On 02/15/2012 06:48 PM, Tony Graziano wrote:
> Restart call resolver.
>
I did a "/etc/init.d/sipxecs restart", but the problem was still there.
Restarting only CDR solved the problem.
> Plan to upgradeto4.4
ASAP. :-)
Thank you.
Massimo
--
Massimo Vignone
UniMORE -
3:18.665072Z":DEBUG:getActiveCalls 6
"2012-02-15T16:43:48.66Z":DEBUG:getActiveCalls 6
"2012-02-15T16:44:18.657144Z":DEBUG:getActiveCalls 6
"2012-02-15T16:44:48.653167Z":DEBUG:getActiveCalls 6
"2012-02-15T16:45:18.647663Z":DEBUG:getActiveCalls 6
ate on the display.
Do you have experienced some problems like this? Is it a firmware bug?
Thank for your help.
Massimo
--
Massimo Vignone
UniMORE - Servizi Informatici - Reti e Sistemi
Phone: +39.059.2058032 Fax: +39.059.2058034
sip:massimo.vign...@unimore.it IS
Quoting from 3.2.5 Release Notes: "Although it is not a requirement, it
is recommended that BootROM 4.2.3 be used in conjunction with SIP 3.2.5."
HTH,
Max
--
Massimo Vignone
UniMORE - Servizi Informatici - Reti e Sistemi
Phone: +39.059.2058032 Fax: +39.059.2058034
sip:ma
ns Hicom, and we have plans to
dismiss all the Siemens switches in the next months...
I think that this can be replicated with every pbx with an available
E1/T1 port.
HTH,
Massimo
--
Massimo Vignone
UniMORE - Servizi Informatici - Reti e Sistemi
Phone: +39.059.2058032 Fax: +39.059.2058034
Hi,
Actually I have 2 Siemens Hipath and 3 Siemens Hicom pbxs connected to
SipXecs.
On each pbx I've used an E1 port connected to a Patton Smartnode 4960,
with Q.SIG license.
Patton gateways are configured on SipXecs as unmanaged gateways.
HTH,
Massimo
--
Massimo Vignone
Un
On 12/03/2010 05:25 PM, Tony Graziano wrote:
> Anyone seeing issues with incoming calls on these?
>
I have some spa942 and some spa922, running 6.1.5a firmware, with no issues.
What's the problem?
--
Massimo Vignone
UniMORE - Servizi Informatici - Reti e Sistemi
Phone: +39.059.2
using any SBC, I have a public address on my proxy. To make it
work do I need to enable the SIP trunking role?
--
Massimo Vignone
UniMORE - Servizi Informatici - Reti e Sistemi
Phone: +39.059.2058032 Fax: +39.059.2058034
sip:massimo.vign...@unimore.it ISN:80
17:B62FAB90:SipRegistrar:"[160-ENUM]
SipRedirectorENUM::LookUp NAPTR regexp '^\\+390592058(.*)$' does not
match for ENUM translation of 'sip:*2390592058...@unimore.it;user=phone'
- no contact generated"
Any suggestuion?
Massimo
--
Massimo Vignone
UniMORE - Servi
uot;Invite partecipants" feature in the
conference web page are logged in the SIPXCDR database as transferred calls.
At the moment I can't find how to get those calls from the database.
Do you know how to get back those calls?
Massimo
--
Massimo Vignone
UniMORE - Servizi Informati
Hi,
I need to calculate costs generated by a conference call where most of
partecipants were invited by the conference owner.
Any suggestion?
Massimo
--
Massimo Vignone
UniMORE - Servizi Informatici - Reti e Sistemi
Phone: +39.059.2058032 Fax: +39.059.2058034
sip:massimo.vign
not tagging correct
DSCP value to some packets (Trying,Ringing and OK).
HTH,
Massimo
--
Massimo Vignone
UniMORE - Servizi Informatici - Reti e Sistemi
Phone: +39.059.2058032 Fax: +39.059.2058034
sip:massimo.vign...@unimore.it ISN:8032*881
___
sipx-u
I'm using Kirk WS6000 and I'm quite satisfied: it's simple to configure,
very good audi oquality and radio coverage.and
But as I wrote in a previous mail, it has an issue about call transfer.
Massimo
--
Massimo Vignone
UniMORE - Servizi Informatici - Reti e Sistemi
Phone:
On 10/23/2010 02:55 AM, Tony Graziano wrote:
> (That new fisheye door cam is awesome).
Just trying Mobotix T24: at the moment it seems that DNS SRV records are
not supported.
Massimo
--
Massimo Vignone
UniMORE - Servizi Informatici - Reti e Sistemi
Phone: +39.059.2058032
etween them...
The problem is that I can't reproduce the timeout.
I'm wondering if using NAPTR and "source" validation for INVITEs could
cause a slow answer from the phone. I will try to setup phones to use a
static cache.
Massimo
--
Massimo Vignone
UniMORE - Servizi In
On 10/05/2010 01:09 PM, Michael Picher wrote:
> Are the phones and sipXecs server on the same LAN segment / VLAN with no
> firewall / nat between them?
>
> Mike
No, they are in different subnets and VLANS, packets go through routers,
without any NAT or firewall between them.
nes (3.1.3revC). On these phones
BLF is not used.
--
Massimo Vignone
UniMORE - Servizi Informatici - Reti e Sistemi
Phone: +39.059.2058032 Fax: +39.059.2058034
sip:massimo.vign...@unimore.it ISN:8032*881
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On 10/01/2010 12:14 AM, Worley, Dale R (Dale) wrote:
>
> From: sipx-users-boun...@list.sipfoundry.org
> [sipx-users-boun...@list.sipfoundry.org] On Behalf Of Massimo VIGNONE
> [massimo.vign...@unimore.it]
>
> In a HA scenario, with t
On 09/24/2010 03:25 PM, Tony Graziano wrote:
> what version of polycom firmware?
>
3.1.3 RevC (3.1.3.0439)
--
Massimo Vignone
UniMORE - Servizi Informatici - Reti e Sistemi
Phone: +39.059.2058032 Fax: +39.059.2058034
sip:massimo.vign...@unimore.it ISN:80
On 09/23/2010 10:10 PM, Massimo VIGNONE wrote:
> Hi,
>
> In a HA scenario, with two servers running Sipxecs 4.2.1-018932, users
> registered on one server can't call users on the other servers: they get
> back a "408 Request timeout message".
>
> The problem ha
> On 9/23/10 4:10 PM, Massimo VIGNONE wrote:
> Sounds like DNS issues (if you have multiple a records with internal,
> then external ip's?)
We use only public IPs (we have a class B network addressing)
My DNS setup:
mydomain.net. IN NAPTR 2 0 "s" "SIP+D
Hi,
In a HA scenario, with two servers running Sipxecs 4.2.1-018932, users
registered on one server can't call users on the other servers: they get
back a "408 Request timeout message".
The problem happens randomly, so I can't reproduce it and tracing calls
with Sipviewer doesn't help.
So I have
will not use the absoluthe path, but
will follow the symbolic link /usr/share/www/doc/stdprompts/conf to copy
the files.
Should I open a JIRA issue?
Massimo
--
Massimo Vignone
UniMORE - Servizi Informatici - Reti e Sistemi
Phone: +39.059.2058032 Fax: +39.059.2058034
sip:massimo
I've solved the problem reverting to the package
stunnel-4.26-1.i386.rpm
and editing the file
/etc/sipxpbx/sipxcallresolver-agent-config
removing the line
fips=no
Thanks for your help.
Max
--
Massimo Vignone
UniMORE - Servizi Informatici - Reti e Sistemi
Phone: +39.059.2058032
I resend the mail, because I've replied to the wrong thread.
Sorry!!!
---
In the previous mail I was wrong...
I've tried using stunnel 4.26, 4.28 and 4.33, but the CDR HA Tunnel never
started.
So now I'm wondering about what to do...
Is the 4.2.1 release compatible with CentOS 5.2 or should I
In the previous mail I was wrong...
I've tried using stunnel 4.26, 4.28 and 4.33, but the CDR HA Tunnel never
started.
So now I'm wondering about what to do...
Is the 4.2.1 release compatible with CentOS 5.2 or should I have to
upgrade the OS to a more recent version? If so, which version?
I've
the error message
goes away.
Hope this is enough to get the system working.
Max
--
Massimo Vignone
UniMORE - Servizi Informatici - Reti e Sistemi
Phone: +39.059.2058032 Fax: +39.059.2058034
sip:massimo.vign...@unimore.it ISN:8032*881
___
sipx-use
s not properly created.
Massimo
--
Massimo Vignone
UniMORE - Servizi Informatici - Reti e Sistemi
Phone: +39.059.2058032 Fax: +39.059.2058034
sip:massimo.vign...@unimore.it ISN:8032*881
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sipx-users@list.sipfoundry.org
List Arc
16:07:18 LOG3[20031:3086472912]:
SSL_CTX_set_cipher_list: 1410D0B9: error:1410D0B9:SSL
routines:SSL_CTX_set_cipher_list:no cipher match
Any idea?
Massimo
--
Massimo Vignone
UniMORE - Servizi Informatici - Reti e Sistemi
Phone: +39.059.2058032 Fax: +39.059.2058034
sip:massimo.vign...@unim
Hi,
I want to upgrade my localization package in my production system.
I'm guessing if uploading it will reset the dial plan.
If so, is there a smart upgrading procedure that will not reset the dial
plan to defaults?
Max
--
Massimo Vignone
UniMORE - Servizi Informatici - Reti e Si
ng to multiple group.
Can you help me?
thanks,
Massimo
--
Massimo Vignone
UniMORE - Servizi Informatici - Reti e Sistemi
Phone: +39.059.2058032 Fax: +39.059.2058034
sip:massimo.vign...@unimore.it ISN:8032*881
___
sipx-users mai
s that they don't
accept the check-sync event.
I can restart them only using the sipsend command, using the "reboot"
event (or using the "reboot-cisco" event).
The Auth_Resync-Reboot_1_ parameter is set to no, phones use 5.2.5 and
6.1.5a firmware.
Any idea?
Massimo
Hi,
I'm moving from a 3.8 based large installation to a 4.0.4 sipxecs based pbx.
I have several patton smartnode gateways.
All of them worked correctly with the 3.8 release.
All of them are configured using the following example:
...
gateway sip GW-MYDOMAIN
bind interface LAN router
servi
ely feasible to wrap that up in a tidy VM of some sort. If
> youdo get it working though, please zip it up and share it somewhere!
>
>
> Tony
Thanks,
Max
--
Massimo Vignone
UniMORE - Servizi Informatici - Reti e Sistemi
Phone: +39.059.2058032 Fax: +39.059.2058034
sip:massimo.vign..
can do call setup in SIP and
h.323. I'd like to do a fax server based on hylafax, using t38modem, and
integrated in a sipx based installation.
It is not a problem about t.38, but I'd like to know more about how to
configure t38modem to interact with Sipx.
Thanks,
Max
--
Massimo
Hi everybody,
Anyone has got some experience using t38modem (and hylafax) with a sipx
HA installation?
Thanks,
Max
--
Massimo Vignone
UniMORE - Servizi Informatici - Reti e Sistemi
Phone: +39.059.2058032 Fax: +39.059.2058034
sip:massimo.vign...@unimore.it ISN:8032*881
their own HA sip solutions and run
> over our WAN?
>
How remote sites are connected to the headquarter?
> Thanks,
>
> Kenny Mitchell
> INEOS Refinery
> Grangemouth
>
>
>
>
> ___
> sipx-users mailing list
> sipx-users@l
Scott Lawrence wrote:
> On Fri, 2009-04-03 at 09:57 +0200, Massimo Vignone wrote:
>> Ok, this is the expected behaviour, so I configured an user on the
>> gateway and in sipx with the right permission for long distance calls.
>> But debugging on the gateway, it seems that t
cy pbx were challenged again for authentication.
That's why I wrote to the list: it seems that calls from gateway are
always challenged for authentication with the user in the from header,
and I'm guessing how I can change this behaviuor (if this was possible...)
Today I'll do othe
at most, even for hundreds of users,
> and you get a granular level of control and monitoring out of it. i
> think the ROI is ok there =)
>
Legacy pbxs can manage cdrs on their own so this is not a big problem.
The goal is to maintain gateway configuration as simple as possible.
Max
-
al plan are set with no
particular permission, and if there is a way to bypass/disable
authentication.
Thanks,
Max
--
Massimo Vignone
UniMORE - Servizi Informatici - Reti e Sistemi
Phone: +39.059.2058032 Fax: +39.059.2058034
sip:massimo.vign...@unimore.it ISN:8032*881
___
xml file containing new rules and modify the
/etc/sipxpbx/sipxconfig.properties.in file to inject the rules in Sipx.
Am I right? Where I can find some examples?
Max
--
Massimo Vignone
UniMORE - Servizi Informatici - Reti e Sistemi
Phone: +39.059.2058032 Fax: +39.059.2058034
sip:massimo.vign.
Hi everybody,
Take a look at this scenario.
Sipx with two gateways: gateway A is connected to another (traditional)
PBX and gateway B is connected to PSTN.
The traditinal PBX sends outgoing calls from extension 200 through the
gateway A, then Sipx routes calls according its dial plan.
In the di
the Outbound Proxy Server field.
Sorry!
Max
--
Massimo Vignone
UniMORE - Servizi Informatici - Reti e Sistemi
Phone: +39.059.2058032 Fax: +39.059.2058034
sip:massimo.vign...@unimore.it ISN:8032*881
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e *78+ext method on the same phone, directed call-pickup
works well. It seemed to me that the former method bypasses the proxies,
but I can't figure out why.
Max
--
Massimo Vignone
UniMORE - Servizi Informatici - Reti e Sistemi
Phone: +39.059.2058032 Fax: +39.059.2058034
sip:ma
me why?
Thanks,
Max
--
Massimo Vignone
UniMORE - Servizi Informatici - Reti e Sistemi
Phone: +39.059.2058032 Fax: +39.059.2058034
sip:massimo.vign...@unimore.it ISN:8032*881
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Lis
dleMessage
>>
>> transport error"
>
> That last is probably important... it looks like the proxy is unable to
> open the connection to the CDR database on that server. Check to see
> that postgres is running.
>
>
>
>
--
Massimo Vigno
in.netForkingProxyCseObserver-10:B77A0B90:SipProxy:"ForkingProxyCseObserver::handleMessage
transport error"
Any idea?
Massimo
Scott Lawrence wrote:
> On Fri, 2009-03-20 at 13:36 +0100, Massimo Vignone wrote:
>> Hi,
>>
>> In my HA installation, on the distribu
n UNINITIALIZED or
TERMINATED state (1)"
Can you tell what does it mean?
Thanks
Massimo
--
Massimo Vignone
UniMORE - Servizi Informatici - Reti e Sistemi
Phone: +39.059.2058032 Fax: +39.059.2058034
sip:massimo.vign...@unimore.it ISN:8032*881
_
Hi everybody,
I have some phones in a public area. Phones can call only internal
extensions.
I want to enable some users to unlock those phones and make them able to
place outgoing (pstn) calls.
How can i get that? Is it possible to lock/unlock a phone?
TIA,
Max
Actually I have six Patton Smartnode 4960 with Q.SIG license connected
to six Siemens Hicom/Hipath PBXs through E1 connections, plus a Patton
Smartnode 4960 connected to the PSTN via an ISDN PRI.
Everything works well.
Hope this helps,
Max
--
Massimo Vignone
UniMORE - Servizi Informatici
e same intercom interface connected
to a Linksys PAP2.
I can open the door answering the call and using two tones (pressing "#"
and "2").
HTH,
Max
--
Massimo Vignone
CeSIA - Università di Modena e Reggio Emilia
Phone: +39.
27;t figure out how to finish the sipx
installation.
Thanks,
Massimo
--
Massimo Vignone
CeSIA - Università di Modena e Reggio Emilia
Phone: +39.059.2058032 Fax: +39.059.2058034
sip:massimo.vign...@unimore.it ISN:8032*881
___
sipx-users mailing
Massimo Vignone wrote:
> Hi,
>
> I'm installing two Centos 5.2 boxes, with Sipxecs 3.10.3.
>
> I'm trying to configure them as master and distributed servers.
>
> Trying to enable SSL connections on the distributed server, I was not
> able to find th
Hi,
I'm installing two Centos 5.2 boxes, with Sipxecs 3.10.3.
I'm trying to configure them as master and distributed servers.
Trying to enable SSL connections on the distributed server, I was not
able to find the sipxhadistrib.sh script?
Any suggestion?
Max
--
Massimo Vig
.
I've checked the "directed call pickup" feature on SipXconfig, and the
config file is correctly generated.
I think there's some settings on the Polycom phone that I've not
configured properly.
Anyone can help me?
Thanks,
Max
--
Massimo Vignone
CeSIA - Uni
mber all the steps I took to resolve it.
Which firmware version do you have?
Point your browser to the following link, and see if there's a release
note related to a firmware version that addresses your problem.
http://wiki.siemens-enterprise.com/index.php/optiPoint_410/420_S_Release_Notes
By
mented
UDP packets are dropped.
I've submitted this issue to the manufacturer, I hope they could fix it.
Thanks a lot,
Massimo
--
Massimo Vignone
CeSIA - Università di Modena e Reggio Emilia
Phone: +39.059.2058032 Fax: +39.059.2058034
sip:[EMAIL PROTECTED] ISN:8032*881
___
Gateway: 155.185.3.247
CallId: D33B0B93CCC340659F7408F38CDB4BC00x9bb90303
Thanks,
Max
--
Massimo Vignone
CeSIA - Università di Modena e Reggio Emilia
Phone: +39.059.2058032 Fax: +39.059.2058034
sip:[EMAIL PROTECTED] ISN:8032*881
test-gsm-2-gw.xml.gz
Description: GNU Zip compressed data
Thanks for your reply.
Attached you will find a sipx-trace generated file.
Call-id is dd42d8f69c661ea5
Sipx (v3.10.2) is 155.185.3.99
Grandstream phone is 155.185.3.121
GSM gateway is 155.185.3.247
The Grandstream phone calls a cellular phone, but gsm gateway doesn't
respond to INVITE sent by Si
t;algorithm=MD5" string.
Attached to this mail you can find two scenario traces, both using
Grandstream phones.
Can you help me to understand where is the problem (and how to solve it?).
Thanks in advance,
Massimo
--
Massimo Vignone
CeSIA - Università di Modena e Reggio Emilia
Phone:
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