Don't you want to check mark the Register on Init? I've always found it
doesn't work unless that is marked...
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Michael
Scheidell
Sent: Thursday, August 19, 2010 2:19 PM
To: Hearty, John
Tony,
Could you transfer the call to a specific extension that has a long time
before it goes to voicemail, and then have it ring back to another extension
on the console when it isn't answered. Have the party that it is parked for
do a call pickup instead of an unpark?
Call comes into
Hello
I have sipxecs behind a pfsense router doing NAT. I had one ITSP account on it
with a 100 number range working well. I've just added a second gateway with a
second ITSP account to the system. They are both using the same bridge -
'internal SBC'. They are also both using the same ITSP - In
It just occurred to me that sipx on centos has iptables. maybe not
active, but its got it.
can I use iptables, internally, without involving natting to do
selective port forwarding.
example:
private ip address of 192.168.0.2 sipx.secnap.com.
public ip of ITSP: 4.2.2.2
I want to do somethin
On 8/19/2010 9:26 PM, Martin Steinmann wrote:
>> I went through the requirements in detail on this list, and I never
>> could find a plan that would work.
>> There were 2 huge issues.
>> 1)I never understood a way I could have a seamless transition to an fxo
>> for outbound calls if the connectio
I nominate Paul Scheppens to be our proxy to CounterPath… it seems
that he has some sort of magic touch. CounterPath at least seems to be
somewhat responsive to him.
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Thursday, August 19, 2010 1:16 PM
To: Michael Picher
Cc: Sipx
THANKS! possibly if I have the cisco rule I can transliterate it to
pfsense or pf.
On 8/19/10 10:31 PM, Matthew Kitchin (Public) wrote:
I will have to ask my network engineer tomorrow what he did. Not my
area of expertise.
--
Michael Scheidell, CTO
o: 561-999-5000
d: 561-948-2259
ISN: 125
I will have to ask my network engineer tomorrow what he did. Not my area of
expertise.
-Original Message-
From: Michael Scheidell
Date: Thu, 19 Aug 2010 22:27:46
To:
Cc: sipx-users@list.sipfoundry.org users
Subject: Re: [sipx-users] port 5060/ port 5080, proxy why?
just to conclude:
o
just to conclude:
on cisco, you can tell it that if one certain source ip hits port 5060
on your public sip ip, it redirects it to the private ip on port 5080.
then EVERYONE else hitting port 5060 goes to the internal port 5060?
(I know pfsense can't do this nativity, I tried)
On 8/19/10 10:2
>
> I went through the requirements in detail on this list, and I never
> could find a plan that would work.
> There were 2 huge issues.
> 1)I never understood a way I could have a seamless transition to an fxo
> for outbound calls if the connection to the central office went down.
Audiocodes GWs
I'm not sure what would work for your situation. I don't have this issue. I was
just throwing an idea out there.
-Original Message-
From: Michael Scheidell
Date: Thu, 19 Aug 2010 21:48:05
To:
Cc: sipx-users@list.sipfoundry.org users
Subject: Re: [sipx-users] port 5060/ port 5080, proxy
On 8/19/10 9:46 PM, Matthew Kitchin (Public) wrote:
Maybe NAT 2 external IPs to it. One that does 5060->5080 and one that
doesn't. Whatever needs to hit it on 5060 would use the IP with no
translation.
as I suspected. tcp connection is one way.
telnet ip2 5060 does indeed to private 5080, but
trying that now, but the reflection (internal ip back out) would use,
what public ip?
On 8/19/10 9:46 PM, Matthew Kitchin (Public) wrote:
Whatever needs to hit it on 5060 would use the IP with no translation.
--
Michael Scheidell, CTO
o: 561-999-5000
d: 561-948-2259
ISN: 1259*1300
> *| *SECNA
I went through the requirements in detail on this list, and I never could find
a plan that would work.
There were 2 huge issues.
1)I never understood a way I could have a seamless transition to an fxo for
outbound calls if the connection to the central office went down.
2)I never understood a
Why not a centralized deployment with only phones and optional gateways in
the remote office?Having to manage 110 small ITX boxes does not sound
pretty.
--martin
> -Original Message-
> From: Matthew Kitchin (Public) [mailto:mkitchin.pub...@gmail.com]
> Sent: Thursday, August 19, 2010 9
Go to system / servers and create a new host. Assign the SIP Trunking role
to the new host and not the main one.
Fixing XX-8692 should not be a big project. Dale likely will remember the
details. It was almost finished in an earlier cycle. See here:
http://track.sipfoundry.org/browse/XX-4818
Maybe NAT 2 external IPs to it. One that does 5060->5080 and one that doesn't.
Whatever needs to hit it on 5060 would use the IP with no translation.
-Original Message-
From: Michael Scheidell
Date: Thu, 19 Aug 2010 21:43:55
To:
Cc: sipx-users@list.sipfoundry.org users
Subject: Re: [si
it can do port translation, as in if it sees ANYTHING hit port 5060 it
can translate it to an internal 5080.
it can't do this:
itsp.public hits sipxpublic:5060 it goes to sipxprivate:5080
everyone else hits port 5060 it goes to sipxprivate:5060
On 8/19/10 9:39 PM, Matthew Kitchin (Public) wr
Using 2 hosts (sipxbridge on a dedicated one) was the other option we looked
at. I didn't do it for 2 reasons. I was a total novice and wanted to keep
things simple. And, our corporate office was the model we would follow at our
110 small remote locations. We wanted to do small mini itx (on a bb
can';t run it on different NICS' as the JIRA says.
how do I run sipx on a different host?
On 8/19/10 9:37 PM, Martin Steinmann wrote:
That is not what I meant. If you run the proxy and sipXbridge on
different NICs or different hosts, then they both can use port 5060
--
Michael Scheidell, CT
I don't send them SIP on 5080 if that is what you mean. Every trunk I have with
Verizon has had a different port for outbound.
As far as why, I asked about changing what ports Polycoms talked to Sipx on
sometime early this year. I was told it was almost possible with 4.04 (at the
time) and may
That is not what I meant. If you run the proxy and sipXbridge on different
NICs or different hosts, then they both can use port 5060
--martin
From: Michael Scheidell [mailto:michael.scheid...@secnap.com]
Sent: Thursday, August 19, 2010 9:33 PM
To: Martin Steinmann
Cc: sipx-users@list.sipfoun
I can simulate that with pfsense, anything hitting ip1:5060 goes to
sip:5080 anything ip2:5060 goes to sip:5060. but I assume the reverse
would be a problem.
the phones and itsp won't like traffic coming from an ip that they
didn't connect to.
On 8/19/10 9:30 PM, Martin Steinmann wrote:
The
The easy solution to this problem is this:
http://track.sipfoundry.org/browse/XX-8692. A quick fix would be to run
sipXbridge on a separate host.
The reason two ports are uses is that trunk calls come on over sipxbridge
and and the proxy listens on port 5060.
--martin
From: sipx-user
it doesn't look like pfsense can do this specifically. so I am going to
try to add two rules (4 I guess)
two natting rules and two firewall rules.
the last question everyone at level 3 asks is WHY.? (you do know that
this is some of the issues with firewalls, QOS services, etc. they
don't exp
Because it listens for phones on 5060.
I translate everything inbound from my carrier on 5060 to 5080. Works
perfectly.
-Original Message-
From: Michael Scheidell
Sender: sipx-users-boun...@list.sipfoundry.org
Date: Thu, 19 Aug 2010 21:15:22
To: sipx-users@list.sipfoundry.org users
Sub
Lost in several old emails is 'why' sipx needs to have trunk calls come
in on port 5080 and not 5060.
Several ITSP's insist that they will only send the calls on port 5060
since that is the standard.
I am working with Level3 on a different issue than the first one, and it
involves a different SI
The messages point to email. I think updating one version at a time is
prudent. I would have tested at 4.20 before updating to 4.21.
In turning off email messaging, I might also remove user information as a
test, then re-enter it upon enabling to see of that gives any success.
==
The major changes were made to iser xml files for unified messaging. Perhaps
disable/enable that on a user will yield some success.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control
What is happening her?
The system was recently updated to 4.2.1...
The update to 4.2.1 was done because it was available and because for some
vague reason 4.2.0 stopped forwarding calls. It worked nico a few weeks ago.
During my holiday it started failing.
Standard error
* java.lang.NoSuc
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Jos?? Tudela de la Rosa
[pptud...@gmail.com]
SIP/2.0 420 Extension Not Supported
From: "John
Doe"mailto:1235...@example123.com>;tag=2f7000-6f01a8c0-13c4-45
Content-Type: text/plain;
charset="utf-8"
Content-Transfer-Encoding: 8bit
Organization: SipXecs Forum
In-Reply-To:
X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <50721>
Message-ID:
Good day,
I have what appears to be a problem similar to René's.
There seems to be a
missing voicemail paramet
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Stiles Watson
[wat...@datatek-net.com]
Anyone have any suggestions on how to get 6730i to dial another ext
other than 0 & 101?
_
On 8/19/10 5:07 PM, Hearty, John wrote:
I was assisting in troubleshooting problems doing SIP Trunking to
sipXecs. When accessing the service over public internet, we require
digest authentication. When a User name for SIP authentication is
configured and the "Use default asserted identity"
What version and build are you using?
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 Contract Customers:
http
I was assisting in troubleshooting problems doing SIP Trunking to sipXecs.
When accessing the service over public internet, we require digest
authentication. When a User name for SIP authentication is configured and the
"Use default asserted identity" box is checked, the value in the "User nam
I think on the conferences themselves you could set MoH to the sound card?
On Thu, Aug 19, 2010 at 4:00 PM, Austin Curry wrote:
> I have a request to remove all MOH at a site.
>
> .
>
> Rather than removing the URL from each phone, I uploaded an empty wav file
> and deleted the other files and
On Wed, Aug 18, 2010 at 4:48 AM, Krisztian Ganyai <
krisztian.gan...@onrelay.com> wrote:
> Hi,
>
>
>
> I would insert it in the onWebUi() function of /usr/bin/sipxconfig.sh as a
> new line somewhere after the “exec $JavaCmd \”. I have 4.2 vanilla and the
> suitable line numbers are 201-204 there.
I have a request to remove all MOH at a site.
.
Rather than removing the URL from each phone, I uploaded an empty wav
file and deleted the other files and restarted the park and conference
services.
It works for all MOH instances except the conferences I have setup.
A couple of question
Is
I totally understand, it is nice to see how "it could be" done, but the
valet parking IMO breaks things further, as it becomes "invisible" which (I
think), makes it less useable overall.
Is there not a way to view BLF, etc. to valet parking to make them visible
under FS?
On Thu, Aug 19, 2010 at 3
Yes... we had a bit of a discussion about that.
I guess the problem right now is that the park orbit is all still written on
the old media services so there's low desire to make modifications to it
where the effort would be better put to use getting over to a freeswitch
based solution.
Mike
On
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Michael Scheidell
[michael.scheid...@secnap.com]
If I use the built in sipxbridge on a ITSP gateway, and make in inbound call to
a user who has call forwar
On 8/19/10 12:13 PM, Gerald Drouillard wrote:
If the call comes in via a uri ( j...@example.com ) instead of via the
registration with the ITSP does it get the 1 or 70?
I don't know.. my cell phone can't dial URI's
But I don't think sipx should be arbitrarily setting the max-forwards to
+1 on the Bria control proposal
This has burned me also and huge waste of time.
Matt Burleigh
EII Global, Inc.
http://www.eiisolutions.net
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony
Graziano
Sent: Thursday, August 19
I have wasted around 80 hours tracking issues with every single version of
the bria phone with them. not one helpful hint, clue or resolution from
them. I think their support system really only works for very large
customers.
Now I have to try to convince them to make 3.0 available again... such
On Thu, Aug 19, 2010 at 11:30 AM, Tony Graziano <
tgrazi...@myitdepartment.net> wrote:
> but.. yet another reason i do subdomains for sipx anymore...
>
>
Ya, I'm starting to see the light on that one. Too bad godaddy isn't more
friendly with SRV records on subs.
__
but.. yet another reason i do subdomains for sipx anymore...
On Thu, Aug 19, 2010 at 12:30 PM, Jim Canfield wrote:
> On Thu, Aug 19, 2010 at 11:16 AM, Tony Graziano <
> tgrazi...@myitdepartment.net> wrote:
>
>> In the wiki...
>>
>> http://wiki.sipfoundry.org/display/xecsuserV4r2/DNS+File+Generat
On Thu, Aug 19, 2010 at 11:16 AM, Tony Graziano <
tgrazi...@myitdepartment.net> wrote:
> In the wiki...
>
> http://wiki.sipfoundry.org/display/xecsuserV4r2/DNS+File+Generation
>
> dns master
>
Thanks Tony
DNS_MODE="Manual"
Should do the trick! I had forgotten about this change.
-Jim
_
In the wiki...
http://wiki.sipfoundry.org/display/xecsuserV4r2/DNS+File+Generation
dns master
On Thu, Aug 19, 2010 at 11:03 AM, Jim Canfield wrote:
> Hey Folks.
>
> Where would I need to DNS changes in sipXecs so they are permanent. I just
> noticed changes in the /var/named/domain.zone file
On 8/19/2010 11:56 AM, Michael Scheidell wrote:
> On 8/19/10 10:58 AM, Tony Graziano wrote:
>> Sipx thinks its a gateway, so the call
>> sets up differently.
>>
> any way you know of to set the max-forwards to, 70 instead of 1?
>
>
If the call comes in via a uri ( j...@example.com ) instead of via
no. i have not had that issue with any of the itsp's or trunks i normally
use though.
On Thu, Aug 19, 2010 at 11:56 AM, Michael Scheidell <
michael.scheid...@secnap.com> wrote:
> On 8/19/10 10:58 AM, Tony Graziano wrote:
>
> Sipx thinks its a gateway, so the call
> sets up differently.
>
>
> an
On 8/19/10 10:58 AM, Tony Graziano wrote:
Sipx thinks its a gateway, so the call
sets up differently.
any way you know of to set the max-forwards to, 70 instead of 1?
--
Michael Scheidell, CTO
o: 561-999-5000
d: 561-948-2259
ISN: 1259*1300
> *| *SECNAP Network Security Corporation
* Ce
Hey Folks.
Where would I need to DNS changes in sipXecs so they are permanent. I just
noticed changes in the /var/named/domain.zone file are being overwritten on
server restart. I'd like to add a few A records.
-Jim
___
sipx-users mailing list
sipx-use
External sbc's handle the siptrunk. Sipx thinks its a gateway, so the call
sets up differently.
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
it actually, it works if you don't need autentication. no p-asserted, etc.
(I haven't looked at packet traces yet to see any differences, and, yes,
the max-forwards was 70.)
only time max-forwards isn't 70 is when I select an sbc route that isn't
sipxbridge.
I was wondering why sipx decides
(which is what the discussion says, how polycom uses alert header for
intercom & paging, can also be used for this item... examples are already in
the comments)
On Thu, Aug 19, 2010 at 10:09 AM, Josh M. Patten wrote:
> Let us not forget that most phones, Polycom included, support distinctive
> ri
Let us not forget that most phones, Polycom included, support distinctive ring
by signaling. An example of how to do this in Asterisk is here:
http://www.technicallyamusing.com/?p=44
If I'm not mistaken the intercom system already uses this.
From: sipx-users-boun...@list.sipfoundry.org
[mailto
I was under the understanding you were testing against an independent SBC
device of some sort. if that is not the case, then ignore my statement.
When we use independent SBC's, we use them for trunking and remote users and
sipx is not behind nat or supporting remote users. The dialplan can point
pretend I am stupid, since I have tried dozens of things to get this to
work.
are you saying don't use an SBC? create an unmanaged gateway through
'gateways'?
NOT a SIP trunk?
I still use sipxbridge as my SBC?
this doesn't give the option of using authentication? (I guess the sbc
didn't
I dug out http://track.sipfoundry.org/XX-5641 and added the original dev
email discussion to it, which shows it might be feasible to use an alert
header (like paging, etc.) to notify a nortel or polycom phone that a call
park ringback is occuring by changing the ring tone and defining it in the
sys
I would try removing that if you have a dial plan and just add the IP as an
unmanaged gateway and send it the calls that way and see if the max-forwards
resolves...
On Thu, Aug 19, 2010 at 9:35 AM, Michael Scheidell <
michael.scheid...@secnap.com> wrote:
> I went into 'sbs route', add new sbc ro
I went into 'sbs route', add new sbc route (it is an unmanaged gateway,
didn't know any other)
I had done it earlier when level 3 wanted me to send to port 5070
nothing I could do on the ITSP gateway page would do that.
(wonkeyness as described by an ezuce person)
On 8/19/10 9:31 AM, Tony Graz
What SBC? Is it setup as an unmanaged gateway or sip trunk (I actually
prefer unmanaged gateway)
On Thu, Aug 19, 2010 at 8:52 AM, Michael Scheidell <
michael.scheid...@secnap.com> wrote:
> If I use the built in sipxbridge on a ITSP gateway, and make in inbound
> call to a user who has call forw
If I use the built in sipxbridge on a ITSP gateway, and make in inbound
call to a user who has call forwarding enabled via sipx, I see
max-forwards=70. (good)
If I change the route to use a seperate SBC, normal calls go out with
max-forwards=70, but external calls forwarded via sipx show
max-fo
On Thu, Aug 19, 2010 at 9:00 AM, Michael Picher wrote:
> Well, you did say (hold, park, etc...).
>
> I don't know that there's a way to do that with park. Park orbit timeout /
> 0-out just go back to the park-er.
>
Right, which is why I suggest a way to 'inject'...
>
> Distinctive ring is done
I was installing a new plugin for labels.
___
sipx-users mailing list
sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archive/sipx-users/
Well, you did say (hold, park, etc...).
I don't know that there's a way to do that with park. Park orbit timeout /
0-out just go back to the park-er.
Distinctive ring is done by line on the phones, so you'd have to return to a
different line to make that happen. The other way I could see would
On Thu, Aug 19, 2010 at 2:43 AM, Staffan Kerker wrote:
>
> On 18 aug 2010, at 16.26, Douglas Hubler wrote:
>
>> On Wed, Aug 18, 2010 at 7:28 AM, Staffan Kerker wrote:
>>> I'm having almost the same problem. I added a "ciscoplus 7975" phone as a
>>> device and suddenly I'm getting
>>> "Internal Er
Yes, but call park is not "hold". I am trying to uderstand if there is a way
to have call park do something similar.
On hold is done at the phone. I was hoping there was a way to inject a
spefici ring back pattern, tone, or even display message when a call back is
sent to a phone from specific ex
create a tracker account, login, go to tracker item, pick 'more options'
menu drop down and select 'vote'.
On Thu, Aug 19, 2010 at 8:41 AM, Mike Haun wrote:
> I really like that feature. How do I vote?
>
> On Thu, Aug 19, 2010 at 6:52 AM, Staffan Kerker wrote:
>
>> > To create a request, go to
Posted something on their forums:
http://forums.counterpath.com/viewtopic.php?f=29&t=17186
Have you called them yet?
On Wed, Aug 18, 2010 at 10:00 AM, Michael Picher wrote:
> I can confirm this. I was actually running 3.1.1 which was operating fine
> and now with 3.1.2 they broke it again..
>
I really like that feature. How do I vote?
On Thu, Aug 19, 2010 at 6:52 AM, Staffan Kerker wrote:
> > To create a request, go to track.sipfoundry.org and create a login.
> Next,
> > create an issue as a new feature.
>
> Thanks,
>
> http://track.sipfoundry.org/browse/XX-8661
>
> Now vote! =)
>
>
I think that they Polycoms have a hold reminder setting...
On Wed, Aug 18, 2010 at 7:29 PM, Tony Graziano wrote:
> One of the features I see on LOTS of PBX systems is a "ring back" pattern
> (hold, park, etc.).
>
> I think it might be possible to implement a "distinctive" ring/pattern for
> call
> To create a request, go to track.sipfoundry.org and create a login. Next,
> create an issue as a new feature.
Thanks,
http://track.sipfoundry.org/browse/XX-8661
Now vote! =)
/Staffan
--
Staffan Kerker
mail/sip/xmpp. staf...@kerker.se
___
sipx-u
To create a request, go to track.sipfoundry.org and create a login. Next,
create an issue as a new feature.
On Wed, Aug 18, 2010 at 4:02 PM, Staffan Kerker wrote:
>
> On 18 aug 2010, at 15.58, Michael Picher wrote:
>
> > This would be an awesome feature request.
>
> So, how do we do this? What w
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