Dear All,
Is IPv6 already supported in SIPx ?
Best,
Michael HUANG
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On Wed, Mar 10, 2010 at 3:10 AM, sipxuser sipx michael.s...@gmail.comwrote:
Dear All,
Is IPv6 already supported in SIPx ?
Best,
Michael HUANG
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When a softphone user places a call on hold to an outside caller, should
sipxbridge be providing MOH assuming it is enabled there? MOH is provided by
the AA during the transfer process, but not when an individual MOH is
enacted.
I expect an answer like when the softphone supports
Polycom IP550 and Polycom IP650 phones
Bootrom: 4.2.0.0310
Firmware: 3.1.3.0439
sipXecs: 4.0.4
Phones are on the same subnet as the sipXecs server. Calls are placed through
a SIP Trunk via an ITSP.
The ITSP is a CLEC with Class 4/5 switches (i.e. we can provide dial-tone just
like the LEC).
On Wed, Mar 10, 2010 at 8:19 AM, Burden, Mike m...@lynk.com wrote:
Polycom IP550 and Polycom IP650 phones
Bootrom: 4.2.0.0310
Firmware: 3.1.3.0439
sipXecs: 4.0.4
Phones are on the same subnet as the sipXecs server. Calls are placed
through a SIP Trunk via an ITSP.
The ITSP is a CLEC
Ignore this. I didn't have use local ip address set ad need to test
against a hardphone for this example. Sorry.
On Wed, Mar 10, 2010 at 3:05 AM, Tony Graziano tgrazi...@myitdepartment.net
wrote:
this is the pcap from my remote user firewall, for packets destined to the
sipx server ip
On Tue, Mar 9, 2010 at 10:25 AM, Francis Tinio fti...@toqen.com wrote:
Thanks!
I'll be in my client's office later and should be able to troubleshoot
further.
Thanks again
On Mar 9, 2010, at 10:23 AM, Robert Joly wrote:
three phones so that their registration process gets captured and
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Wednesday, March 10, 2010 8:36 AM
To: Burden, Mike
Cc: Scott Lawrence; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] DTMF Issues Revisited
A real siptrace of a call that DTMF did not work properly on with a
I had an opportunity earlier than expected, so I made calls to two of
our Customers and ran traces.
ftp://ftp.lynk.com/DTMF.xml
ftp://ftp.lynk.com/DTMF2.xml
Mike Burden
Lynk Systems, Inc
e-mail: m...@lynk.com
Phone: 616-532-4985
From:
I am somewhat doubtful that the DTMF setting are at issue here. I say that
because whether through an Ingate, PRI or sipxbridge I have never seen this.
All the above does is indicate someone pressed a key a bunch of times.
Is a siptrace available?
Tony
Tony,
His trace is actually
What would be really interesting trace wise is the SDP headers showing
the offer/response for telephony events (from the invite and the ack).
If the ITSP is accepting these, at 101 even, and then not acting on them
on it sure seems like they're broken. If it were me I would take the
trace of a
Is it possible to get Sipx to send both the caller's phone number as well as a
15 character string of user name (or whatever) when placing outbound calls to
an ISP? Or is there only the 15 character string (which could contain either
text or a phone number)?
From Wikipedia:
There are two
On Wed, Mar 10, 2010 at 2:15 PM, Jeff Gilmore j...@thegilmores.net wrote:
Is it possible to get Sipx to send both the caller's phone number as well
as a 15 character string of user name (or whatever) when placing outbound
calls to an ISP? Or is there *only* the 15 character string (which
I think the key is that the delivery of name and number is a dip into the
central directory database. Most carriers charge for access to that
database as an option, as a type maintenance fees to the directory database.
can also provide the directory listed name for the particular number.
On Wed, 2010-03-10 at 14:15 -0500, Jeff Gilmore wrote:
Is it possible to get Sipx to send both the caller's phone number as
well as a 15 character string of user name (or whatever) when placing
outbound calls to an ISP? Or is there only the 15 character string
(which could contain either text
Oops, I meant to send this to the whole group.
Begin forwarded message:
From: Pizza Napoletana pizzai...@gmx.com
Date: March 10, 2010 12:59:25 PM PST
To: Jeff Gilmore j...@thegilmores.net
Subject: Re: [sipx-users] Outgoing caller ID name number?
My ITSP says that there is no way for them
There are two types of caller ID, number only and name+number. Number
only caller ID is called Single Data Message Format (SDMF), which
provides the caller's telephone number, the date and time of the call.
Name+number caller ID is called Multiple Data Message Format (MDMF),
which in addition to
Content-Type: text/plain;
charset=utf-8
Content-Transfer-Encoding: 8bit
Organization: SipXecs Forum
In-Reply-To: a892.4b980...@forum.sipfoundry.org
X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 43159
Message-ID: a897.4b981...@forum.sipfoundry.org
Sipx is not designed to work like that. You
On 3/10/10 4:54 PM, Burden, Mike wrote:
It appears that for incoming calls, our ITSP will only deliver SDMF to us.
You usually have to pay for CNAM access.
voip.ms charges 1/c per lookup (but cache it for a week)
also, if you want your DID to show up in CNAM, your ITSP has to submit it.
--
On Wed, Mar 10, 2010 at 4:54 PM, Burden, Mike m...@lynk.com wrote:
*There are two types of caller ID, number only and name+number. Number
only caller ID is called Single Data Message Format (SDMF), which provides
the caller's telephone number, the date and time of the call. Name+number
I am having trouble registering with Telasip ITSP.
Below is what their Tech Support told me:
. sipX seems to use the P-Asserted-Identity header for
authentication, which we cannot support.
. The userid must be in the from header, and callerid must be in the
Remote-Party-ID
On Wed, Mar 10, 2010 at 5:15 PM, Marcello Manzardo marce...@discsox.comwrote:
I am having trouble registering with Telasip ITSP.
Below is what their Tech Support told me:
· sipX seems to use the P-Asserted-Identity header for
authentication, which we cannot support.
Devices,
There are multiple databases as well. I get stale call information from
VOIP.ms as an example, yet current from Broadvox. There is either multiple
databases, or they don't all update as frequently.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org]
An other option might be to use the directory in the polycom's -- it will
convert the number to a speed dial (or regular) directory entry and display
that info when the phone rings. With 4.0.4 you can also hack up some .vm files
in sipxecs and make distinctive rings on the polycoms (based on
On Wed, Mar 10, 2010 at 5:15 PM, Marcello Manzardo marce...@discsox.com wrote:
I am having trouble registering with Telasip ITSP.
Below is what their Tech Support told me:
· sipX seems to use the P-Asserted-Identity header for
authentication, which we cannot support.
·
Hello,
I am trying to locate the sipx file that contains the presence status for
the phone?
Mike
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Thank you very much for the suggestions.
I was able to trace a REGISTER request followed by a 200 OK which seems to
be ok.
Still having trouble with making outbound calls...
I will post back with the configuration settings if I ever get it to work.
Kind regards,
Marcello
-Original
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