://www.myitdepartment.net/gethelp/
- Original Message -
From: sipx-users-boun...@list.sipfoundry.org
To: Ken Fulmer
Cc: sipx-users@list.sipfoundry.org
Sent: Tue Jul 20 11:12:06 2010
Subject: Re: [sipx-users] VoIP Providers
I've had good luck with Voip.ms as well.
The only technical issue with them is
We've been using Broadvox but orders are taking a long time to complete. Has
anyone on the list had good experiences with other VoIP providers? We're
looking for one with many rate centers and the ability to get or port
numbers in the Southeast.
Thanks,
Ken
-Original Message-
From: WORLEY, Dale R (Dale) [mailto:dwor...@avaya.com]
Sent: Thursday, May 20, 2010 9:50 PM
To: Ken Fulmer; sipx-users@list.sipfoundry.org
Subject: RE: [sipx-users] Xeon or Pentium
From: sipx-users-boun...@list.sipfoundry.org
[sipx-
Based on experience, do you guys prefer a Xeon or Pentium processor to run
sipXecs? Would a dual Xeon 2.8 with 4 GB RAM be considered a strong system
or should we go with a different blend?
Thanks,
Ken Fulmer
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he CC machine? If it's pretty old then it may not have error
correction capabilities needed to compensate for latency and jitter etc.
introduced by IP
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 5/14/2010 8:29 AM, Ken Fulmer wrote:
> I should also say
Not sure what you are trying to say.
Ken
-Original Message-
From: Michael Scheidell [mailto:scheid...@secnap.net]
Sent: Friday, May 14, 2010 8:34 AM
To: Ken Fulmer; 'Picher, Michael'; 'Tony Graziano'
Cc: sipx-users@list.sipfoundry.org
Subject: RE: [sipx-users]
he CC machine? If it's pretty old then it may not have error
correction capabilities needed to compensate for latency and jitter etc.
introduced by IP
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 5/14/2010 8:29 AM, Ken Fulmer wrote:
> I should also say
I should also say we are using "modem pass-through" for the fax
transmissions over ip.
Ken
-Original Message-----
From: Ken Fulmer [mailto:kenful...@icstechnologysolutions.com]
Sent: Friday, May 14, 2010 8:28 AM
To: 'Picher, Michael'; 'Tony Graziano'; '
Michael Scheidell
> Cc: sipx-users@list.sipfoundry.org
> Subject: Re: [sipx-users] Cisco ATA and Credit Card Terminals
>
> modems over sip sucketh. dont do that.
>
> On Thu, May 13, 2010 at 3:29 PM, Michael Scheidell
> wrote:
> > On 5/13/10 3:20 PM, Ken Fulmer wrote:
&
ga [mailto:justin.me...@gmail.com]
Sent: Thursday, May 13, 2010 3:59 PM
To: Ken Fulmer
Cc: JOLY, ROBERT (ROBERT); sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Music on Hold
Are you using SIPXecs 4.2?
Doesn't the Freeswitch Voicemail implementation in SIPXecs 4.2 store all
that
need to be tweaked for credit card terminals?
Thanks,
Ken Fulmer
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half Of Matthew Kitchin
(public/usenet)
Sent: Thursday, May 13, 2010 1:03 PM
To: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Memory Usage
Which process is using all the memory?
On 5/13/2010 12:53 PM, Ken Fulmer wrote:
When we SSH into our 4.2 server and use Top or Free -m, we see al
the system reserve the
memory for certain functions?
Thanks,
Ken Fulmer
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True, but there are music files encoded specifically for 729. Problem is,
the system won't allow them to be uploaded.
Ken
-Original Message-
From: JOLY, ROBERT (ROBERT) [mailto:rj...@avaya.com]
Sent: Thursday, May 13, 2010 8:46 AM
To: Ken Fulmer; sipx-users@list.sipfoundry.org
Su
We are transcoding 711 to 729 for calls to the PSTN. Music on Hold sounds
choppy to the remote side.
Is there a way to tweak the system so it will accept wav files recorded in a
more 729 friendly format?
Thanks,
Ken Fulmer
___
sipx
I've been studying the Freeswitch architecture and I know a portion of
Freeswitch has been incorporated into version 4.2.
Can someone point me in the right direction to find the hierarchical
architecture of the sipX / Freeswitch combination of 4.2?
Thanks,
Ken F
When a call is on hold, we want to play a one second beep tone every 15-30
seconds. We uploaded the .wav file and we're hearing a series of quick
beeps. Can this be changed in 4.2?
Thanks,
Ken Fulmer
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Does anyone know of a good source for MoH files encoded for g.729?
Thanks,
Ken Fulmer
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They are on the same LAN segment.
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Thursday, April 29, 2010 11:33 AM
To: Ken Fulmer
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] 4.2 Polycom phones losing registration
are the phones local or remote?
if
They are on the same LAN segment.
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Thursday, April 29, 2010 11:33 AM
To: Ken Fulmer
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] 4.2 Polycom phones losing registration
are the phones local or remote?
if
timer.
I did a search on the user list and didn't find any posts regarding this
problem. Should we check something else in the system?
Thanks,
Ken Fulmer
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They are using Broadworks.
-Original Message-
From: M. Ranganathan [mailto:mra...@gmail.com]
Sent: Tuesday, April 27, 2010 8:59 AM
To: Ken Fulmer
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] SIP Diversion vs P-Asserted Identity for External
Call Forwarding
On Tue, Apr
Ok, thanks for your reply. We'll have to work something out with our
provider.
Ken
-Original Message-
From: M. Ranganathan [mailto:mra...@gmail.com]
Sent: Monday, April 26, 2010 3:39 PM
To: Ken Fulmer
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] SIP Diversion
PaeTec won't support the P-Asserted Identity for external call forwarding.
They require a Diversion Header that contains a PaeTec number on the SIP
trunk.
Does anyone know how to accomplish this with sipX?
Thanks,
Ken Fulmer
___
Ok, thanks for the feedback. We'll have to tread carefully.
Ken
From: Picher, Michael [mailto:mpic...@cmctechgroup.com]
Sent: Friday, April 23, 2010 11:52 AM
To: Francis Tinio; Ken Fulmer
Cc: sipx-users@list.sipfoundry.org
Subject: RE: [sipx-users] sipX 4.2 and VMware?
You'
Are there any known issues with running sipX 4.2 on VMware? As I understand
it, there were problems with earlier releases and VMware.
Thanks,
Ken Fulmer
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rward no answer / busy, etc. How do these fields relate to
the Call Forwarding screen under the user extension?
Thanks,
Ken Fulmer
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sipX? We are using Polycom 3XX / 5XX / 6XX
phones.
Thanks,
Ken Fulmer
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Thanks guys!
I'll definitely use that command to clear my log files.
Ken
From: Picher, Michael [mailto:mpic...@cmctechgroup.com]
Sent: Wednesday, April 21, 2010 10:11 AM
To: Ken Fulmer; sipx-users@list.sipfoundry.org
Subject: RE: [sipx-users] sipXproxy.log
Thanks. I deleted the file on a test machine and it reappeared when I
restarted the SIP Proxy service.
Ken
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Wednesday, April 21, 2010 10:05 AM
To: Ken Fulmer
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users
e SIP Proxy service's logging level to INFO.
We noticed TONS of subscribe messages (I assume reported from the Registrar
service). Should we be looking at lengthening the registration interval?
Thanks,
Ken Fulmer
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/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
Fax: 434.984.8427
Helpdesk Contract Customers:
http://www.myitdepartment.net/gethelp/
- Original Message -
From: sipx-users-boun...@list.sipfoundry.org
To: Ken Fulmer
Cc: sipx-users@list.sipfoundry.org
Sent: Tue Apr 20 18:56
t: Re: [sipx-users] g.729
I believe the newer models support iLBC, but I couldn't get it working.
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 4/20/2010 4:08 PM, Ken Fulmer wrote:
We are using Polycom phones and I don't think they suppor
We are using Polycom phones and I don't think they support speex or ILBC (at
least not all the models).
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Josh Patten
Sent: Tuesday, April 20, 2010 3:57 PM
To: sipx-users@list.sipfoundry
So, would it be possible to terminate g.729 to media services if the license
is installed under FreeSwitch?
Thanks,
Ken
From: Eric Varsanyi [mailto:sip...@eljv.com]
Sent: Monday, April 19, 2010 6:43 PM
To: Picher, Michael
Cc: Ken Fulmer; sipx-users@list.sipfoundry.org
Subject: Re
going through
10.10.3.11.
Thanks,
Ken
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Ken Fulmer
Sent: Monday, April 19, 2010 1:44 PM
To: 'M. Ranganathan'
Cc: sipx-users@list.sipfoundry.org
Subject:
mail related services running. This seems to break
Music on Hold. When we move the VM services to the primary server in the
cluster, MoH works properly.
-Original Message-
From: M. Ranganathan [mailto:mra...@gmail.com]
Sent: Monday, April 19, 2010 12:16 PM
To: Ken Fulmer
Cc: sipx-users
the first, music on
hold breaks. We are using the 711u codec.
Thanks,
Ken Fulmer
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Has support for g.729 improved with the 4.2 version?
Thanks,
Ken Fulmer
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Ok, sounds good. Thanks for all the hard work!
Ken
-Original Message-
From: Scott Lawrence [mailto:xmlsc...@gmail.com]
Sent: Friday, April 16, 2010 4:10 PM
To: Ken Fulmer
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] 4.2 Documentation
On Fri, 2010-04-16 at 15:29 -0500
Ok, thanks. Just checking.
Ken
-Original Message-
From: Scott Lawrence [mailto:xmlsc...@gmail.com]
Sent: Friday, April 16, 2010 4:11 PM
To: Ken Fulmer
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] 4.2 Cluster Scalability
On Fri, 2010-04-16 at 15:35 -0500, Ken Fulmer
I'd asked a question regarding how many servers could exist in a cluster.
The reply was no more than 5 had been tested.
Has this number been scaled higher in 4.2?
Thanks,
Ken Fulmer
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Sorry if already asked:
Is there a specific area in the wiki for 4.2 information?
Thanks,
Ken Fulmer
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I've been wondering the same thing. Thanks for clarifying for the group.
Does the Dial Plan file need to be edited in any way?
Thanks,
Ken Fulmer
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf O
Can someone explain the difference between a SIP Trunk with an Unmanaged SBC
and an Unmanaged Gateway? How exactly is a SIP Trunk with an Unmanaged SBC
supposed to be configured?
Thanks,
Ken Fulmer
___
sipx-users mailing list sipx-users
he user is configured with
an internal extension and a 10-digit DID as an alias. I also reconfigured
the user as a 10-digit DID and got the same result.
Has anyone seen this behavior?
Thanks,
Ken Fulmer, CCIE #20639
Senior Systems Design Engineer
ICS, Inc.
7008 Champions Blvd,
phone: 434.984.8426
Fax: 434.984.8427
Helpdesk Contract Customers:
http://www.myitdepartment.net/gethelp/
- Original Message -
From: Ken Fulmer
To: 'Tony Graziano'
Cc: sipx-users@list.sipfoundry.org
Sent: Wed Mar 24 16:58:23 2010
Subject: RE: [sipx-users] 4.1.7
Hmm.I'm w
Hmm.I'm wondering about a new install instead of an upgrade. Is anyone doing
that yet?
Ken
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Wednesday, March 24, 2010 3:37 PM
To: Ken Fulmer
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] 4.1.7
O
This may be a crazy question, but is anyone running the 4.1.7 beat in
production?
Thanks,
Ken Fulmer
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Ok, thanks.
From: Picher, Michael [mailto:mpic...@cmctechgroup.com]
Sent: Monday, March 22, 2010 10:19 AM
To: Ken Fulmer; sipx-users@list.sipfoundry.org
Subject: RE: [sipx-users] Polycom ILBC
G.711u and a are the only codecs that work with the media services.
From: sipx-users-boun
Has anyone gotten ILBC to work with Polycom phones in the sipX system? The
codecs don't show up on the phone voice codec page. I logged into the
phone's web browser and selected ILBC 15.2 kbps but get a fast busy when
attempting an outbound call.
Thanks,
Thanks for the reply. How are you getting the phones to register with sipX?
I don't see them in the drop down box. Are you using a manual method?
Ken
From: Bob Anderson [mailto:bob.ander...@cyrand.com]
Sent: Monday, March 22, 2010 8:10 AM
To: Ken Fulmer
Cc: sipx-users@list.sipfoundr
Has anyone used the Polycom Spectralink wireless phones with sipX?
Thanks,
Ken Fulmer
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Ok, thanks for the link. We'll work on that configuration in our lab.
Ken
-Original Message-
From: Scott Lawrence [mailto:xmlsc...@gmail.com]
Sent: Thursday, March 18, 2010 11:51 AM
To: Ken Fulmer
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Call Routing
On Thu,
)?
Thanks,
Ken Fulmer
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sipXecs IP PBX -- http://www.sipfoundry.org/
(s) at the head-end.
Thanks,
Ken Fulmer
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sipXecs IP P
We've been talking about internal extensions as our default dial plan. I'll
convert the phones and see what happens.
Thanks,
Ken
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Tuesday, March 16, 2010 9:31 AM
To: Ken Fulmer
Cc: sipx-users@list.sipfoundry.o
user and the last 4 digits as an alias.
For some reason, when I call using the 4 digit alias, the attendant says
that number is not valid. Strange!
Thanks,
Ken
From: Picher, Michael [mailto:mpic...@cmctechgroup.com]
Sent: Monday, March 15, 2010 8:16 PM
To: Ken Fulmer; Tony Grazia
phones can pull the necessary information to utilize the DNS SRV records?
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Monday, March 15, 2010 5:36 PM
To: Ken Fulmer
Cc: Josh Patten; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] DNS SRV and Polycom
The
...@myitdepartment.net]
Sent: Monday, March 15, 2010 5:13 PM
To: Ken Fulmer
Cc: Josh Patten; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] DNS SRV and Polycom
If you do an nslookup for the SRV records for that domain, and it
returns the correct records, you should be fine so long as the correct
hones will look for the
following uri: _sip._udp.example.com?
Ken
From: Josh Patten [mailto:jpat...@co.brazos.tx.us]
Sent: Tuesday, March 09, 2010 4:49 PM
To: Ken Fulmer
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] DNS SRV and Polycom
I do, but I don't know about
The SRV records appear to be configured correctly in our MS DNS server. When
I perform an NSLOOKUP for "_sip._udp.example.com", I get the expected two
sipX servers.
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Tuesday, March 09, 2010 5:00 PM
To: Ken Fulme
10 4:37 PM
To: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] DNS SRV and Polycom
You must set the DNS servers in your DHCP scope on whatever DHCP server you
are using.
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 3/9/2010 4:32 PM, Ken Fu
I have DNS SRV setup on a MS DNS server. Using NSLOOKUP, the records return
the correct information.
I'm not sure how to get the phones to contact the DNS server to look up the
SRV records. Do we still use option 66 or some other mechanism?
Thanks,
Ken F
Thanks!
-Original Message-
From: M. Ranganathan [mailto:mra...@gmail.com]
Sent: Tuesday, March 09, 2010 10:34 AM
To: Scott Lawrence
Cc: Ken Fulmer; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] sipXbridge Redundancy
On Tue, Mar 9, 2010 at 11:30 AM, Scott Lawrence
wrote:
>
Thanks!
-Original Message-
From: Scott Lawrence [mailto:scottlawr...@avaya.com]
Sent: Tuesday, March 09, 2010 10:28 AM
To: Ken Fulmer
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Simultaneous Calls vs Calls Per Second
On Tue, 2010-03-09 at 10:11 -0600, Ken Fulmer wrote
tween simultaneous calls and calls per second?
Thanks,
Ken Fulmer
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The design doc mentioned one sipXbridge per cluster. The link above suggests
two can be used in a HA setup. Can anyone clarify?
http://www.sipfoundry.org/component/content/article/29-test-read-more-link-p
age.html
Thanks,
Ken Fulmer
some results in the
> mailing list. A quick google shows this thread from back in 2008:
> http://www.mail-archive.com/sipx-...@list.sipfoundry.org/msg00038.html
>
> -Matt
>
>>>> "Ken Fulmer" 03/04/10 6:42 PM
>>>
> I appreciate your responses. However
rtment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
Fax: 434.984.8427
Helpdesk Contract Customers:
http://www.myitdepartment.net/gethelp/
- Original Message -
From: Ken Fulmer
To: 'Tony Graziano'
Cc: sipx-users@list.sipfoundry.org
Sent: Thu Mar
26
Fax: 434.984.8427
Helpdesk Contract Customers:
http://www.myitdepartment.net/gethelp/
- Original Message -
From: Ken Fulmer
To: 'Tony Graziano'
Cc: sipx-users@list.sipfoundry.org
Sent: Thu Mar 04 17:43:29 2010
Subject: RE: [sipx-users] 250 Concurrent Calls?
I'
I'm trying to understand how the 250 number was generated.
We'd want to use a dedicated server for the sipXbridge.
Ken
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Thursday, March 04, 2010 4:15 PM
To: Ken Fulmer
Cc: sipx-users@list.sipfoundry.org
S
?
Thanks,
Ken Fulmer
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sipXecs IP PBX -- http
eneck."
Since the sipXbridge is passing RTP through the server at 50 packets per
second (20 ms sample), is the concurrent call number (250) based on server
specs? If so, does documentation exist for varying processor speeds, number
of processors, etc.
Thanks,
s SIP REFER messages?
Thanks,
Ken
-Original Message-
From: Dale Worley [mailto:dwor...@avaya.com]
Sent: Tuesday, February 16, 2010 9:42 AM
To: Ken Fulmer
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Phones that don't use SIP REFER messages
On Tue, 2010-02-16 a
We've had some problems with the Polycom phones use of SIP REFER messages
with Cisco and Adtran voice gateways. Call transfers and holds fail due to
incompatibilities in the way the Cisco and Adtran routers handle these
events. My understanding is the routers use a mid-call INVITE rather than
SIP R
ase acknowledge that.
On Thu, Jan 7, 2010 at 12:49 PM, Ken Fulmer
wrote:
A user in sipX is configured for call forwarding to an external destination.
A call that originated in the PSTN gets forwarded back out to a destination
in the PSTN. However, there is no audio in either direction.
W
A user in sipX is configured for call forwarding to an external destination.
A call that originated in the PSTN gets forwarded back out to a destination
in the PSTN. However, there is no audio in either direction.
We are using the sipXbridge and a sip trunk. Routine inbound and outbound
calls
Can anyone provide a short list of good SIP trunk providers?
I've seen Bandwidth.com mentioned on the sipX user list. Can anyone share
their experience with that company?
Thanks,
Ken
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We are testing the Unmanaged Gateway setting on UDP port 5060. We can make
outbound calls with no problem.
However, inbound calls get a fast busy. Looking at a debug, we see the
message, 404 Not Found, coming from the sipX server. The phones are
registered with the system. All phones and the g
Good info. We've had issues with Polycom phones and REFER for call
transfers. So, we've been sending calls through the sipXbridge.
Ken
From: Josh Patten [mailto:jpat...@co.brazos.tx.us]
Sent: Wednesday, December 23, 2009 1:00 PM
To: Ken Fulmer
Cc: sipx-users@list.sipfoundry.o
We have users with iphones that register to the sipX server with only the
user being created (no phone). The application on the iphone registers and
works well.
Can an Adtran router with analog ports register to the sipX server in the
same manner?
Thanks,
Ken
Do you know when version 4.1 will be available?
Thanks,
Ken
-Original Message-
From: M. Ranganathan [mailto:mra...@gmail.com]
Sent: Monday, December 14, 2009 9:43 AM
To: Ken Fulmer
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Call Admission Contol (CAC)
On Mon, Dec 14
Does anyone know how to limit the number of calls in / out of the sipX
system? We are using the sipX bridge (sip trunking) functionality.
Thanks,
Ken
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What happens at the packet level when Server Behind Nat and Nat Traversal
are enabled?
Thanks,
Ken
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No, I didn't receive those directions earlier. Thanks for re-sending. I'll
see if it corrects the problem.
Thanks,
Ken
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Wednesday, October 28, 2009 3:47 PM
To: kenful...@icstechnologysolutions.com; sipx-users@list.sipfoundr
Does anyone know how to install the sipxbridge patch?
Thanks,
Ken
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- Original Message -
From: Ken Fulmer
To: 'Tony Graziano'
Cc: sipx-users@list.sipfoundry.org
Sent: Wed Oct 28 15:15:20 2009
Subject: RE: [sipx-users] MoH in CDR
Not sure how to patch.
Not sure how to patch. Is there a link that has instructions?
Ken
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Wednesday, October 28, 2009 2:05 PM
To: Ken Fulmer
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] MoH in CDR
Are these internal calls or
On two different systems, when we place a call on hold, it fails. Then the
CDR shows a call to MH for an extended period of time.
We're using Polycom phones with load 3.1.3RevC.
Is there a workaround for this issue?
Thanks,
Ken
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s
That document helped me understand sipX DNS much more thoroughly.
Thanks!
Ken
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Wednesday, October 28, 2009 10:34 AM
To: Ken Fulmer
Cc: Picher, Michael; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] sipXbridge
:mpic...@cmctechgroup.com]
Sent: Wednesday, October 28, 2009 10:21 AM
To: Ken Fulmer; sipx-users@list.sipfoundry.org
Subject: RE: [sipx-users] sipXbridge Question
You just need to get DNS stuff all setup properly.
Mike
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users
A sipX server is in a DSL site with several phones on the local lan. We also
have remote phones that need to register across the Internet from other DSL
sites (home, travel, etc).
We have nat traversal and server behind nat checked. Should anything else be
configured to get this scenario workin
Thanks,
Ken
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sipXecs IP PBX -- http://www.sipfoundry.o
Does Internet Calling need to be enabled? I'm still unclear about its
functionality.
Thanks,
Ken
From: Robert Joly [mailto:rj...@nortel.com]
Sent: Monday, October 26, 2009 11:53 AM
To: Ken Fulmer; sipx-users@list.sipfoundry.org
Subject: RE: [sipx-users] Server Behind NAT
The
If the setting, "Server behind NAT" is checked, can phones that are local to
the site connect to the system? Or will all phones need to access the server
on its publicly translated address?
Thanks,
Ken
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sipx-users mailing list sipx-users@
What about the release timeline for version 4.2?
I understand it has some new features.
Ken
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Scott Lawrence
Sent: Monday, October 26, 2009 11:13 AM
To: Josh Patten
I'm downgrading them from 3.2.1 to 3.1.3RevC now.
Ken
From: M. Ranganathan [mailto:mra...@gmail.com]
Sent: Wednesday, October 21, 2009 2:29 PM
To: Ken Fulmer
Cc: sipx-users@list.sipfoundry.org
Subject: Re: FW: [sipx-users] Best B2BUA?
Polycom firmware has a bug. I posted about
either way works.
However, for outbound calls, Call hold / resume works fine.
Ken
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Ken Fulmer
Sent: Wednesday, October 21, 2009 1:34 PM
To: 'M. Ranganathan'
Cc: sipx-users
AM
To: Ken Fulmer
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Best B2BUA?
On Wed, Oct 21, 2009 at 11:41 AM, Ken Fulmer
wrote:
Anyone have a recommendation for a solid B2BUA for the sipX system? The
built-in version seems to only work for outbound calls. We need a device
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