> From: sipx-users-boun...@list.sipfoundry.org
> [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
> Sent: donderdag 10 januari 2013 14:06
> To: Discussion list for users of sipXecs software
> Subject: Re: [sipx-users] Strange behavior in dial plan regardin
Can you blind transfer the call from a different User Agent (i.e. Polycom
not bria)? Are you sure the ITSp is sending the calls in via port 5080? Can
you call the auto attendant from the outside and transfer a call? If you
cant call the AA and transfer to an extension they are probably sending the
>
> Telecats BV
>
>
>
>
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Telephone: 434.984.8430
s
#x27;ve
>> installed stunnel that comes w/sipxecs and not from centos5.
>> ___
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>> sipx-users@list.sipfoundry.org
>> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>
> _
o doesn’t seem to be sipx related?
>
> Thanks.
>
>
>
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Telephone: 434.984.8430
sip: tgrazi...@voice.myitdep
gt; Kind regards,
>> Met vriendelijke groet,
>>
>>
>> Elwin Formsma
>> Telecats BV
>> -
>> Elwin Formsma | Telecats bv | KvK Enschede 06069106 | Tel: 053 488 99 44 |
>> Fax: 053 488 99 10 | E-mail: e.form...@telecats.nl |
>>
>> _
By the same token you might be able to craft a rule (dial plan) to redirect
or null process that destination. It would be simpler to turn the ringer
off and simply ignore the calls and add the line to an admin assistant who
would field answering those calls.
--
LAN/Telephony/Security and Control
onaws.com
> ",nonce="448038dd4fbe27452d637428dc1a565950e76936",uri="sip:
> ec2-50-18-193-48.us-west-1.compute.amazonaws.com
> ",qop=auth,nc=0001,cnonce="35993a6f",response="ad299e3bc739b3d5882f39ea7a64e04f",algorithm=MD5
> Expires: 3600
, from your computer.
> Registered in England No 04006093 ¦ Registered Office 1st Floor, 236 Gray's
> Inn Road, London WC1X 8HB
>
>
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I tend to think its a basic Linux install issue with your server hardware.
If you choose to do the minimal install consider doing it from the network
if you have sufficient bandwidth.
Did you query the Google oracle for its wisdom in these matters as it
relates to you hardware platform?
On Dec 27,
ry.org Port 80
--
~~
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
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Us
> Malibu, California
> 424.781.1666 itsp/fax
> Yuma, Arizona
> 928.597.4777 itsp/faxmatthewe...@audiopivotpatc.com
>
>
> "pure technical artistry"
>
>
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> sipx-users@list.sipfoundry.org
>
424.781.1666 itsp/fax
> Yuma, Arizona
> 928.597.4777 itsp/fax
> matthewe...@audiopivotpatc.com
>
>
> "pure technical artistry"
>
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> List Archive: htt
g
> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>
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sip: tgrazi...@voice.myitdepartment.net
Fax: 434.465.6833
~~
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Ask
luck.
> --
> Tommy Laino
> Dome Technologies
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Is port 5060 exposed in the firewall? If so it is potentially a script
from outside trying to abuse your system you would need to inspect your
logs to verify.
On Dec 16, 2012 10:52 AM, "Tommy Laino" wrote:
>
>
> For the last 2 weekends I have had a SipXecs 4.4 with 30
> Polycom 335's that has al
Tel: 053 488 99 44
> | Fax: 053 488 99 10 | E-mail: e.form...@telecats.nl |
>
>
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>
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T
Did you follow the guidelines in the wiki on hunt groups?
On Dec 13, 2012 3:46 PM, "glomos-info" wrote:
> Hi,
>
> ** **
>
> 1st problem
>
> If i create a hunt group containing some extensions ringing at the same
> time, then when i pick up a phone the connection is established, but soun
Its not a big deal. Very glad it is working. This us one if those instances
where a change log would come in handy in deed.
On Dec 12, 2012 4:33 PM, "Tommy Laino" wrote:
>
>
> Tony I am embarrassed lol I know better and I take full
> responsibility for my stupidity.
>
> I changed 2 phones and sen
ions.
>>>>
>>>> We have a TA908e 2nd gen running AOS A5.02.00.E. We currently have
>>>> not noticed any issue with having an external caller forwarded to and
>>>> external number. cell => user for 30sec => external number.
>>
ipx-users@list.sipfoundry.org
> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>
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~~
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
Fax: 434.465.6833
~~
Linked-In Profile:
http://www.linkedin.com/pub/tony-
> AJ
>
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>
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Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
Fa
ally and didnt use the auto
> provision (another brain dead moment on my part).
> --
> Tommy Laino
> Dome Technologies
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les. Do you think
> that could have been part of the problem?
> --
> Tommy Laino
> Dome Technologies
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> List Archive: http://list.sipfoundry.org/archive/sipx-use
to revert this?
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sip: tgrazi...@voice.myitdepartment.net
Fax: 434.465.6833
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Using or
I suggested top not the Mrtg statistics. Are toy using any swap?
On Dec 11, 2012 5:40 PM, "Tommy Laino" wrote:
>
>
> I do not see either of those logs in my snapshot. I checked
> the statistics for the server and I am not having any issues
> with resources
> --
> Tommy Laino
> Dome Technologies
>
Also co spider looking g at top and seeing if you have. Resource issues in
general.
On Dec 11, 2012 5:17 PM, "George Niculae" wrote:
> On Wed, Dec 12, 2012 at 12:15 AM, Tommy Laino wrote:
>
>>
>>
>> I got SipXecs 4.4 with all Polycom 335/550 phones with 3.2.6
>> firmware. Everything was working
>
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>
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~~
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.
as the sip domain name for the ITSP.
Once I removed this it actually resolved itself.
Sorry for the false alarm!
On Tue, Dec 11, 2012 at 12:01 PM, Joegen Baclor wrote:
> Highly important that you confirm that this works for you pre dec-5
> update.
>
>
> On 12/12/2012 12:01 AM
his should not be possible because the rule 2 requires
> permissions.
>
>
> Met vriendelijke groet,
>
>
> Elwin Formsma
> Telecats BV
> -
> Elwin Formsma | Telecats bv | KvK Enschede 06069106 | Tel: 053 488 99 44
> | Fax: 053 488 99 10 | E-mail: e.form.
creating INVITE at SipUtilities.java:936
The logging in debug really doesn't show anything I saw as meaningful. Can
anyone replicate this?
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~~~~~~
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
Fax: 434.465.6833
~~
Linked-In Profile
I think it is well known that you should not park or upoark calls using the
BLF button on a Polycom.
On Dec 7, 2012 1:07 PM, "Bryan Anderson" wrote:
> So I noticed some talk in a previous email "Call forward fails to
> external number" about the Adtran 900 series. I have a couple of comments
> a
, "Laurentiu Ceausescu" wrote:
> On Fri, Dec 7, 2012 at 2:10 PM, Douglas Hubler wrote:
>
>> On Fri, Dec 7, 2012 at 6:50 AM, Tony Graziano
>> wrote:
>> > Sounds like the wiki needs a how to page on 4.4 to 4.6 migration...
>> >
>> > Backup.
>> >
sense in this discussion anyway...
>
>
> On Fri, Dec 7, 2012 at 6:36 AM, Tony Graziano <
> tgrazi...@myitdepartment.net> wrote:
>
>> I think it would help a lot if sipxbridge could send a failure message to
>> the proxy so the proxy could try another gateway (i.e. put
Sounds like the wiki needs a how to page on 4.4 to 4.6 migration...
Backup.
Install 4.6
Restore
Perform superadmin password reset
Login and use
On Dec 7, 2012 6:03 AM, "Michael Picher" wrote:
> I am seeing that we are going to have to have giant flashing neon letters
> on the restore page... ug
f Technical Services
> eZuce, Inc.
>
> 300 Brickstone Square
>
> Suite 201
>
> Andover, MA. 01810
> O.978-296-1005 X2015
> M.207-956-0262
> @mpicher <http://twitter.com/mpicher>
> linkedin <http://www.linkedin.com/profile/view?id=35504760&trk=tab_pro>
> www.ezuce.com
>
>
> -
Director of Technical Services
> eZuce, Inc.
> 300 Brickstone Square
> Suite 201
> Andover, MA. 01810
> O.978-296-1005 X2015
> M.207-956-0262
> @mpicher <http://twitter.com/mpicher>
> linkedin <http://www.linkedin.com/profile/view?id=35504760&trk=tab_pro>
> www.ezuce.com
>
>
> ---
og/audiocodes-mp-202b-httpsfax-adapter). I'm
>> curious if you (or anyone else reading) have tested the 202B while we're on
>> the topic. My apologies if that's considered thread jacking.
>>
>> ____
>> From: sipx-users-bo
__
> sipx-users mailing list
> sipx-users@list.sipfoundry.org
> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>
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~~
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
Fax: 434.465.6833
And put sipxivr and sipxproxy in debug log Lev, send a fax And inspect the
logs including sipxconfig.to see if there are errors.
On Nov 30, 2012 7:24 PM, "Tony Graziano"
wrote:
> Do a search for the DID number and see if it is in multiple places on the
> system.
> On Nov 30, 2
Do a search for the DID number and see if it is in multiple places on the
system.
On Nov 30, 2012 6:21 PM, "Tommy Laino" wrote:
>
>
> Yes I do hear the fax tone and the sending machine says that
> the transmission was successful.
> --
> Tommy Laino
> Dome Technologies
> __
red
on the unified messaging page.
On Nov 30, 2012 5:39 PM, "Tony Graziano"
wrote:
> The user is 12345 and their unifies messaging has a fax mailbox of 54321
> and a fax DID ON THE UNIFIED messaging page of 2223334567. Right?
> On Nov 30, 2012 5:33 PM, "Tommy Laino" wrote
The user is 12345 and their unifies messaging has a fax mailbox of 54321
and a fax DID ON THE UNIFIED messaging page of 2223334567. Right?
On Nov 30, 2012 5:33 PM, "Tommy Laino" wrote:
>
>
> The user is extension 5011. The fax extension is 6011. The
> fax DID is set to the T.38 fax number. What d
Depending on what patch version you are on this might actually go out as a
PDF file. The user account needs to have the D ID number from the provider
put on their fax account in Unified Messaging. This means you need to
create both a fax box and 50 ID number on the Unified Messaging page for
that u
gt; sipx-users@list.sipfoundry.org
> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>
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>
--
I think the firmware on the phones limits this to (reliably) a max of 12
devices.
On Nov 30, 2012 11:41 AM, "Bryan Anderson" wrote:
> Hello,
>
> We have an office of ~83 endpoints all but one phone are Polycom
> SoundPoint IP 331's, the receptions uses a 650 with 1 sidecar.
>
> The firmware's ar
; Thank you
>
> Daniel Peinado López
>
>
> El 28/11/2012, a las 00:29, Tony Graziano
> escribió:
>
> > If it matters to anyone, I seem to recall there were structural or DB
> > changes between 4.2 and 4.4. If that is the case (I could be wrong but
> > I'
Elite Partner for EMEA
>>
>> IANT is Member of GROUPLINK
>>
>> ___
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>> sipx-users@list.sipfoundry.org
>> List Archive: http://list.sipfoundry.org/archive/sipx-users/
> _
6794 450
> Fax: +49 (5331) 6794 499
> Mail: daniel.pein...@iant.de
> Web: www.iant.de
>
>
> IANT is eZuce Elite Partner for EMEA
>
> IANT is Member of GROUPLINK
>
> ___
> sipx-users mailing list
> sipx-users@list.sipf
If they are not using any pots or catb services is the actiontec, replace
it since it would be acting simply as an Ethernet router. Good luck!
On Nov 25, 2012 4:06 AM, "Todd Hodgen" wrote:
> I suspect your issue is with Verizon, and not the router. They are
> blocking your ability to send emai
Could item be moved where folks could comment on it?
I've thought at first maybe the user could request a pull from the user
portal, but that didn't make sense since the UA does that already.
The only thing that makes sense is that the config gets auto pushed for
that phone type of a user change
t; directly - we have direct support from Yealink R&D so even custom firmware
>> is often no problem.
>>
>> 2012/11/21 Michael Picher
>>
>>> I think it's in 4.7 now... somebody will pipe up and let me know if I'm
>>> wrong.
>>>
&
disallow certain
call types if desired.
On Nov 21, 2012 7:46 AM, "Michael Picher" wrote:
> huh, a new use I didn't know about :-)
>
> now supporting DISA !
>
> not sure if that's a good thing, but if folks want to open themselves up,
> so be it.
>
>
>
You use *81 as an AA destination for option "9". Then use the authcode like
anyone inside would. That is the while purpose of it. Read the wiki page.
On Nov 20, 2012 10:41 PM, "Todd Hodgen" wrote:
> You could set up multiple extensions with call forwarding to a particular
> number. Ext 111 - 20
Which is why I'm suggesting what you are asking for is not a transfer from
as to dual a pstn number.
See ... USERS>AUTHORIZATION CODES
the wiki has a page on it too.
On Nov 20, 2012 9:41 PM, "文军" wrote:
> The PSTN I want to transfer is not a fixed number.
>
>
&
You would have to set the pstn # as the "option" for 9. In other words
pressing 9 might transfer the call to 5551234.
I think what you are trying to do is use the authorization codes and that's
not part of the as. See the wiki if this is the case.
On Nov 20, 2012 8:46 PM, "文军" wrote:
> Is there
response I got but wanted to check first.
>>
>> ** **
>>
>> I completely agree that going with Polycoms would solve all of our
>> problems.
>>
>> ** **
>>
>> Thanks for the info/insight.
>>
>> ** **
>>
>&g
_
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>
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~~
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Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
Fax: 434.465.6833
~~
Linked-In Prof
**
>
> *Thanks,*
>
> *Mark W. Wood*
>
> *office:* (760)202-0224 X2010
>
> *Direct: *(760)459-1981
>
> *[image: New Image.BMP]*
>
> www.redphonetech.com
>
> ** **
>
> ** **
>
> * *
>
> ** **
>
> ___
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> sipx-users@list.sip
>
> ___
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>
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~~
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartmen
It was and pushed into patch 23 posted today...
On Mon, Nov 19, 2012 at 11:15 AM, Michael Picher wrote:
> I'm not so sure that those were pushed back to 4.4...
>
>
> On Mon, Nov 19, 2012 at 10:29 AM, Tony Graziano <
> tgrazi...@myitdepartment.net> wrote:
>
>
/sipx-users/
>
>
>
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>
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Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi..
s a serious issue.
>
> ~Noah
>
>
> On Nov 16, 2012, at 7:30 PM, Tony Graziano
> wrote:
>
> That is with ssh open or available from the outside.
>
> I still suggest a JIRA...
> On Nov 16, 2012 6:41 PM, "Noah Mehl" wrote:
>
>> I would also like to
pport issues please email supp...@a-1networks.com or call 707-703-1050
>
> On 11/16/12 3:17 PM, Tony Graziano wrote:
>
> can you provide the output of: lsof -i | grep LISTEN
>
> and post what SMTP is listening to?
>
>
>
> On Fri, Nov 16, 2012 at 6:11 PM, Noah Mehl
t I've illustrated, the port is tunneled via SSH. Then on
> the remote machine (the sipxecs server) the traffic originates as
> LOCALHOST. That's why it's a OOTB security flaw.
>
> I have not made changes to the smtp config.
>
> ~Noah
>
> On Nov 16, 2012, a
my
> arsenal (ssh port forwarding), and I don't want to change the default
> passwords (because of provision stock phones). But I HIGHLY suggest
> everyone takes a quick look at their settings, because I bet a lot of
> people are susceptible to this. Thanks.
>
> ~Noah
&g
//list.sipfoundry.org/archive/sipx-users/
>
>
>
>
>
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>
--
~~
Tony Grazian
ess to the machine.
>
> ~Noah
>
> On Nov 16, 2012, at 1:48 PM, Tony Graziano
> wrote:
>
> Fwiw I can test the exploit and my ids (commercial snort rules).
>
> so polycom provisioning in Sipx will cease using ftp and the user account
> will be removed at that time and
t; $telnet localhost 25
>
> Tell me if your ids stops that?
>
> This works on a stock SipXecs 4.4.0 install.
>
> ~Noah
>
> On Nov 16, 2012, at 11:46 AM, Tony Graziano >
> wrote:
>
> The user doesn't have login via ssh. Ssh in and of itself is not
>
But there again SMTP is for some reason open on that machine and unless you
are also using it as a mail server I don't see the point in making it
available to the public at large. Send mail does not need to have SMTP open
in order to send. This is yet another thing that confuses me about your
firew
utes with the help of Google
> http://www.semicomplete.com/articles/ssh-security/:
>
> $sudo ssh -vN -L25:localhost:25 PlcmSpIp@sipxecsip
> $sudo ssh -vN -R25:localhost:25 PlcmSpIp@sipxecsip
> $telnet localhost 25
>
> Tell me if your ids stops that?
>
> This works on a st
uld
> be denied from ssh OOTB. This is also, not documented as far as I can
> tell...
>
> ~Noah
>
> On Nov 16, 2012, at 11:26 AM, Tony Graziano
> wrote:
>
> It really sounds like you don't have a method to harden your server if
> you are exposing it. Its entir
mat of this file. (search for access_db in that file) # The
> >> /usr/share/doc/sendmail/README.cf is part of the sendmail-doc # package.
> >> #
> >> # by default we allow relaying from localhost...
> >> Connect:localhost.localdomain RELAY
> >> Conn
I don't see port 123 anywhere so I guess it is not being allowed.
On Nov 15, 2012 5:36 PM, "Kyle Haefner" wrote:
> Hi All,
>
> Finally getting around to putting phones on my fresh install of openUC
> 4.6. If I have the firewall disabled the phones get time from the sipx
> cluster. If I have the
configuration of anything related to the PlcmSpIp
> user. It does however make me feel better that it is related to the vsftpd
> service and the polycom phones.
> >
> >> From /etc/passwd:
> >
> >
> PlcmSpIp:x:500:500::/var/sipxdata/configserver/phone/profile/
01101110 0010 01100010 01101001 01101110 0111
> 01110010 0001 00101110
>
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>
--
~
quot;2012-08-03T09:31:03.817728Z":43815:KERNEL:DEBUG:testpbx.labsip2ser.net:
> SipClientTcp-30:22CEF700:SipXProxy:"OsSocket::isReadyToWrite
> > poll returned 1 in socket: 21 0x7f5eec002070"
> >
> > I hope this helps..
> >
> > bye
> > Domenico Ch
It needs two interfaces. It can sit behind a firewall. Both interfaces can
be numbered differently (same man).
On Nov 12, 2012 5:56 PM, "Chris Rawlings" wrote:
> i have been reading through the Karoo bridge setup cookbook.. wow
>
> so i was wondering if anyone had any sample configs that would wo
could probably move to a FS based park orbit...
>>
>> I'd love nothing better... :-)
>>
>>
>>
>> On Mon, Nov 12, 2012 at 1:06 AM, Tony Graziano <
>> tgrazi...@myitdepartment.net> wrote:
>>
>>> http://track.sipfoundry.org/browse/XX-8
>>>>
>>>> On Mon, Nov 5, 2012 at 8:51 AM, Melcon Moraes wrote:
>>>>
>>>>> Can we simply drop in a new FS RPM package to replace the 1.0.7
>>>>> version that comes with sipXecs and all the current working stuff will
>>>>> c
http.log(date) then
> sipregistrar.log and then sipxlogwatcher.log-(date) then sipxpage.log
>
> ** **
>
> --****
>
> Geoff
>
> ** **
>
> ** **
>
> ** **
>
> *From:* Tony Graziano [mailto:tgrazi...@myitdepartment.net]
> *Sent:* Friday, Novembe
it.
>
>
> ** **
>
> Thanks in advance.
>
> ** **
>
> --
>
> Geoff
>
> ___
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> sipx-users@list.sipfoundry.org
> List Archive: http://list.sipfoundry.org/archive/sipx-use
gt; Thanks in advance.
>
> ** **
>
> --
>
> Geoff
>
> ___
> sipx-users mailing list
> sipx-users@list.sipfoundry.org
> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>
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~~
Tony Grazia
There is a setting for the user to define a different voicemail system
(foreign). You should try this with a single user and then apply it to a
user group if it works.
On Nov 9, 2012 7:20 AM, "Gael Ravot" wrote:
> Hello all,
>
> I am trying to use sipXecs as a voicemail system for extensions on
>
Ugh.
http://www.sendmail.org/~gshapiro/8.10.Training/DaemonPortOptions.html
On Nov 7, 2012 8:20 PM, "Tommy Laino" wrote:
>
>
> I just had a thought that had not occured to me until just
> now. Verizon FiOS in this area blocks port 25. All my
> outbound mail servers use port 587. How do I change
mpicher <http://twitter.com/mpicher>
> linkedin <http://www.linkedin.com/profile/view?id=35504760&trk=tab_pro>
> www.ezuce.com
>
>
>
> There are 10 kinds of pe
t; ----
>
>
> There are 10 kinds of people in the world, those who understand binary and
> those who don't.
>
> ** **
>
> ___
Probably not. Sip bridge does not have the ability to determine to use
another gateway. The proxy only knows a REAL unavailable code and then can
use the alternate gateway.
Sip bridge doesn't have the ability to properly determine the availability
or status of a gateway or a call and relay "kill t
That's the directory. There is a file there "sip domain.zone". You need to
edit the appropriate zone file in that directory.
Good luck.
On Nov 6, 2012 4:24 PM, "Tommy Laino" wrote:
>
>
> Mike,
> When I enter the command to edit the file /var/named/ I get
> an error that says "Illegal File Name".
SSL issue. Does anyone have any ideas on how to
figure this out?
On Mon, Nov 5, 2012 at 2:16 PM, Tony Graziano
wrote:
> I am looking at a strange issue with a system which had a drive failure.
> We replaced the drive and reloaded (did not restore) the system, then
> updated it to
se have an
install similar and can verify whether they are seeing this or not?
--
~~
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
Fax: 434.465.6833
~~
Linked-In Profile:
http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
Ask
An unmanaged gateway is just that. Can I assume that the address for both
systems are on the same subnet? Unmanaged gateways would assume that the
other and knows how to handle the SIP REFER method.
Asterisk as a sip trunking system is not exactly compliant.
If REFER is not supported, then the me
Karoo Bridge also uses the FS libraries. What FS lacks is an admin GUI,
just like karoo bridge. Does MOH work in your tests too?
On Nov 4, 2012 4:22 PM, "Josh Patten" wrote:
--
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
sip: helpd...@voice.myitdepartment.net
He
ng trouble retrieving parked calls
> usingDigium G100 T1 Gateway
>
>
>
> The other part that I had noticed was this:
>
>
>
> Subscription-state: terminated;reason=noresource
>
>
>
> Richard Bruce
>
> Dimensional Communications
>
_
> sipx-users mailing listsipx-us...@list.sipfoundry.org
> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>
>
>
>
> ___
> sipx-users mailing list
> sipx-users@list.sipfoundry.org
>
Archive: http://list.sipfoundry.org/archive/sipx-users/
>
--
~~
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
Fax: 434.465.6833
~~~~~~
Linked-In Profile:
http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
Ask about our
I think in the last update (#22) this .might have been introduced.
SIP/2.0 481 Call leg/transaction does not exist^M ---
should not be there and I am seeing the same retrieving calls using
siptrunks.
I think this is a bug in sipx update #22.
On Oct 31, 2012 11:01 PM, "Richard Bruce"
wrote:
>
I have also
>> tired another phone that has bootrom 4.3.1 and firmware 3.2.4.
>>
>> I will fire up my 4.6 machine tonight and pull the updates and see if it
>> works on there maybe.
>>
>> Thanks.
>>
>> -Bryan Anderson
>>
>>
>>
>>
>
son
>
>
>
> On Tue, Oct 30, 2012 at 4:52 PM, Tony Graziano <
> tgrazi...@myitdepartment.net> wrote:
>
>> Always a good idea to reset a foreign phone if it has been used.
>> On Oct 30, 2012 6:19 PM, "Bryan Anderson" wrote:
>>
>>> bootro
;
>
> On Tue, Oct 30, 2012 at 2:36 PM, Tony Graziano <
> tgrazi...@myitdepartment.net> wrote:
>
>> You need to have firmware 3.2.7 (not 3.3 or above!) and a recent version
>> of bootrom (4.3.1 i think is the latest).
>>
>> What version are you using? Is t
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