If I understand your requirements properly, I see a couple devices that
should do what you want...
The Gigaset from Siemens ~ $100 (will pair with 3 phones and offer
distinctive ring for each, would be nice to find a gateway that supports
inbound distinctive ring on the fxo but I don't know of
:13 AM
To: Picher, Michael; 'Michael Scheidell'; 'Eric Varsanyi'
Cc: sipx-users@list.sipfoundry.org
Subject: RE: [sipx-users] sip - bluetooth
Nice find Mike!
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Picher,
Michael
Sent: Monday
I thought about that too but it wouldn't really be very automagic.
-Original Message-
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Monday, June 28, 2010 10:41 AM
To: sip...@eljv.com; Picher, Michael
Cc: scheid...@secnap.net; sipx-users@list.sipfoundry.org
Subject
How far are you from the CO?
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Nathaniel
Watkins
Sent: Friday, June 25, 2010 1:07 PM
To: sipx-users@list.sipfoundry.org
Subject: [sipx-users] Echo - PRI - Patton?
Anyone know if there
Call your provider to see if they can pad the circuit... it's probably too hot.
-Original Message-
From: Nathaniel Watkins [mailto:nwatk...@garrettcounty.org]
Sent: Saturday, June 26, 2010 10:45 AM
To: Picher, Michael; 'sipx-users@list.sipfoundry.org'
Subject: Re: [sipx-users] Echo
You can call pickup to a hunt group?
That's actually a new one on me, but I have never tried it before
either.
Mike
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-
boun...@list.sipfoundry.org] On Behalf Of Josh Patten
Sent: Friday, June 25, 2010
To: Picher, Michael
Cc: sipx-users@list.sipfoundry.org
Subject: RE: [sipx-users] Voice Transmission stops after 29 Minutes
Hey Mike,
Here is some more information.
No cisco phones were used. I tried it with those phones:
* Our Softphone Client (isPhone)
* Polycom IP670
Correct Douglas.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-
boun...@list.sipfoundry.org] On Behalf Of Douglas Hubler
Sent: Wednesday, June 23, 2010 7:55 AM
To: Staffan Kerker
Cc: sipx-users
Subject: Re: [sipx-users] Sipxconfig on seperate
The world famous Douglas Hubler will also be speaking on Wednesday the 4th with
his presentation “Contributing to the sipXecs Project”.
Hope to see many of you there!
Thanks,
Mike
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org]
Has anybody come across or written an app that can take an incoming
call, do a database pop on the incoming callerid and then route/transfer
the call to a database returned SIP url?
There would have to also be a failover ext for no callerid / database
timeout.
This seems simple enough...
It looks like Freeswitch could do this with mod_xml_curl ...
I'm just not into programming pain. :-)
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Picher,
Michael
Sent: Wednesday, June 23, 2010 9:13 AM
To: sipx-users
-
From: Douglas Hubler [mailto:dhub...@ezuce.com]
Sent: Wednesday, June 23, 2010 10:05 AM
To: Picher, Michael
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] call routing from database pop
On Wed, Jun 23, 2010 at 9:46 AM, Picher, Michael
mpic...@cmctechgroup.com wrote:
It looks
Usually what we do is leave the user with a regular extension and then
add the DID as an alias on the user account...
This does work.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-
boun...@list.sipfoundry.org] On Behalf Of maybelater
Sent:
@list.sipfoundry.org
Subject: Re: [sipx-users] Voice Transmission stops after 29 Minutes
On 6/23/10 11:57 AM, Picher, Michael wrote:
You left out some critical information...
Type of phones and how is user A reaching user B (ie., sipXbridge,
gateway)?
Are you running Cisco phones? If you
://list.sipfoundry.org/mailman/listinfo/sipx-users
sipXecs IP PBX -- http://www.sipfoundry.org/
From: Michael Scheidell [mailto:scheid...@secnap.net]
Sent: Wednesday, June 23, 2010 1:55 PM
To: Picher, Michael
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Voice Transmission stops
Josh,
I think when I do it I just create a new gateway and use the ip but give
it a different gateway name in sipX.
I'm pretty sure that works fine.
Mike
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Josh
Of Gerald Drouillard
Sent: Tuesday, June 22, 2010 9:19 AM
To: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Use one IP address for multiple gateways in
sipX?
On 6/22/2010 6:12 AM, Picher, Michael wrote:
Josh,
I think when I do it I just create a new gateway and use the ip
Yes.
-Original Message-
From: Josh Patten [mailto:jpat...@co.brazos.tx.us]
Sent: Tuesday, June 22, 2010 9:57 AM
To: Picher, Michael
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Use one IP address for multiple gateways in
sipX?
And this works for location (branch
I would suspect if you can figure out how to get it to do static nat
outbound translations you might fix it...
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tim Byng
Sent: Monday, June 21, 2010 10:40 AM
To: Nathaniel Watkins
Cc:
Well, what have you done in your network switch? Is QoS enabled with
DSCP 46 (ef) piped to the low latency queue?
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tran, Ly V.
Sent: Thursday, June 17, 2010 3:32 PM
To: Josh Patten;
Huh... my email address is denied access to SCS Community Portal...
won't even let me register an account...
Maybe I am evil.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Gerald
Harper
Sent: Thursday, June
Looks like there is a problem answer: 0
From iptools.com
; DiG 9.3.4-P1.1 -t MX list.sipfoundry.org
;; global options: printcmd
;; Got answer:
;; -HEADER- opcode: QUERY, status: NOERROR, id: 4629
;; flags: qr rd ra; QUERY: 1, ANSWER: 0, AUTHORITY: 1, ADDITIONAL: 0
;; QUESTION SECTION:
Open another Jira on it... I think it's an entirely reasonable request.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-
boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Sunday, June 06, 2010 8:40 AM
To: Sipx-users list
Subject:
If 3.10 was working for you I'd setup a separate 3.10 system and route ACD
calls over to it.
I've stopped using ACD because of the problems with it and there is no work
being done on the current ACD. There is work being done to replace it.
Mike
-Original Message-
From:
Have you captured traffic from the audiocodes or watched the message log
in the audiocodes?
Mike
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Philippe
Laurent
Sent: Wednesday, June 02, 2010 12:21 AM
To:
Jeff, your best bet is to file a Jira request at
http://track.sipfoundry.org so that the developers know about it.
Thanks,
Mike
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Jeff
Ferrara
Sent: Tuesday, June
Go to http://track.sipfoundry.org, create a user account and then you can open
a jira issue.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-
boun...@list.sipfoundry.org] On Behalf Of Wen Jun
Sent: Monday, May 31, 2010 10:03 PM
To: 'Tony Graziano'
I've always done a 4.0 as an intermediate step.
Have not done it with any external aliases.
Mike
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Pete
Burgess
Sent: Thursday, May 27, 2010 12:47 PM
To: 'sipXecs users'
Subject:
You can use multiple gateways. Also, Patton has a gateway device with
12, 16, 24 or 32 FXO's
http://www.patton.com/products/pe_products.asp?category=364
SmartNode 4900 series.
Mike
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-
It does with the Patton gateways I use...
-Original Message-
From: Josh Patten [mailto:jpat...@co.brazos.tx.us]
Sent: Wednesday, May 26, 2010 9:13 AM
To: Picher, Michael
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Question regarding outbound routing
So I am
The PBX can't do it... You could do it with a Patton gateway...
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Vasiliadis,
Anthony
Sent: Wednesday, May 26, 2010 3:48 PM
To: sipx-users@list.sipfoundry.org
Subject: [sipx-users] Wait or
It might be best to have the system call an extension and then record the
prompt for AA's.
The other thing that might be nice would be a prompts screen that would show
all the prompts and allow upload of custom WAV or have the system call an
extension to record.
-Original Message-
My experience with these types of things is that you have some sort of
bad data somewhere... make sure your extensions are in the proper
format (no spaces / no funky characters). Ditto to Auto Attendant
entries (I've messed these up before and had it causes all sorts of
problems).
Mike
, Michael mpic...@cmctechgroup.com;
sipx-users@list.sipfoundry.org sipx-users@list.sipfoundry.org
Sent: Tue May 18 08:09:29 2010
Subject: Re: [sipx-users] No Voice/IVR on Site-to-Site
Hello Michael,
Thank you for your reply.
On Tue, May 18, 2010 at 12:38 PM, Picher, Michael
mpic
Fairly often in larger installations...
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Mossman,
Paul (Paul)
Sent: Tuesday, May 18, 2010 1:42 PM
To: sipx-users@list.sipfoundry.org
Subject: [sipx-users] More than
I came across this one the other day... not an endorsement and haven't
tried it yet...
http://www.ictfax.org/
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony
Graziano
Sent: Wednesday, May 19, 2010 5:32 PM
To: Tran, Ly V.
I always do that after an upgrade... send phone profiles, send server
profile, etc...
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Josh Patten
Sent: Wednesday, May 19, 2010 9:01 PM
To:
Yes, this is generally an easier method to manage the system. Keep the
gateways as simple and similar as you can (sometimes that is
unavoidable... ).
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Scott
Subject: Re: [sipx-users] No Voice/IVR on Site-to-Site
I'm using IPSEC GRE and pfsense interfaces have private IPs. should I
still need NAT for that matter?
Thanks
On Tue, May 18, 2010 at 3:03 AM, Picher, Michael
mpic...@cmctechgroup.com wrote:
It should be set to manual and yes.
From
at 3:03 AM, Picher, Michael
mpic...@cmctechgroup.com
wrote:
It should be set to manual and yes.
*From:* Rhon [mailto:c4rdi...@gmail.com]
*Sent:* Monday, May 17, 2010 9:33 AM
*To:* Picher, Michael; sipx-users@list.sipfoundry.org
*Subject:* Re: [sipx-users] No Voice/IVR on Site
You can rename your yum repo file to sipxecs.repo.old... then it won't check.
Mike
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Rene Pankratz
Sent: Monday, May 17, 2010 2:44 AM
To: sipx-users
Subject: [sipx-users] Remove New
Static NAT port on the pfSense?
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Rhon
Sent: Monday, May 17, 2010 9:14 AM
To: sipx-users@list.sipfoundry.org
Subject: [sipx-users] No Voice/IVR on Site-to-Site
Hi,
I have a problem with
It should be set to manual and yes.
From: Rhon [mailto:c4rdi...@gmail.com]
Sent: Monday, May 17, 2010 9:33 AM
To: Picher, Michael; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] No Voice/IVR on Site-to-Site
Hello Michael,
I have the static NAT port set to NO on pfsense.
Also
I'd open a ticket on this one Josh... probably on the BLF prob too...
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-
boun...@list.sipfoundry.org] On Behalf Of Josh Patten
Sent: Sunday, May 16, 2010 11:28 AM
To: sipx-users@list.sipfoundry.org
I'll second that. You would need gateways that support MoIP (Modem over IP...
v.150 I think). And you won't be able to bring them in over a SIP trunk. This
would only be gateway to gateway and sketchy at best.
Run for the hills... and grab local copper.
-Original Message-
From:
Echo is not something introduced by VoIP. True echo is an analog
imbalance typically at the gateway / pots provider.
If you don't use POTS into a gateway you need to determine if this is
really echo or if this is jitter or something different.
Mike
From:
Again, echo is not a QoS problem... jitter and drop-outs are.
The Polycoms are set for default of DSCP 46 (express forwarding). You
need to enable QoS on your network switches... and make sure that DSCP
46 traffic is going into the low latency traffic queues.
If you have an HP switch
What do you mean you don't need DNS servers?
http://wiki.sipfoundry.org/display/xecsuserV4r0/DHCP+and+DNS+Server+Conf
iguration
http://wiki.sipfoundry.org/display/xecsuser/DNS+Concepts+for+sipXecs
The wiki is your friend Nalbi.
From: sipx-users-boun...@list.sipfoundry.org
I had this happen recently... I'm not 100% sure if I have nailed the
problem yet, but it may be related to outbound port randomization in
NAT. What kind of firewall are you using?
Mike
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-
Well, this may have the affect of adding a separate adapter to the PBX
and sipXecs only likes a single Ethernet interface.
Mike
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-
boun...@list.sipfoundry.org] On Behalf Of Djerk Geurts
Sent: Tuesday,
You could do it with a single gateway and dial T1/PRI interfaces (Patton
of course is my recommendation).
Multiple T1 gateway boxes however would give you a complete redundant
unit in the future.
Mike
From: sipx-users-boun...@list.sipfoundry.org
: Wednesday, May 12, 2010 9:03 AM
To: Picher, Michael; sipx-users@list.sipfoundry.org
Subject: RE: [sipx-users] Another remote worker configuration
question...
The firewall is an older SonicWall SOHO unit. It has no knowledge
of
SIP---just a basic NAT/PAT unit with stateful packet inspection. I
That's for digging for that one Robert!
Sounds like passing back through to Cisco engineering would be worth it
if anybody here has a Cisco support contract on their phones.
Mike
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
I usually just setup 8443 through the firewall and know that I have to
hit: https://iporname:8443/sipxconfig/app
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-
boun...@list.sipfoundry.org] On Behalf Of Jake Ballamis
Sent: Monday, May 10, 2010
You could do that in a Patton gateway, but the PBX itself can't do it.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Burleigh,
Matt
Sent: Monday, May 10, 2010 3:19 PM
To: sipx-users@list.sipfoundry.org
Subject: [sipx-users] Forward to
No, has nothing to do with one gateway on two rules...
But this is expected behaviorResearch multiple dial plans matching same
dial string. Discussed over, and over, and over...
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On
And you made sure 'internet dialing' was disabled right?
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Jermaine
Pinder
Sent: Friday, May 07, 2010 3:42 PM
To: tgrazi...@myitdepartment.net
Cc:
Dial in when you want to hit a live attendant during the day but roll to
an Auto Attendant at night.
Dial in when you want to ring a hunt group (can't put a DID on a hunt
group).
Controlling whether to ring to an ACD for a particular time of day or
roll somewhere else.
I also use them for
Make sure it's a 3g with 64 GB before you send it though because that's
the only one that Nate can test with :-)
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-
boun...@list.sipfoundry.org] On Behalf Of Nathaniel Watkins
Sent: Thursday, May 06, 2010
If the pbx is on the back side of a firewall I've just used port mapping
in the firewall to take care of the problem.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-
boun...@list.sipfoundry.org] On Behalf Of Staffan Kerker
Sent: Thursday, May 06,
Join the ever expanding list...
Only seems to be with ITSP's through sipXbridge and not with standard
gateways... is that right?
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-
boun...@list.sipfoundry.org] On Behalf Of Geoff Brozny
Sent:
I agree with Tony. It seems like the problem is the firewall at the customer
site. It probably has some sort of 'sip helper' enabled.
Mike
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-
boun...@list.sipfoundry.org] On Behalf Of Jermaine Pinder
The HD Polycom phones are pretty specific as to which headsets they
support. The non-hd are more accepting.
Mike
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Josh Patten
Sent: Tuesday, May 04, 2010 10:53 PM
To:
Well, the MultiTech used to have ports that could connect to FXO or
FXS... The newer units are FXO only which means you have to plug into
an FXS gateway.
The one down side to this (from a patton perspective) is that you can't
send DTMF digits after the fax picks up with an FXS gateway... so
with polycom?
Do you mean in terms of potential damage trying or it just won't work
at worse?
On Wed, 5 May 2010 05:24:07 -0400, Picher, Michael wrote:
The HD Polycom phones are pretty specific as to which headsets they
support. The non-hd are more accepting.
Mike
From: sipx-users-boun
Try Audacity...
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Michael
Scheidell
Sent: Monday, May 03, 2010 11:04 AM
To: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] how do I lower volume on MOH?
On 5/3/10 9:48 AM,
I tend to think of the one contact advantage of not using the
subdomain... that way my email, sip phone number and IM can all be the
same address...
Mike
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-
boun...@list.sipfoundry.org] On Behalf Of
Yes for the wildcard cert, and yes it is easier (from what I
understand).
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-
boun...@list.sipfoundry.org] On Behalf Of m...@grounded.net
Sent: Sunday, May 02, 2010 7:53 PM
To: sipx-users
Subject:
Andreas, also try a 'yum update yum'
Older versions of yum had troubles... specifically with proxy servers.
Mike
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-
boun...@list.sipfoundry.org] On Behalf Of Andreas (Around the Clock
Information
I don't believe that is an option...
Maybe make a feature request in the tracker
(http://track.sipfoundry.org).
Mike
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Paul Stulac
Sent: Friday, April 30, 2010 10:00 PM
To:
Calls need to route in to port 5080.
To ring to a hunt group, setup a phantom user (just a user that a phone doesn't
register to) and use forwarding on that user to direct the call wherever you'd
like (and add some scheduling if you want to hit an AA after hours).
Mike
-Original
Alright, been fighting with this for a few hours...
With the new IM integration I can get signed in with Pidgin just fine
and see status back and forth and things are working across the
interwebs as well (split DNS is working well). Will I only see on the
phone status if I have a phone that
Looks good...
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-
boun...@list.sipfoundry.org] On Behalf Of Djerk Geurts
Sent: Saturday, May 01, 2010 1:06 PM
To: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Call routing for DID from ITSP
Definitely B.
Try doing an export of the users / phones to ease the migration.
I don't think a backup will do you that much good.
Mike
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-
boun...@list.sipfoundry.org] On Behalf Of Nathaniel Watkins
Marcello,
1. You need to pick a number for their conference bridge. If your
users are 200 series, maybe make their conf bridge be the equivalent in
the 400 series.
2. When you log into the web GUI, use the user's ext. as a userid
and the PIN as the password... this will get you
Proper planning prevents *iss poor performance
From: sipx-dev-boun...@list.sipfoundry.org
[mailto:sipx-dev-boun...@list.sipfoundry.org] On Behalf Of Nikolay
Kondratyev
Sent: Thursday, April 29, 2010 1:06 PM
To: 'Tony Graziano'; 'Sipx-users list'; 'Sipx-dev list'
Subject: Re: [sipX-dev]
I like Use Internal SIP Trunking SBC or Use Internal SBC
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-
boun...@list.sipfoundry.org] On Behalf Of Mossman, Paul (Paul)
Sent: Monday, April 26, 2010 2:30 PM
To: sipx-users@list.sipfoundry.org
I'm also having good luck with 3.2.3 on 4.2.0.
Mike
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Josh Patten
Sent: Tuesday, April 27, 2010 10:22 AM
To: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Polycom Auto
Lara, I'd either keep working around them or 3 step migrate to 3.10.3,
4.0.4 and then 4.2.0. You're approaching the end of the comfortable
upgrade range. I'd start planning...
Mike
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-
Yes... HA really is just DNS SRV routing of SIP traffic. So, as long
as a phone supports SRV you should be good to go.
Mike
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-
boun...@list.sipfoundry.org] On Behalf Of Marcello Manzardo
Sent:
From what I have seen, you should be good to go with 4.2.0 sipXecs and
Polycom 3.1.3c firmware... Shared line setting not withstanding.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Matthew
Kitchin
Well, there might be a way to do that...
It would be a bit strange but if you were to make a custom dial plan entry that
you would have your users select with a different prefix. Then setup a phantom
gateway that has the callerid forced on the gateway.
Not sure if I'm being clear or not...
Yes, thanks!
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Wednesday, April 28, 2010 9:43 AM
To: Picher, Michael
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] cisco hold
I'm assuming 0.0.0.0 (rfc2543) if set to false.
If set to true, it uses RFC3264
Ah, that's right...
Well, if you were going out of a Patton gateway I could help :-)
Sorry for the confusion.
Mike
-Original Message-
From: Scott Lawrence [mailto:xmlsc...@gmail.com]
Sent: Wednesday, April 28, 2010 9:23 PM
To: Picher, Michael
Cc: M. Ranganathan; fti...@toqen.com; sipx
There's one in every crowd... just glad it wasn't me this time...
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-
boun...@list.sipfoundry.org] On Behalf Of Josh Patten
Sent: Monday, April 26, 2010 9:00 AM
To: sipx-users@list.sipfoundry.org
Also, not long ago I think somebody posted a method for extracting those
from the database so that they were unencrypted.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony
Graziano
Sent: Saturday, April 24, 2010 8:16 PM
To: Pizza
Ranga,
Am I correct in assuming that multiple users are a remote site are
hair-pinned at the sipXecs server? If not, great. If so, is there a
feature request we should be considering to keep that media path local?
Thanks,
Mike
___
Arda,
Be careful of dial plans and permissions. They are like ACL's in that
if the dial plan digits match and your permissions do not then the call
fails... even if there is a matching entry lower in the dial plan that
also matches and has the proper permissions.
As Tony pointed out the
If you're still not getting anywhere, try using wireshark to capture the
traffic from that phone to see where it is making the ntp request to.
Also, check your ntpd.conf file on the pbx if your phones are getting
time there. Pay attention to the 'restrict ...' lines.
If you make any changes in
fixed it.
Closing this thread.
Many thanks to Tony!
Rgds,
Rhon
On Sat, Apr 24, 2010 at 1:07 AM, Picher, Michael
mpic...@cmctechgroup.com wrote:
Ah, well, that never worked unless that is how the PBX is setup (not
using SRV records).
-Original Message-
From: Tony Graziano
He could have at least used a Mini in his story... geesh.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-
boun...@list.sipfoundry.org] On Behalf Of Josh Patten
Sent: Friday, April 23, 2010 11:33 PM
To: sipx-users@list.sipfoundry.org
Subject: Re:
How about a wireshark trace... this will really show everything going
on.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Rhon
Sent: Saturday, April 24, 2010 8:29 AM
To: Scott Lawrence; sipx-users@list.sipfoundry.org
Subject: Re:
Make sure you only have the sipxecs.repo in that folder.
Run a 'yum clean all'.
The re-run your 'yum upgrade'.
Mike
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-
boun...@list.sipfoundry.org] On Behalf Of Scott Lawrence
Sent: Thursday, April 22,
Cisco follows IETF standards for SIP
That right there is funny! Hahahaha Good one Nathan!
From: Nathan Nieblas [mailto:nathan.nieb...@sacatech.com]
Sent: Thursday, April 22, 2010 3:25 PM
To: Picher, Michael; Rhon; sipx-users@list.sipfoundry.org
Subject: RE: [sipx-users] Cisco and sipX
This Wiki is your friend...
http://wiki.sipfoundry.org/display/xecsuserV4r2/Bridged+Line+Appearance
Works great.
Mike
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-
boun...@list.sipfoundry.org] On Behalf Of Robert B
Sent: Thursday, April 22,
The 5400 can but the 5200 isn't listed with that capability.
-Original Message-
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Friday, April 23, 2010 7:20 AM
To: Picher, Michael; r.vanv...@raffel.nl; sipx-
us...@list.sipfoundry.org
Subject: Re: [sipx-users
How are you reaching the remote AA?
You really aren't giving us much to go on.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Rhon
Sent: Friday, April 23, 2010 8:36 AM
To: Tony Graziano; sipx-users@list.sipfoundry.org
Subject: Re:
It can be. You just either have DNS messed up or your gateways
improperly defined. It's one of those two things.
Mike
From: Rhon [mailto:c4rdi...@gmail.com]
Sent: Friday, April 23, 2010 8:47 AM
To: Picher, Michael; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Directly Call
Yes it is.
On each phone has multiple lines, in the phone config go into those
lines on each of the phones and set the ring tone.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
cyrill.rei...@iscoord.com
Sent: Friday, April 23, 2010
How are you able to actually call the remote AA.
And did you define the remote gateway as the SIP DOMAIN NAME and not as
the SIP pbx HOST NAME.
You want to use the SIP DOMAIN NAME.
Mike
From: sipx-users-boun...@list.sipfoundry.org
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