After talking with Tony, I unfortunately know I can't put a G729 freeswitch
license in the version of freeswitch that runs in sipx 4.2.1. I got Verizon to
up my bandwidth, so I can switch sipxbridge back to blocking G729. I can't
really do this at my low bandwidth sites though.
Does anyone hav
They fixed a memory leak related to INVITE with replaces that might resolve
your issue in 3.2.4 (check the release notes)
The only way to find out is to set up a test system and test test test.
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list
On Tue, 25 Jan 2011 19:16:13 -0500, Tony Graziano wrote:
> There is just the 4.4 snapshot and beyond (4.5.2) as of right now.
I need to get the latest fixes and from what I understand, they are in at least
4.3 so need to update a system It's not in production until next week but I
need to make s
There is just the 4.4 snapshot and beyond (4.5.2) as of right now.
On Tue, Jan 25, 2011 at 7:09 PM, m...@grounded.net wrote:
> I believe 4.3 is in another repo, can someone tell me which repo I need to
> update a system to 4.3 but not letting it go any higher.
>
>
I believe 4.3 is in another repo, can someone tell me which repo I need to
update a system to 4.3 but not letting it go any higher.
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Patton.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
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We are looking at setting up a new SipX server using 4 pots lines. I would
like recommendations for gateways. The main feature I am looking for is
directing incoming and outgoing calls. Two lines go to an auto-attendant
that callers can then dial an extension, line two is a roll over for line
on
if your firmware is 4.x, split is all you need.
As for provisioning via mac, it can be done I think, but would require
sixconfig work, logic work, etc. Since there is another big firmware
creeping up it would be nice to get that underneath and working.
Version 3.3.x has new features and older pho
On 1/24/11 5:13 PM, Josh M. Patten wrote:
Polycom firmware 3.2.4 is finally out
quick question: split vs combined.. why? and why not?
I ASSUME that since cisco's phones can download a different FW per
phone, and poly can't, you SHOULD:
Load the 'legacy' firmware (my 301's and 4000 should run
I'm running sipXconfig (4.2.1-018971 2010-08-17T02:20:18 build20),
sipxbridge, Verixon VOIP, Polycom 450s, 550s, and 1 650.
It has been a rough week. I'll leave out some of the details as to how I
got where I am, and just describe the problem.
I have sipxbridge set to allow all codecs.
Phones ar
Thanks for your quick response!
I think I was following an old set of instructions for setting up OpenACD.
Before the Call Center -> "Lines" and "Commands" were added to sipXconfig, I
was manually editing the freeswitch dial plan template file, and I first
thought you were referring to this. I
Sorry for the multiple emails.
It looks like this is an actually a g711 vs g729 issue. If the
phones are set to use g729, they die. Sorry for any confusion. I
will start an entirely new thread once I put out my many fires...
On 1/25/2011 2:34 PM, Matthew Kitchin (
Activate the firmware, but reboot ONE phone and test... and meanwhile
in an office not far away one person unplugs their phone, and three
set a speeddial and wham! mixed firmware!
Firmware based on mac would be a pretty sweet feature, easy to support
older phones too!
On Mon, Jan 24, 2011 at 4:
Is anyone out there able to confirm that the SLA issue (I have to reboot the
phones every few days, or they exhibit behaviors like no ringing as well as
dropping longer phone calls) has been resolved in 3.2.4? I would test if I
had the spare equipment...
On Tue, Jan 25, 2011 at 12:03 PM, Matthew K
3.2.4 transferring to either 3.2.4 or 3.2.1B doesn't work for me.
I'm not sure where the issue is, but that is what I'm experiencing.
On 1/25/2011 10:46 AM, Michael Scheidell wrote:
On 1/25/11 11:33 AM, Matthew Kitchin (public/usenet) wrote:
Blind transfer to an extension. That extension rings o
I just tried this with an inbound call from voip.ms - it worked as well.
I also tried this out on 4.4.0 beta without issue trunking with an Audiocodes
PRI gateway. Anyone else out there running sipXbridge care to give this a whirl?
This message and any files tr
On 1/25/11 11:33 AM, Matthew Kitchin (public/usenet) wrote:
Blind transfer to an extension. That extension rings one time, and the
call drops. I've tried it with the destination extension running 3.2.1
B and 3.2.4. Same thing
to confirm: FW < 3.2.1B works fine? so is this a 3.2.4 and 3.2.1B FW
Verizon VOIP/ITSP.
Sipxbridge.
sipXconfig (4.2.1-018971 2010-08-17T02:20:18 build20)
I'm hoping to get a wireshark or sip trace in the next hour or 2.
On 1/25/2011 10:45 AM, Josh M. Patten wrote:
What are you using for your inbound calls? An ITSP, a gateway?
*From:*sipx-users-boun...@list.sip
What are you using for your inbound calls? An ITSP, a gateway?
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Matthew Kitchin
(public/usenet)
Sent: Tuesday, January 25, 2011 10:33 AM
To: Discussion list for users of sipXecs software
Subj
great question. short answer is it must be INVITE, not REFER as it
stays in the media path.
Looking for a REFER based connection to agent would allow for some
insane scalability as OpenACD really only consumes a fraction of the
CPU to do it's job. OpenACD today uses the FS session to determine
It is being transferred to a handset.
Blind transfer to an extension. That extension rings one time, and the
call drops. I've tried it with the destination extension running 3.2.1 B
and 3.2.4. Same thing. It only happens on inbound calls. If the call
originated from the Polycom handset, it is f
Is the call being transferred to Vm when it is being dropped (i.e. blind
xfer to an AA or a user who has DND on or doesn't answer and goes to VM)?
I have no issues with it but am testing that right now on a snapshot...
On Tue, Jan 25, 2011 at 9:39 AM, Matthew Kitchin (public/usenet) <
mkitchin.
I guess it is siptrace time.
If I try and do a blind transfer on an inbound call, the call is dropped.
On 1/24/2011 5:41 PM, Matthew Kitchin (usenet/public) wrote:
> Great. I loaded it on 10 phones this afternoon. Maybe this will be the mother
> of all firmwares and will fix a bunch of my issues.
as an incoming call center the call would go to the agent via an invite.
i am assuming it will bridge the call via its FS config, and there is no
transfer.
though i might be wrong and will wait to be corrected...
On Tue, Jan 25, 2011 at 9:28 AM, Matt White wrote:
> Sorry for being too lazy to t
Sorry for being too lazy to test this for myself. I was just talking with
someone and the question came up.
Does the new openacd implementation use the same conference method to join the
caller to the agent the current ACD uses or does it use a standard refer?
-M
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