I am following the guide of Setup sipXecs to sipXecs Calling.
It tells me to use 'site-to-site' dial plan type. While I have to use prefix
to route A site call to B site, such as 55+ B site ext., how shall I make
A site sipx drops the prefix when deliver call to B site sipx ? I've
observed the
I guess I don't understand your tunnel outside of the pfSense box.
Unless this diagram is inaccurate.
If you'd like to send be a visio or an true picture how this is
configured I might be able to help.
Mike
From: sipx-users-boun...@list.sipfoundry.org
If the resulting call is set to send matching suffix, the 55 is dropped.
If you were to trace the call or look at the proxy logs, you will see only
the suffix is sent to the far end.
Prefix 55 suffix x digits, resulting call send matched suffix.
On Wed, May 19, 2010 at 5:03 AM, Wen Jun
Either you create the tunnel on pfsense or ...
The cisco routers are also on your private network, in which case add a
gateway and route on pfsense to use the cisco routers to connect those 2
sites.
No filters in pfsense should be needed in a vpn setting.
Tony
On 5/18/2010 1:42 PM, Mossman, Paul (Paul) wrote:
Hi all,
I understand that you could have a second AutoAttendant dial plan rule. This
allow you to have second AA Extension, with distinct Default/Working/Holiday
attendant configuration.
How often are multiple AutoAttendant dial plan
Hi Everyone,
I manage to fix our problem. It was a NAT issue. And setting pfsense to NO
NAT takes care of the problem.
Thank you all for your help.
Best regards and have a nice day!
Rhon
On Wed, May 19, 2010 at 12:05 PM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
Either you create
How is the other way around behaving, as in a 4.0.4 box calling the 4.2.0
box?
-
MM
On Tue, May 18, 2010 at 14:39, Tony Graziano
tgrazi...@myitdepartment.netwrote:
I am testing an ipsec connection and routing calls from a 4.0.4 system and
a 4.2 system via a site-to-site rule.
Both my sipx
I stopped at 4.2 dialing to 4.0.4. I've done 4.0.4 to 4.0.4 before without
issue.
I plan to do some upgrades, one at a time, this stops be from doing some
work in production in 4.2 until the nat traversal is explained. Nat
traversal in 4.2 is being called only when I use the sipdomain as the
-Original Message-
From: Mossman, Paul (Paul)
Sent: Tuesday, May 18, 2010 10:19 PM
To: Saint, David (David); sipx-users@list.sipfoundry.org
Subject: RE: More than one VM dial plan rule? (XX-7822)
Dave wrote:
I use two Voicemail dial plan rules when creating a private network
We have a minor problem with the setup with our current ITSP that uses
+1 as the prefix. The way we have it configured now is that we have +1
as the prefix on the gateway for this ITSP. Our dial plan is 10 digits
for all calls, -local, LD and Toll Free. The problem we have is all the
incoming
Yes.
http://www.myitdepartment.net/support/Three_things_I_really_like_about_sipXecs_4.pdf
Go to page 7 and look at dial plans. There you will find the solution under
the return calls dial plan entry which strips the +1 before it is sent to
the gateway.
On Wed, May 19, 2010 at 10:35 AM, Tran, Ly
Have you tired creating a 1+XXX the XXX being local area codes and have the
resulting call drop the 1? Then if a person dials 1 or not for a local call it
would go through.
ie: for greater Vancouver:
Prefix = 1604 and 7 digits
Resulting call = dial 604 and append matched suffix
Prefix = 1778
I think you miss his point. Dialing from the missed calls on the handset
shows the +1 (and 10 digits) on the display. The outbound call fails because
+1 is ALSO being added at the gateway. Not all carriers do this, not all
phones display the +1 properly either.
His question is a simple one, and
I would concur. We depend on this functionality. The only way it could
work for us is if personal auto attendant support a complex config like
the full blown AA.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
Thanks Tony, I believe this should solve our problem and will configure
it as shown in your document. Can't do now during business hours, but
will later this evening.
Ly Tran
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Wednesday, May 19, 2010 10:26 AM
To: Gerald
I understand.
EVEN if you are using siptrunking, making a dial plan change during working
hours will not disrupt any in progress calls (only proxy and registrar are
reloaded).
This type of change is designed to be flexible and implemented during the
work day. If you were making a change to the
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Alfred Campbell
Sent: Wednesday, May 19, 2010 3:54 AM
To: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] More than one AA dial plan rule? (XX-7822)
On
In trying to understand where and when CPU spikes were occurring, I
noticed an oddity in the following process (this is not something I'm asking
anyone to do in a production environment)...
In disabling the default IM account and renaming it to something more
meaningful (from 200 to jdoe), then
Because you also like to use the term square line :-P
***Proceeds to get off your lawn***
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 5/19/2010 1:10 PM, Tony Graziano wrote:
On a lark (I like to use that term a lot, don't know why)
I'm running sipx 4.2 with sipbridge and bandwidth.com as a service provider the
phones firmware are 3.23 and bootrom 4.22 and a pfsense firewall. I'm having
an issue where they calls are being dropped in the middle of a call.
Has anyone experienced this?
thanks
When this has happened to me as well, it's simply a matter of packet
loss. sipX for some reason is extremely sensitive to packet loss when it
comes to the signaling and it likes to send BYE whenever something isn't
100% perfect.
Josh Patten
Assistant Network Administrator
Brazos County IT
Content-Type: text/plain;
charset=utf-8
Content-Transfer-Encoding: 8bit
Organization: SipXecs Forum
In-Reply-To: b645798cb7dd6643910a368dd9cc8d8fa83...@lynkvm-mail01.lynk.com
X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 46893
Message-ID: b72d.4bf44...@forum.sipfoundry.org
A new chapter in
Asterisk versions 1.4 and lower tend to do weird things with DTMF unless
certain precautions are taken. The best thing you could do at this point
is try to find an ITSP that uses Asterisk 1.6 and up or one that doesn't
use Asterisk.
Josh Patten
Assistant Network Administrator
Brazos County IT
As well, there needs to be some consideration in how you setup any
firewall with a siptrunk that is also an Internet connection for generic
use, so I use a bandwidth shaping script for pfsense to give bandwidth to
voice over data/sipxconfig stuff..
On Wed, May 19, 2010 at 3:46 PM, Josh Patten
Content-Type: text/plain;
charset=utf-8
Content-Transfer-Encoding: 8bit
Organization: SipXecs Forum
In-Reply-To: 4bf444f7.6020...@co.brazos.tx.us
X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 46895
Message-ID: b72f.4bf44...@forum.sipfoundry.org
Josh,
We are *NOT* using Asterisk. See the
I used your script to give bandwidth for UDP 3-31000, 5080 and 5060.
-Original Message-
From: Tony Graziano [tgrazi...@myitdepartment.net]
Date: 05/19/2010 04:12 PM
To: Josh Patten jpat...@co.brazos.tx.us
CC: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] IP 550 dropping
Ah, Josh is implying (correctly) that a lot of ITSP's use Asterisk. You can
see the UA version in your failed transactions.
Get another ITSP. There are a lot of them out there that seem to work well.
On Wed, May 19, 2010 at 4:13 PM, Michael W. Burden m...@lynk.com wrote:
Content-Type:
Good.
When you look at your RRD graphs look at the quality for high latency...
should tell you plenty if it is a general internet connection issue.
On Wed, May 19, 2010 at 4:15 PM, Jermaine Pinder
jpin...@pinderconsulting.com wrote:
I used your script to give bandwidth for UDP 3-31000,
Content-Type: text/plain;
charset=utf-8
Content-Transfer-Encoding: 8bit
Organization: SipXecs Forum
In-Reply-To: aanlktineupupxgufwcl2m6gps6rldpdszfuuozp_c...@mail.gmail.com
X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 46901
Message-ID: b735.4bf44...@forum.sipfoundry.org
Ah! Gotcha. Sorry
Using PRI and mediant 1000, so no ITSP
I do have a Trixbox system in production at a small office and they
sometimes have DTMF troubles, hence my comment.
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 5/19/2010 3:23 PM, Michael W.Burden wrote:
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Josh Patten
[jpat...@co.brazos.tx.us]
When this has happened to me as well, it's simply a matter of packet
loss. sipX for some reason is extremely
You would see surely see dropped calls if the latency with an internet
connection becomes so high it is problematic. Refer to your RRD graph for
quality. Look for red spikes.
I think 3.2.3 is not ready for primetime. Unless SAa is absolutely necessary
drop back to 3.1.3 (IMO).
Has anyone here installed the Hylafax fax server in a SipX environment?
We have a project in need of automating the testing of faxing at
different locations and would like a standalone faxserver. This might
be a dumb question, but we have a single sip trunk with multiple call
paths on the SipX
Noone had done this as t.38 support is somewhat lacking.
Does your itsp supports t.38?
Create a user and forward it to a standalone hylafax server with their DID
as the pointer.
I have a FS appliance I've built and have to get back to it to do just this
scenario.
On 5/19/10 4:54 PM, Tony Graziano wrote:
You would see surely see dropped calls if the latency with an internet
connection becomes so high it is problematic. Refer to your RRD graph for
quality. Look for red spikes.
I think 3.2.3 is not ready for primetime. Unless SAa is absolutely necessary
My question would be, why? Why not just build a separate server and keep things
simple?
On Wed, 19 May 2010 16:00:20 -0500, Tran, Ly V. wrote:
Has anyone here installed the Hylafax fax server in a SipX environment? We
have a project in need of automating the testing of faxing at different
No. 3.2.2 had issues with SAA in general. 3.2.3 does not work well in remote
cases. I have a standalone ethernet bridge with nothing on it except 2 650's
and 3.2.3 sends all calls to VM.
3.1.3 is safe. I don't know where you get your information. Mine is from
testing.
Its a good place to be now. Separate.
There is a tracker or community request to add fax via FS, which would
detect a fax call to your extension and receive it without ringing your
phone. I think it might be a time coming, hence keeping it separate.
Tony Graziano,
One other thought I can offer is that asterisk can be used as the switch if
you've got your trunk coming into that. Depending on the extension the call is
coming into or whether it's a fax, calls can be routed to sipx, or another pbx
or a fax server.
On Wed, 19 May 2010 16:00:20 -0500, Tran,
Sorry, that's what I meant.. a separate server with asterisk + hylafax
in an environment with an existing SipX PBX. The current ITSP does
support T.38 faxing. I will give it a try, just need to dig some more
in how to build it. Thanks!
Ly Tran
-Original Message-
From:
Sorry, that's what I meant.. a separate server with asterisk + hylafax
That is very doable and as I mentioned, you could even use the asterisk box as
a switch if you have different DID's on your single trunk.
in an environment with an existing SipX PBX. The current ITSP does
support T.38
create a phantom user, call forward to the userid in hylafax like
1...@1.2.3.4
but i never got hylafax to work with t.38 worth a doo doo.
On Wed, May 19, 2010 at 5:26 PM, Tran, Ly V. lt...@rrtgi.com wrote:
Sorry, that's what I meant.. a separate server with asterisk + hylafax
in an
On 5/19/10 5:15 PM, Tony Graziano wrote:
3.1.3 is safe. I don't know where you get your information. Mine is from
testing.
from this mailing list.
http://www.mail-archive.com/sipx-users@list.sipfoundry.org/msg12954.html
Um. - don't believe everything I read on the internet. There is nothing in
3.1.3 that won't work with the exception of SAA. It works fine in 4.0.4
and 4.2. Latency issues cause 3.2.3 to be somewhat unuseable even in some
lan environments.
I would not consider 3.2.2 safe, but I would consider
I'm about to integrate a Rightfax setup with Sipx. 2 separate machines,
but one sip trunk. My fax volume in this case is too low to warrant a
dedicated trunk. The calls will come into sipx and be routed to the
Rightfax server. I understand the part about the phantom user. I don't
know how to
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
Me too.
-Main one
-HR
-Help desk
Sent via BlackBerry from T-Mobile
-Original Message-
From: Picher, Michael mpic...@cmctechgroup.com
Date: Wed, 19 May 2010 19:55:24
To: Mossman, Paul (Paul)paulmoss...@avaya.com;
sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] More than one AA
Hi Folks,
I just tried to use the invite participant feature from the admin
console and it doesn't appear to be working. URI, e164, and local all
seem have the same result...nothing. Before I start digging, can
anyone confirm this is working in 4.2?
Thanks,
-Jim
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.
Once I'm done deploying sipX at my employer I'll probably have 50 auto
attendants.
Do NOT remove multiple auto attendants. I already have a hard enough
time managing the ever changing recordings for these things.
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
It appears this was solved this by sending profiles to all servers...not
sure what caused it.
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 5/15/2010 11:49 AM, Josh Patten wrote:
I did the 4.0.4 - 4.2 upgrade last night and everything appears to have
I have snapshots but they're obviously too big for the list. Should I
create a ticket or email to someone off list?
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 5/18/2010 12:39 PM, WORLEY, Dale R (Dale) wrote:
On 2010-05-18 14:00, Mossman, Paul (Paul) wrote:
Hi all,
How often are multiple Voicemail dial plan rules actually used?
I can think of one reason... You have some 3 digit extensions, and some 4
digit extensions, but you want both 8XXX and 8 to go directory to the XXX
(or )
On 2010-05-19 11:26, Tony Graziano wrote:
I think you miss his point. Dialing from the missed calls on the
handset shows the +1 (and 10 digits) on the display. The outbound call
fails because +1 is ALSO being added at the gateway. Not all carriers
do this, not all phones display the +1
I just entered my conference bridge from an extension, and was able to do an
invite out to a cell phone with no issues. 4.2.1 build.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Jim Canfield
Sent: Wednesday,
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:
The funny thing is I already hadguess I just needed to do it again.
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 5/19/2010 8:41 PM, Picher, Michael wrote:
I always do that after an upgrade... send phone profiles, send server
profile, etc...
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
On Wed, May 19, 2010 at 8:35 PM, Todd Hodgen thod...@verizon.net wrote:
I just entered my conference bridge from an extension, and was able to do an
invite out to a cell phone with no issues. 4.2.1 build.
Thanks Todd. I'll see what I did wrong.
___
Hi Tony. I've got a similar problem. 4.0.2 to 4.0.4 works fine, 4.0.4
to 4.0.4 works fine. DNS SRV records in place and fully resolvable.
Site-to-Site dial plan using sip domain names etc. When I try 4.2 to
4.0.4 I experience the same problem as you. No NAT involved on a fully
routable LAN. I
One other thing Tony. I don't have the Sip Trunking role configured on
both the 4.2 and 4.0.4 system, so no internal SBC. Can you test without
SIP Trunking role on your systems?
Dave
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
I have not seen anyone say they had any problems with sipx 4.2 and
polycom 3.1.3 (that weren't there with 4.0.4). My plan is to upgrade
sipx from 4.0.4 to 4.2 (that will fix my only major issue) and then
probably wait for polycom 3.3 to see if that will have all major
outstanding issues
62 matches
Mail list logo
