On Tue, 11 May 2010 19:54:30 -0700, Todd Hodgen wrote:
> I think there have been a few people that struggled with that model in the
> past, you might want to look at the archives.
I've seen some of those posts and have also talked with many who say avoid the
phones. Thing is, we had them in use w
I think there have been a few people that struggled with that model in the
past, you might want to look at the archives.
I was afraid my response would offend. Really, it's important that we start
putting out some details that will help others grow from this project with
their own learning curve
Thanks Todd, this is great information. I was just looking for a 'ya, does that
for me too' or 'known to do this' as this is secondary to phones working to
begin with.
However, I'll follow your input and see if I can gather more info which might
help solve the problem or at least provide input
Have you done a trace on these calls to see what is going on?
You might want to do a trace of a call that works, and a call that doesn't.
Get a big table, print it out and do what we call a stare and compare. The
issue should become apparent.
Or, copy your two traces into siptrace and see where
Seem to have a problem with Caller ID setting but might be the phones
themselves.
X-lite client - works both with and without the Caller ID setting. Can hear
the dial tones and ringing on outbound calls on remotes.
Remote Linksys SPA-941 - caller ID setting with user's full phone number - wi
Engineering hat on...
From: Nathaniel Watkins [mailto:nwatk...@garrettcounty.org]
Sent: Tuesday, May 11, 2010 1:59 PM
To: Todd Hodgen; sipx-users@list.sipfoundry.org
Subject: RE: [sipx-users] Linking sipx to another phone system via T1
Todd:
A couple issues/fears with using SIP trunks as
I think I've come to my own conclusion that the Bria 3.0 is not able to be
properly provisioned using the cmc plugin. Nor does presence seem to work.
The 2.5 is not available for sale any longer. How much does that suck?
Just a friendly heads up till the folks at Counterpath work through this and
On Tue, May 11, 2010 at 5:02 PM, Stephen D. Miller
wrote:
>
> Great feedback...answers/comments below:
>
> 1. Interestingly, I configured the intranets on the sip server to
> include only 172.20.1.0/24. The wiki recommends removing the defaults
> and adding back an entry that is representative of
Am a Patton reseller.
What i find I can do with the patton is light years ahead of some of the
others though.
I am getting ready to test the following:
Century Link PRI (emergency failover if D channel is gone) -->> SIP trunk
provider which will route inbound calls, if sip trunk fails to see va
Great feedback...answers/comments below:
1. Interestingly, I configured the intranets on the sip server to
include only 172.20.1.0/24. The wiki recommends removing the defaults
and adding back an entry that is representative of your network. Is
there a config file or db that I can probe t
Tony - do you get a kick back for selling those pattons :)
I've only used the audiocodes analog gateways - but if you're willing to help
me configure, I am willing to give it a shot. I'm assuming 'run of the mill'
wasn't aimed at any particular product line...
From: Tony Graziano [mailto:tgra
Todd:
A couple issues/fears with using SIP trunks as our main connection:
1) I have a slight political game to play - our 911 center is physically
located at the courthouse as well - and we enjoy a very good relationship with
our Verizon techs. They are very accommodating with us as we are
Look at you big boy!
The patton 4960 can have multiple T1/E1 interfaces and supports signalling
types for all kinds of stuff. It's also highly configurable as opposed to
the run of the mill stuff out there.
On Tue, May 11, 2010 at 4:44 PM, Todd Hodgen wrote:
> You might consider bringing in
I'm getting ready to write a proposal to replace our existing phone systems
with a sipxecs implementation.
I'm envisioning purchasing 2 T1 gateways - 1 for bringing dial tone in - the
other for linking to our old phone system (during the transition period) - then
re-using the 2nd one as we e
You might consider bringing in SIP trunks for your service, via your
internet connection(s), and a single T1 to your legacy PBX. This will give
you a redundant path for today, and when the Legacy PBX is gone, it can
still be your redundant path, yet you will see the significant cost savings
of usi
On Tue, 11 May 2010 13:32:47 -0700, Dave Deutschman wrote:
> We have a customer who has updated the kernel to use PAE. They are running
> sipXecs 4.0.4 without any issues.
I'm running 4.2 in this case but all of my servers are identical. One would
think it should be ok. I'm just nervous about ge
I'm getting ready to write a proposal to replace our existing phone systems
with a sipxecs implementation.
I'm envisioning purchasing 2 T1 gateways - 1 for bringing dial tone in - the
other for linking to our old phone system (during the transition period) - then
re-using the 2nd one as we expa
>
> Posting failedcase.pcap for review.
>
> Here's a brief on the players in the trace:
> ---
> Extension 18 (166.137.14.58) : iphone (remote sip soft phone)
> Extension 13 (172.20.1.213) : office (local sip hard phone)
> Sipx server (172.20.1.102)
>
He's in that there nightmare place.
I think that's an idea for an improvement request. Show the user caller id
or branch when dialing 911 for this group, and disable callerid blocking
when that kind of call is made... One group and tell it to use user callerid
in the account, assign everyone to t
On 11 maj 2010, at 20.27, M. Ranganathan wrote:
> On Tue, May 11, 2010 at 2:02 PM, Staffan Kerker wrote:
>> Hi
>>
>> I just noticed that sipXbridge seems to remove the crytpo-attributes in SDP
>> if a call setup with SRTP information
>> is recieved. The RTP/SAVP profile is still there, but the
On Tue, May 11, 2010 at 2:02 PM, Staffan Kerker wrote:
> Hi
>
> I just noticed that sipXbridge seems to remove the crytpo-attributes in SDP
> if a call setup with SRTP information
> is recieved. The RTP/SAVP profile is still there, but the crypto attributes
> are gone...
Since sipxbridge/sipxr
> of using a PAE kernel. I usually use a PAE kernel when using more than 4GB
> of mem. I have 4GB on this server, should I update to the PAE?
I see that kernel-PAE is available. Now I can't recall if one needs it to see
up to 4GB or beyond. I was sure it was beyond?
Any problem with updating to P
Hi
I just noticed that sipXbridge seems to remove the crytpo-attributes in SDP if
a call setup with SRTP information
is recieved. The RTP/SAVP profile is still there, but the crypto attributes are
gone...
--- SDP in INVITE sent to sipXbridge (outgoing call)
v=0
o=- 1273600686040637 127360068
I've got a server that just suddenly crashed. It was rebuilt last weekend so is
pretty new.
The last main errors I see in the messages log is a lot of journal aborted
messages.
I've pasted a snip of the log in case anyone is curious or catches something I
didn't. There are the drive errors bu
Strictly speaking from an IP and operating system perspective:
In most Linux distro's routing is turned off by default. This means that
if the machine has multiple IP subnets it can reach all of them, but
will not pass packets between them for other devices.
The question what IP address SIPX proc
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Austin Curry
[aus...@dnsms.com]
Can anyone shed some light on from the trace below?
No; from the trace of a single
I've posted (about an hour ago) a pcap file of the traffic involved. It
was ~400K in size so I don't know if it got flagged somehow...I haven't
seen it show up over on forum.sipfoundry.org yet either.
Let me know if there's an alternate/preferred method to distribute the
capture file and I'll be
Greetings,
Can I create a VLAN e.g. etho.200 and assign a static IP address (172.0.200.1)
Can I have eth0 with an IP of 192.168.1.1 (default because VLAN will not load
without etho)
Can i have the DHCP server assign IP to Polycom phones via CDP or do I have to
manually assign the VLAN at boot?
It is now hard-coded to "2", which I believe is "on with sipx adaptations"
Scott.
From: Josh Patten [mailto:jpat...@co.brazos.tx.us]
Sent: Tuesday, May 11, 2010 11:15 AM
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Polycom "Transfer on Proceeding"
That setting used to be under th
Tony’s earlier points notwithstanding, perhaps you could address your
problem by establishing separate “branches/gateways” pointing to
unique subaccounts with your ITSP for each user requesting anonymous
caller ID. Of course, if you have a very large user base, this quickly
becomes an administrati
I am recieving a "500 Internal Server error" from my sipx server when
attempting to place outside calls from a branch. Can anyone shed some light on
from the trace below? It occurs after a 100 trying, The phone recives a busy
signal.
No. TimeSourceDestination
Hi,
I'm installing SipXecs from source on my Debian Lenny. After
installing all needed dependencies, and succesfull './configure[...]'
my instalation fail in 'make build' step.
I followed this manual:
http://sipx-wiki.calivia.com/index.php/Building_from_source
(downloaded source by svn co
http://
That setting used to be under the SIP configuration section. Has it been
moved or removed in 4.2? Is it now defaulted to on?
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 5/11/2010 9:34 AM, Scott Richesson wrote:
It finally seems to be working! (for me
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Organization: SipXecs Forum
X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <46473>
Message-ID:
if someone is willing to can you send me a screenshot of
your vitelity sip trunk screen ( block out your secret info)
I jus
>
> I have a newly installed sipxecs 4.2 system (replaced a
> previously installed v3.10 system). System is behind a NAT
> firewall. All works as expected (internal extension to
> extension calls, outbound calls, inbound calls, etc). I have
> also followed the v4.2 remote worker config guide
know 3.2.2 is known to have issues and 3.2.3 is recommended.
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 C
It finally seems to be working! (for me anyway) (Polycom SIP 3.2.2, sipx 4.2)
Scott Richesson
Cincinnati Fan
From: Josh Patten [mailto:jpat...@co.brazos.tx.us]
Sent: Tuesday, May 11, 2010 10:19 AM
To: sipx-users@list.sipfoundry.org
Subject: [sipx-users] Polycom "Transfer on Proceeding"
Was the
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Organization: SipXecs Forum
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Message-ID:
you will need to have an analog gateway like Audiocodes.
you cannot hook up digital phones to sipxdo you m
Is it possible to get a wireshark of the call at either end. Acrobits
probably cannot be configured for the media ports, but not being able to see
why rtp doesnt establish just leads to conjecture.
Have you tried to get a siptrace of the call?
I am thinking the wireshark or siptrace will give a c
Given that the Acrobits softphone registers with the server successfully
and receives responses from the INVITE, I think it's safe to assume 5060
isn't being blocked by AT&T in this area.
All call-handling related traffic (REGISTER, INVITE, etc) appears to be
flowing between the endpoints corre
On Tue, May 11, 2010 at 9:46 AM, Staffan Kerker wrote:
> Hi
>
> This question may be stupid, but is TLS for SIP supported directly in the
> sipXproxy now, without using the sipXbridge?
> A simple netstat doesn´t show me any sipXproxy listening to port 5061... Any
> special tweaks needed?
Hi.
Was the Polycom "Transfer on Proceeding" issue ever resolved?
By "Transfer on Proceeding" I mean:
* User hits transfer
* User dials number
* While call is ringing (proceeding) user hits transfer again to
blind transfer the call.
I remember this had issue a while back. It would be
Is you provider blocking port 5060? Might consider connecting via vpn first?
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Stephen D. Miller
Sent: Tuesday, May 11, 2010 10:02 AM
To: sipx-users@list.sipfoundry.
I have a newly installed sipxecs 4.2 system (replaced a previously
installed v3.10 system). System is behind a NAT firewall. All works as
expected (internal extension to extension calls, outbound calls, inbound
calls, etc). I have also followed the v4.2 remote worker config guide
and configured
Hi
This question may be stupid, but is TLS for SIP supported directly in the
sipXproxy now, without using the sipXbridge?
A simple netstat doesn´t show me any sipXproxy listening to port 5061... Any
special tweaks needed?
/Staffan
--
Staffan Kerker
mail/sip/xmpp: staf...@kerker.se
"Don't get
Describe your gateway or trunk. It sounds as if there is an issue with DTMF
not being recognized. Different gateways have to use different settings, but
more importantly you need to understand on the first and second calls which
codec is negotiated and what your gateways DTMF method is. Normally is
Hi everyone,
I'm currently having problems w/ the sipx auto attendant. Upon boot up of
the sipx server, the auto attendant works just fine. When i call from the
pstn and reach the AA and dial an extension, it sends me there. Upon hanging
up and calling the AA from the pstn again, it no longer re
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