Hi,
first I would like to know if it is even possible (now as of SipX 4.2)
to combine a DimDim WebConference with a conference held via local SipX
Server. I'm planning on also bringing together an audio and a multimedia
conference with just audio available to users of IP phones.
So please
Dim Dim integration is already in the 4.2 release. It's not a real tight
integration, more of a link between the two system. Enter your dim dim
account information in your conference bridge is basically all you do. It
creates links you can send to your conference invitee's, you click on your
Il lunedì 09 agosto 2010 21:03:38 Tony Graziano ha scritto:
You need to tell me what is in the field for the MOH uri in your phone
config file.
Copy and paste:
sip:~~...@skoer.it
--
Claudio Succa
PERTEL - Torino - Italy
+39-011-19826800
http://www.pertel.it
http://www.uniassist.it
Hello List members.
I did the following steps in my update: updated my old SipX 4.0.4 to 4.2.1
and archived the Config Vm via webinterface.
Then i did a clean install of Sipx 4.2.1-018930 and restored Config VM.
Unfortunately the Voicemail system does not work any more. I can login but
when I
Gary, usually that problem is some sort of refer problem or the provider is
sending calls in on 5060 instead of 5080.
Mike
On Mon, Aug 9, 2010 at 11:38 PM, Gary Luca garyluc...@gmail.com wrote:
Tony,
I'm using a SIP trunk through Broadvox with the default settings as shown
below in your
That usually works fine. Were you able to get in, or was the superadmin
password set to blank?
On Tue, Aug 10, 2010 at 1:57 AM, Jun.Wen jun.wen.s...@gmail.com wrote:
Hi, I tried to reset superadmin password as the guide of “sipxconfig.sh
--database reset-superadmin”. Whereas, after
Clean install from ISO?
On Tue, Aug 10, 2010 at 3:51 AM, Rene Pankratz
rene.pankratz.l...@iant.dewrote:
Hello List members.
I did the following steps in my update: updated my old SipX 4.0.4 to 4.2.1
and archived the Config Vm via webinterface.
Then i did a clean install of Sipx
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If in doubt take a
Actually, I remember this I think...
It had to do with extra xml settings in config that had not populated in VM
settings, because there are more parameters than there were before. I
believe a workaround I had at the time was to disable Vm for a user, then
re-enable it. Can you try that for one
Yes, a clean install from ISO. Then restored config/voicemail in
Webinterface.
After cleaning up all /inbox directories in mailstore it seems to me the VM
is acting correct right now.
But I will do further testing to clarify that there is no further problem
with VM.
René
2010/8/10 Michael
I disabled and re-enabled voicemail for a few users. No failed job appeared
in Webinterface.
So I repeated the steps for all users.
The VM Service started up without any errorsafter the update. The only error
I could find is the one I posted in my first Post in this thread.
Everything works fine
FWIW - roadmap for 5.0...
Call Announce (aka Call Screening)
Phase 2 will also include the classic Call Announce functionality. If the PA
is front ending all calls then (based on user config), if the user chooses
the option to transfer to the user, the caller is prompted to record his/her
name.
Il martedì 10 agosto 2010 11:35:27 Claudio Succa ha scritto:
Following your suggestions I changed the moh in the snom phone to
sip:~~...@sip.skoer.it and now it works.
There is another problem about snom configuration.
The default codecs order is:
Codec 1 G.711u
Codec 2 G.711a
Codec 3 G.722
The media server supports g722 as well as g771u/a. What would be interesting
to know is why in your case you need to change the order of codecs. During a
media negotiation, at least several codecs should be offered before an
agreement is made.
Internally it should never be an issue. On gateway
Il martedì 10 agosto 2010 12:33:21 Tony Graziano ha scritto:
The media server supports g722 as well as g771u/a. What would be
interesting to know is why in your case you need to change the order of
codecs. During a media negotiation, at least several codecs should be
offered before an
I think this is more like the way snom DOES NOT truly resolve DNS SRV
records properly.
example: internally you use a private DNS server (or acl that uses views),
for this you add an A record to DNS that uses sipx or another internal DNS
server so the phone can resolve sipdomain.com as the same
Il martedì 10 agosto 2010 16:36:06 Tony Graziano ha scritto:
I think this is more like the way snom DOES NOT truly resolve DNS SRV
records properly.
example: internally you use a private DNS server (or acl that uses views),
for this you add an A record to DNS that uses sipx or another
Il martedì 10 agosto 2010 18:32:22 Tony Graziano ha scritto:
How are you connecting to the pstn? Siptrunk or gateway?
Siptrunk
If trunk is the itsp sending to you on port 5060 or 5080?
5080
--
Claudio Succa
PERTEL - Torino - Italy
+39-011-19826800
http://www.pertel.it
your sbc routes page looks like this?
SIPPublic portSet this if your ITSP requires a specific port.
External port(Default: 5080)The ITSP facing port. The ITSP will send
signaling here.
Signaling keep-alive interval(Default: 20)Keepalive timer for SIP Signaling.
sipXbridge will re-register with
I there any interest in a sipx-users channel on IRC. Maybe I'm old
school, but IRC has always been a great way to collaborate and get
people involved with sipXecs.
Just a thought.
-Jim
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List
Il martedì 10 agosto 2010 19:02:03 Tony Graziano ha scritto:
your sbc routes page looks like this?
There are two differences: Permitted Codecs is void and Incoming calls
destination has an alias where to send inbound calls.
Filling the Permitted Codes with its default does not change the
On 10-08-10 10:36 AM, Tony Graziano wrote:
*ATTN: Snom guru's, wasn't there some configuration nuance with Snom's
that related to the port number
(i.e. port=0 meant use SRV records, while manually encoding port 5060
meant ignore srv records?
Or something like that?*
You are thinking of
thanks for the memory jog!
On Tue, Aug 10, 2010 at 1:26 PM, Norman Branitsky
nor...@cherniaksoftware.com wrote:
On 10-08-10 10:36 AM, Tony Graziano wrote:
*ATTN: Snom guru's, wasn't there some configuration nuance with Snom's
that related to the port number
(i.e. port=0 meant use SRV
there is plenty of room on #sipx, i often wish there was only one mailing list.
On Tue, Aug 10, 2010 at 1:24 PM, Jim Canfield jcanfi...@emstar.com wrote:
I there any interest in a sipx-users channel on IRC. Maybe I'm old
school, but IRC has always been a great way to collaborate and get
Perhaps you want to put the codecs back in.
PCMU,PCMA,G722,L16
If all calls are going to a predefined destination, that's fine.
I would limit the codecs allowed at sipxbridge, for simplicity sake. I think
it might be problematic that either the ITSP or snom (or both) are
negotiating codecs that
(I'd like to hear Josh's response to this)
A log snippet from the gateway would be good too. It sounds like disconnect
supervision, but does the call show active while you think you are recording
it in the CDR table? If you think you are recording it but you get DEAD
SILENCE is one thing. Can you
Any help apprecaited!
I amusing this:
http://sipx-wiki.calivia.com/index.php/HowTo_configure_Xten_SIP_softphones_with_sipX
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Ujjval Karihaloo
Sent: Monday, August 09, 2010 8:19 PM
To:
Try to start using the new wiki
http://wiki.sipfoundry.org
http://wiki.sipfoundry.org/display/xecsuser/Configuring+Counterpath+X-Lite+Softphone
What you will want to do is use the SIPDOMAIN you used when setting up sipx
using the install script.
ex: if your sip domain is simplesignal.com and
according to snom they only support
G.711, G.729A,
G.723.1, G.722, G.726,
GSM 6.10 (full rate)
on that device. your itsp is making offers not even in that list. I'm
guessing that is an issue but I still need a useable siptrace to see
enough.
PCMU,PCMA,G722 should be the list in your sipxbridge
Got it..have it working..
Appreciate your expertise!
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Tuesday, August 10, 2010 12:36 PM
To: Ujjval Karihaloo
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Registering Endpoints
Try to start using the new wiki
Just noticed that we are unable to make any outbound calls to toll free numbers
after this latest update. We were able to on the previous version. We are
using Voxitas as the ITSP. Has anyone else seen this or tested TFN dialing
after the update? Normal local and long distance phone calls
On 8/10/10 3:14 PM, Tran, Ly V. wrote:
Just noticed that we are unable to make any outbound calls to toll
free numbers after this latest update. We were able to on the
previous version. We are using Voxitas as the ITSP. Has anyone else
seen this or tested TFN dialing after the update?
Well, it would help to know what your gateway is doing and how your dialing
rules work. I don't use voxitas so I can't help with anything that might be
peculiar there. Is your gateway set to add any digits to all call through
the gateway? Is you dialplan doing this either? Do you have a separate
I am using the same gateway with Voxitas for local, LD and tollfree. Because
they are using +1, I've set up the dial plan as you suggested to enable dialing
from the missed calls on the phone since the incoming caller ID shows +1 as
well. I have a basic dial plan of 10 digits, 1 appended then
Well, the rule to take +1 at the gateway should be removed and +1 should be
added to the individual dialplans or if your gateway does not require
registration you can create anotgher instance of it and not add +1 and send
10 digit toll free calls to that gateway.
Or you can get a voip.ms account
Whats the most common way to route a call based on incoming 10 digit DID to a
particular Auto Attendant (AA)
Cannot find the info on the latest wiki on setting up the AA.
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List Archive:
It an alias for the AA in the dial plan.
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:
You got it ...
Thx man
-Original Message-
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Tuesday, August 10, 2010 3:41 PM
To: Ujjval Karihaloo; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Auto Attendant setup
It an alias for the AA in the dial plan.
All the SIPX default announcements, AA, Conferencing etc seem to have choppy
Audio. Is it the Media path or the announcements themselves that need to be
professionally re-recorded if we were to use sipX in a Production environment.
What do most users out there do?
How about making sure you have:
a: no virtualization madness
b: proper hardware (proper clocking, proper bus and enough RAM)
Am 11.08.2010 um 00:18 schrieb Ujjval Karihaloo:
All the SIPX default announcements, AA, Conferencing etc seem to have choppy
Audio. Is it the Media path or the
Is your server virtual?
-Original Message-
From: Ujjval Karihaloo ujj...@simplesignal.com
Sender: sipx-users-boun...@list.sipfoundry.org
Date: Tue, 10 Aug 2010 15:18:52
To: sipx-users@list.sipfoundry.orgsipx-users@list.sipfoundry.org
Subject: [sipx-users] sipX Recordings
That usually indicates a resource issue. If you are using a softphone it
could be on the pc or the audio device on the pc.
I never install sipx with less than 4gb of ram. I rarely do a virtual server
and when I do I make sure kernel timing, ram and storage access are
configured so they give the
Can I send P-Assert-ID over the SIP trunk so that which ever extension is
calling their username gets send over the user portion of u...@domain in the
PAID.
So that would mean wild carding the user portion of the PAID
E.g.
I have extention with username - 1234
In the SIP Trunk ITSP setting
I have 4Gig RAM, No Virtualization...
I am using a softphone..will try a Polycom
Any guidance on setting up clocking ?
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Ujjval Karihaloo
Sent: Tuesday, August 10, 2010 5:08 PM
To: Michal
Kernel timesync is really only an issue if running on a virtual platform.
A good usb headset is important.
Run
top -U sipxchange
Are you using any swap? When you are listening to audio does it show a
high cpu usage?
On 8/10/10, Ujjval Karihaloo ujj...@simplesignal.com wrote:
I have 4Gig RAM,
On Tue, Aug 10, 2010 at 7:08 PM, Ujjval Karihaloo
ujj...@simplesignal.com wrote:
So if 1234 make a outbound call over the SIP trunk, sipX sends PAID in the
INVIT as following 1...@mydomain.com
latest 4.2.1 stable release, you'll see ability to control PAD on
ITSP settings. Even works if user
I'm guessing this is to capture detail for a billing platform running
on a switch?
On 8/10/10, Ujjval Karihaloo ujj...@simplesignal.com wrote:
Can I send P-Assert-ID over the SIP trunk so that which ever extension is
calling their username gets send over the user portion of u...@domain in the
Yes
Looks like sipX sends the itspusern...@itsp Address in the FROM header...
And PAID has the individual users User Id as follows use...@a.b.c.d (or
1...@1.2.3.4 in my example below) where a.b.c.d is the IP of the sipX server
itself.
I want PAID to look like use...@itsp-provider-address(Like
not pretty, but works.
installed postfix and switch-mail so I could redirect the email to a
local (on sipx) account without changing the headers
used procmailrc (already installed) to collect the headers and then call
the script.
installed lame to convert wav to mp3 (rpm), some might want to
On Tue, Aug 10, 2010 at 7:08 PM, Ujjval Karihaloo
ujj...@simplesignal.com wrote:
So if 1234 make a outbound call over the SIP trunk, sipX sends PAID in
the
INVIT as following 1...@mydomain.com
latest 4.2.1 stable release, you'll see ability to control PAD on
ITSP settings. Even works if user
You are amazing JJava programmers out there have mercy with this soul.
--martin
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Michael
Scheidell
Sent: Tuesday, August 10, 2010 9:19 PM
To: sipx-users@list.sipfoundry.org users
No, Did not see any spikes in CPU...I was not using too much swap either...
I believe it cud be my softphoneEyebeam running G722...cud be the
culprit..I will try G711u
-Original Message-
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Tuesday, August 10, 2010 5:15
Actually for a single call going into an AA, this seems high to me..
Tasks: 167 total, 1 running, 165 sleeping, 0 stopped, 1 zombie
Cpu(s): 14.3%us, 1.3%sy, 0.0%ni, 83.7%id, 0.3%wa, 0.0%hi, 0.3%si, 0.0%st
Mem: 1035080k total, 1008672k used,26408k free,79428k buffers
Swap:
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