola
Från: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] för Sco
First, thank you for great support. Hope to able to contribute back
as i learn more.
Mobility is hot in Sweden and Ericsson and others have had
support for mobile extensions for many years.
"like the mobile is an extension/a phone under the pbx"
People i talk to ask for:
1 twinning, call the o
Thank you.
That helped for the systems that did not work. They now forward, but
what about the system that worked all along?
Can this be a bug since it worked without empty asserted identity?
//Ola
Från: M. Ranganathan [mra...@gi mail.com]
Skickat: den 1
On Thu, 2009-12-10 at 21:56 +0100, Ola Samuelsson wrote:
> Hello!
> Sorry for being unclear.
> I do mean server based forwarding for the users, not the phones.
> So i guess it should work or at least behave consistently.
> Thank you.
> //Ola
Then traces of successful and unsuccessful calls are pro
On Thu, Dec 10, 2009 at 3:56 PM, Ola Samuelsson
wrote:
> Hello!
> Sorry for being unclear.
> I do mean server based forwarding for the users, not the phones.
> So i guess it should work or at least behave consistently.
> Thank you.
> //Ola
>
>
[ Forgot to reply to thread on the mailing list }
Do
I think that is the current behavior. It is similar with AA.
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
I have a Trunk Hunt group set up at extension 99 that rings all of our
"hard" phones simultaneously.
The destination for inbound calls is set up as 99 in the sipXbridge.
The CDR report shows all inbound calls as having "failed", even though
all of the calls were answered.
Is my configurati
Todd,
Thanks. This is exactly what I needed. It took me about 10 minutes to
modify the dial plan and it's working perfectly now using the following
dial plan:
"[2-9]11|0T|100|101|011xxx.T|9011xxx.T|9[2-9]x|*xx|[8]xxx|[2-7]x
x". The only thing that my users aren't going to like is di
Hello!
Sorry for being unclear.
I do mean server based forwarding for the users, not the phones.
So i guess it should work or at least behave consistently.
Thank you.
//Ola
Från: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.o
On Thu, 2009-12-10 at 20:46 +0100, Ola Samuelsson wrote:
Please clarify... when you say:
> - Setting up forward on a phone to any external number
do you mean:
1. that you set forwarding for the user in the sipXecs
configuration for that user.
2. that you configured the phone i
MoH on the Astra does not work. Aastra had semi-committed to fixing this in
the 3.0 firmware release which was due on Nov.
-M
>>> On 12/10/2009 at 01:46 PM, in message <4b214201.4030...@spudland.com>,
>>> Robert B wrote:
So, now that I have a working system to test...
My first MoH test w
To my knowledge MoH does not work on Aastra phones.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Robert B
Sent: Thursday, December 10, 2009 1:46 PM
To: sipx-users@list.sipfoundry.org
Subject: [sipx-users] Music on hold... strange beha
Good evening!
I have a couple of systems set up.
All using Centos 5.2 and latest 4.0.4.
Same ITSP for all. Same config on ITSP side for all.
I have problems with forwarding which works intermittently on one system
and not at all on two others.
This is what i did/have:
- Exactly same config on a
Thanks Tod.
In fact, i did that and it works.
But i got help recording some prompts by someone that could not call in and
also i want to edit prompts (even if they were recorded by phone).
I used sox and audacity and the files are indeed Little endian, 16bit pcm, 8khz
wav files, at least accordin
So, now that I have a working system to test...
My first MoH test was conducted by calling my AT&T cell phone (Nokia
E71) from my Aastra 57i handset.
*Test 1:
*
I press Hold on the Aastra, music playing begins on my cell phone. I was
surprised, since I was told earlier that MoH is broken on t
One very simple way to create your messages for sipXecs and ensure they are
formatted correctly is to leave a recording in a voicemail box, and then
grab it from the user portal as a wav file. Its guaranteed to be in the
correct format, and its very simple to create and move to your recordings.
I don't get it. I thought the UI was already being worked on for 4.2 as
per the roadmap.
http://sipx-wiki.calivia.com/index.php/Next_Generation_UserPortal
-mark
Scott Lawrence wrote:
> On Wed, 2009-12-09 at 05:55 -0800, Jordan Turner wrote:
>
>> It is nice due to its organizational layout.
Bingo!
/var/sipxdata/phonebook was owned by root.
This was a freshly installed system, and I imported the configuration
from my previous system. Sounds like something in the configuration
import isn't careful with permissions...
Thanks!!
Mike Burden
Lynk Systems, Inc
e-mail: m...@lynk.com
Burden, Mike wrote:
> Naming the file Customers.csv and attempting to import it gives me the
> error message, and the attached log file entries.
>
> Doing *NOTHING* except renaming the file to Phonebook.csv allows me to
> import it with no errors.
>
> It's worth noting that there *IS* no colu
Naming the file Customers.csv and attempting to import it gives me the error
message, and the attached log file entries.
Doing *NOTHING* except renaming the file to Phonebook.csv allows me to import
it with no errors.
It's worth noting that there *IS* no column 84 on line 3 (regardless of w
Burden, Mike wrote:
> Good morning,
>
> I had one .CSV file called phonebook.csv that contained Employees,
> Customers, and Vendors. I imported that file without a problem.
>
> Then I decided to split this into Customers.csv, Employees.csv, and
> Vendors.csv. When I try to import any of t
Hi Josh, Tony,
.Because I did not do any testing on my first set of rules I did
not discover that they did not work...
So I tested the new rules, and I know that these rules work.
With this solution you only need 3 rules for each areacode that has the 7
vs 10 digit problem.
So in Ton
Good morning,
I had one .CSV file called phonebook.csv that contained Employees,
Customers, and Vendors. I imported that file without a problem.
Then I decided to split this into Customers.csv, Employees.csv, and
Vendors.csv. When I try to import any of these, I get:
An inter
What kind of gateway(s) do you have?
I let my Audiocodes gateways do all of my number manipulations. They
tend to be a bit more flexible. In the "Dest Number IP -> Tel"
manipulation table set a manipulation entry for each local prefix you
wish to strip the area code from; say 408-555- is a
Hi Tony,
I thought that after applying a rule that the processing of the would
continue with the next rule,
but it doesn't. It starts from the beginning.
So my rules don't work.
I still wanted to find a solution so I made it "more unique"
Seven_only_111 (custom rule, so that you can have mul
Hi all!
Greatful for any clue.
I am on CentOs5.2/4.0.4 and get:
[WARNING] switch_core_file.c:119 switch_core_perform_file_open() Sample
rate doesn't match
when changing a prompt.
The prompt is played and sound like it should but log fills up with
these messages.
I have checked with tools li
Hi!
My logs are full of:
"2009-12-08T12:34:49.465387Z":484:SIP:WARNING:sip.xxx.se:SipRouter-11:B5CBCB90:SipXProxy:"SipUserAgent::send
SUBSCRIBE request matches existing transaction"
"2009-12-08T12:35:14.835795Z":485:HTTP:ERR:sip.xxx.se::B6432B90:SipXProxy:"HttpMessage::get[4]
Receiving failed
I have 25-30 nxx's (npa-nxx-) in one area code and 5-6 nxx's in another
that are local calls. My telco send me 10 digits on callerid. They wont
accept the local calls dialed as 10 digits when I hit return call on the
phone. I don't see how 6 rules would suffice, since I need to strip the area
c
I was off for a few days, but see that Tony is still looking for a
solution.
I still think my solution works and is easy:
http://list.sipfoundry.org/archive/sipx-users/msg19320.html
with only 6 rules instead of many. I don't see any reaction however so
either something is wrong or I am not under
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