On Tue, May 5, 2009 at 11:35 PM, Goran Donev
wrote:
> I am using SipX v 4.0 and am having problems configuring ITSP’s for
> incoming lines to use as a conference server.
>
>
>
> I have tried Voxitas, and with them I am able to make calls but can’t call
> in.
>
If you are able to make calls but n
I am using SipX v 4.0 and am having problems configuring ITSP's for
incoming lines to use as a conference server.
I have tried Voxitas, and with them I am able to make calls but can't call
in.
While with Call Centric I am not able to get it configured, I have been back
and forth with sup
On Tue, May 5, 2009 at 9:20 PM, James Holmes wrote:
> Does anyone have a working vitelity configuration that they can show me
> (minus the username and password of course)? I have les.net working, but
> my sipXecs 4 installation will not register with vitelity. Initially I
> was using the template
As Scott said, in general, you shouldn't copy trace information into the
body of your e-mail, as it rarely survives transit. You should put the
information into a file and attach it. However, in this case, you were
your information didn't get mutilated.
On Tue, 2009-05-05 at 18:28 -0700, Mark Wo
192.168.254.102 = Asterisk
192.168.254.120 = SipXecs
Call was initiated from the asterisk box X201 dialing the SipXecs extension 2010
192.168.254.102 192.168.254.120
| |
1: |UINVITE--->|
2: |<--100 Trying/INVITE--U|
3
Does anyone have a working vitelity configuration that they can show me
(minus the username and password of course)? I have les.net working, but
my sipXecs 4 installation will not register with vitelity. Initially I
was using the template with my username and password added, but when
this didn't wo
I have an ACD Queue configured to receive all incoming calls. When a
call comes in and an agent receives the call they can transfer the
caller to an extension directly, but they are not able to transfer to
another Queue Line. The transfer fails and the caller comes back to the
original agent that
There are two pieces that haven't been fully discussed here, the firewalls at
either end. Just a suggestion that if you have a way in either firewall (the
one where sipx sits behind, or the one where the remote worker sits behind) to
prioritize traffic over other traffic riding on either network
Tim already answered that:
> The sipXecs box is a Pentium 4 3.20GHz with 512MB of RAM dedicated for
sipXecs. It is not running inside a virtual machine. When testing, there
were no other calls on the box.
- MM
On Tue, May 5, 2009 at 5:30 PM, Tony Graziano
wrote:
> A little late to the conversa
OK, I can wireshark it and do the same thing as sipviewer right?
-Original Message-
From: Scott Lawrence [mailto:scott.lawre...@nortel.com]
Sent: Tuesday, May 05, 2009 1:42 PM
To: Mark Wood
Cc: 'Tony Graziano'; 'sipx-users@list.sipfoundry.org'
Subject: Re: [sipx-users] Unmanged Gateways
On Tue, 2009-05-05 at 13:28 -0700, Mark Wood wrote:
> OK, went thru the dialplan and it is identical to the 3.10 version.
> Ran the tail command and got the log info:
Don't paste logs into email - line wrapping and other mail-client
nonsense makes it hard to analyze them.
To debug this, you'll ne
A little late to the conversation. Is your install on a VM?
-Original Message-
From: "Tim Byng"
To: M. Ranganathan
Cc:
To: Robert Joly
Sent: 5/5/2009 3:53:51 PM
Subject: Re: [sipx-users] Remote Worker Call Quality
OK, I understand a little better now about how the media relay works
OK, went thru the dialplan and it is identical to the 3.10 version. Ran the
tail command and got the log info:
"2009-05-05T18:24:00.355374Z":1141:OUTGOING:INFO:sipex.redphonetech.local:SipClientTcp-18:B6570B90:SipXProxy:"SipUserAgent::sendUdp
UDP SIP User Agent sent message:\nRemote Host:192
OK, I understand a little better now about how the media relay works. Thanks
for the clarification.
To rule out RAM and swapping as the issue, I upgraded the machine to 1GB.
The problem still exists. I double checked the swap usage when running the
tests and it is 0.
I have run a capture on both
Geoff Brozny wrote:
> Damian Krzeminski wrote:
>> You are *not* in the same boat. This is perfect output :-)
>> You absolutely should be able to access sipXconfig UI if you see that.
>> Try https://{your host name}:8443/sipxconfig
>>
>> (redirection from http://{your host name} would only work if y
Tony Graziano wrote:
> Also known issues, all in the tracker for 4.01.
Ok, thanks, I'll dig though there and see if I can figure it out then.
geoff
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Also known issues, all in the tracker for 4.01.
-Original Message-
From: Geoff Brozny
To:
Sent: 5/5/2009 3:14:53 PM
Subject: Re: [sipx-users] Unable to get 4.0 started after update through Yum
Damian Krzeminski wrote:
>
> You are *not* in the same boat. This is perfect output :-)
>
Damian Krzeminski wrote:
>
> You are *not* in the same boat. This is perfect output :-)
> You absolutely should be able to access sipXconfig UI if you see that.
> Try https://{your host name}:8443/sipxconfig
>
> (redirection from http://{your host name} would only work if you start
> entire sipxe
Geoff Brozny wrote:
> Boy Aidil Sjam wrote:
>> I've got an error said:
>>
>> [sipxcha...@voip root]$ sipxconfig.sh
>> could not change directory to "/root"
>> Failed to execute: alter table sbc_device add column port int4 not null
>>
>> BUILD FAILED
>> /etc/sipxpbx/database/database.xml:266: The f
known issue:
http://track.sipfoundry.org/browse/XX-5653
>>> Geoff Brozny 05/05/09 2:45 PM >>>
Boy Aidil Sjam wrote:
> I've got an error said:
>
> [sipxcha...@voip root]$ sipxconfig.sh
> could not change directory to "/root"
> Failed to execute: alter table sbc_device add column port int4 not n
Samy Touati wrote:
> Hi,
>
> I would like to remove some options in the user web management interface.
> For example I would like to remove the phonebook tab option, the
> function would still be there, it's just not available for end-users.
> Can this be done by editing config files?
>
> Thanks.
Boy Aidil Sjam wrote:
> I've got an error said:
>
> [sipxcha...@voip root]$ sipxconfig.sh
> could not change directory to "/root"
> Failed to execute: alter table sbc_device add column port int4 not null
>
> BUILD FAILED
> /etc/sipxpbx/database/database.xml:266: The following error occurred whil
Hi,
I would like to remove some options in the user web management interface.
For example I would like to remove the phonebook tab option, the function
would still be there, it's just not available for end-users.
Can this be done by editing config files?
Thanks.
Samy.
This can be investigated only with a sipx-snapshot generated snapshot.
Its tough to tell whats going on without one. Please generate one
with the failing scenario and mail me with it. Remember to set the
logging level to sipxbridge to DEBUG and that of the Proxy server and
Registrar to INFO. I can
When I do a debug on the gateway, all calls look exactly the same, the
one that works to the AA and the one that doesn't to the extension.
Also, the gateway config has no change and it was working fine with
SipXecs 3.10.3.
Tony Graziano wrote:
If the call comes in through a gateway and gets t
I would suggest you go through your dialplan entries and compare them to what
you had before. You might also think of putting a "catchall" rule in for
troubleshooting and turn the proxy log up to debug and see what is being sent
to sipx in the way of digits.
I saw a "weird" thing too trying to
If the call comes in through a gateway and gets to the AA but disconnects on
transfer, it's likely the gateway is not handling the REFER for the transfer
properly.
>>> Jhony Perez 05/05/09 10:47 AM >>>
Thank you for your quick reply, based on your reply I got part of it
working but it broke o
Mark,
This sounds like the same issue I'm having with the Cisco gateways, I
don't have a fix for it yet but you can see the "SipXecs 4 with Cisco
gateway issues (Jhony Perez)" in the list.
When you call in from the Asterisk try sending the call to the SipXecs
Auto Attendant, if this works th
On Tue, May 5, 2009 at 12:29 PM, Tim Byng wrote:
> I think I'm missing something here. I thought that the media (RTP) was
> supposed to take the most direct route possible? Doesn't that mean that the
> media should bypass sipXecs and go directly to the ITSP?
Tim,
That is not the case for calls v
> I think I'm missing something here. I thought that the media
> (RTP) was supposed to take the most direct route possible?
> Doesn't that mean that the media should bypass sipXecs and go
> directly to the ITSP?
Part of the work done to overcome far-end NATs include anchoring the
media to a med
Borginger Rikard wrote:
> Great! Thanks alot for. Sorry for the hijack(?).
>
> Is there any work in progress for UI translation or where shall I apply? :)
>
> / Rick
>
>
No work in progress as far as I know.
This should get you started:
http://sipx-wiki.calivia.com/index.php/SipX_ConfigServe
> > It would help tremendously helpful if I could get a tcpdump of a
> > remote worker call taken from the sipXecs box itself while
> sniffing the 'any'
> > interface. The tcpdump command I usually use is 'tcpdump
> -n -nn -s 0
> > -i any -w capture.pcap. If you are willing to share that with
Fresh ISO install of 4.0
Setup an Asterisk box as an unmanaged gateway for interim purposes, also setup
associated dial plan. I can make calls from the SipXecs system to asterisk and
use the Asterisk box as a PSTN gateway as well. But calls from the Asterisk to
the SipXecs do not go through, it
On Tue, 2009-05-05 at 12:09 -0400, Scott Lawrence wrote:
> That's not much memory. Is the box swapping? (if you run 'top', there
> is a Swap: line in the header - the 'used' number should be very small:
> a few megabytes at most, and should not rise at all once the system has
> been running for a
Keith Gearty wrote:
> I'm experiencing javascript errors in the web ui which is preventing the
> "Show Advanced Settings" link from working.
>
> Firebug reports the following errors when the page loads:
>
> * Could not load 'nls.dojo_en-gb'; last tried '__package__.js
> dojo.js (line
Keith Gearty wrote:
> In 3.10 the hunt groups had a field called "Fallback Destination". That
> field seems to be missing in 4.0. Is this functionality now implemented
> in a different way? What do I do to replicate the fallback destination
> functionality in 4.0?
>
> Thanks,
> Keith.
If y
> It would help tremendously helpful if I could get a tcpdump of a remote
> worker call taken from the sipXecs box itself while sniffing the 'any'
> interface. The tcpdump command I usually use is 'tcpdump -n -nn -s 0 -i
> any -w capture.pcap. If you are willing to share that with me, that
> woul
I think I'm missing something here. I thought that the media (RTP) was
supposed to take the most direct route possible? Doesn't that mean that the
media should bypass sipXecs and go directly to the ITSP?
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> > Could you check your sipxbridge settings ( sans passwords ). You
> > should not need any media keepalive ( that can negatively affect
> > performance ) for les.net.
>
> I checked the settings again and everything seems fine. I
> played around with the sipxbridge Media keep-alive interval.
> That's not much memory. Is the box swapping? (if you run 'top', there
> is a Swap: line in the header - the 'used' number should be very small:
> a few megabytes at most, and should not rise at all once the system has
> been running for a while).
Hmm, didn't realize that I was at the bottom of
On Mon, 2009-05-04 at 12:34 -0400, Tim Byng wrote:
>
> The sipXecs box is a Pentium 4 3.20GHz with 512MB of RAM dedicated for
> sipXecs. It is not running inside a virtual machine. When testing,
> there were no other calls on the box.
That's not much memory. Is the box swapping? (if you run 'top
There has not been and work on XECS-108 and I would really like to
know if there are still plans to support FreeBSD for 4.x and moving
forward?
/carmi
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> Could you check your sipxbridge settings ( sans passwords ). You
> should not need any media keepalive ( that can negatively affect
> performance ) for les.net.
I checked the settings again and everything seems fine. I played around with
the sipxbridge Media keep-alive interval. I was using th
In 3.10 the hunt groups had a field called "Fallback Destination". That
field seems to be missing in 4.0. Is this functionality now implemented
in a different way? What do I do to replicate the fallback destination
functionality in 4.0?
Thanks,
Keith.
Dale,
I have looked at the trace xml (merged log files) in sipViewer, but truth be
told, that's a bit over my pay grade. Also, I wasn't sure if it was
acceptable to post the trace xml here.
1) Is it ok to post the trace xml?
2) If so, should I paste it into the content of the email directly or p
I'm experiencing javascript errors in the web ui which is preventing the
"Show Advanced Settings" link from working.
Firebug reports the following errors when the page loads:
* Could not load 'nls.dojo_en-gb'; last tried '__package__.js
dojo.js (line 96)
* Could not load 'tapestr
On Mon, 2009-05-04 at 17:56 -0500, Clint Anderson wrote:
> Any suggestions would be greatly appreciated and thank you all very much for
> your help and your time.
Have you gotten a trace of a failed call? Without that, you can't tell
which element of the system has malfunctioned.
Dale
On Tue, 2009-05-05 at 16:33 +0200, Gmb wrote:
> Hi,
> I've installed sipx version 4.0.0-015321 in HA configuration with a
> primary server and
> a secondary server with Redundant Sip Router and Conferece server.
> I have some problem with secondary server, I've i tried to force
> registration on
On Mon, 2009-05-04 at 23:16 -0700, Todd Hodgen wrote:
> I've have the same issue that I have been working to figure out. Calls from
> a Polycom phone to two different xlite phones work fine. Calls from xlite
> to xlite, or xlite to Polycom get a re-order tone. Calls from xlite to ITSP
> get reor
Thank you for your quick reply, based on your reply I got part of it
working but it broke other areas.
What I meant for internal extensions is extension that are services
running on the server itself, IE. AutoAttendant, VoiceMail, Park, Etc.
Yes, I did leave the "Inbound Calls Destination" bl
Hi,
I've installed sipx version 4.0.0-015321 in HA configuration with a
primary server and
a secondary server with Redundant Sip Router and Conferece server.
I have some problem with secondary server, I've i tried to force
registration on secondary
server, i see phone successfully registered, but
Paul Mossman wrote:
>
> Tony wrote:
>> I am able to use the system without issue except I cannot
>> upload Polycom soundpoint files. I created a 'deactivated'
>> placeholder for my files, then tried to upload the bootrom
>> (smallest file) through sipxconfig. i was unable to do so
>> (not get
Hi.
Is there any way to allow a conference to stay up indefinitely with only a
single user? It seems to drop the single user after 10 minutes.
I couldn't see a setting that controlled that in the server admin page.
Also, is there a way to stop the "on hold" music for the conference when
there i
On Tue, May 5, 2009 at 4:02 AM, James Holmes wrote:
>> I'm new to sipXecs and I'm having problems getting it to register with
>> either of my ITSPs (Vitelity and Les.net). I'm not sure if the problem
>> is firewall/NAT related or if I simply have something configured wrong.
>>
>> When I configurin
On Mon, 2009-05-04 at 18:15 -0400, Chris Tresco wrote:
> I have the following scenarios:
>
> Office Communicator -> OCS -> OCS Mediation Server -> sipXecs -> ITSP
> Office Communicator -> OCS -> OCS Mediation Server -> sipXecs -> Polycom
> Phone (x)
>
> I can dial from my polycom phone throu
Tony wrote:
> I am able to use the system without issue except I cannot
> upload Polycom soundpoint files. I created a 'deactivated'
> placeholder for my files, then tried to upload the bootrom
> (smallest file) through sipxconfig. i was unable to do so
> (not getting any progress bar in fire
This is known issue
Please go through;
http://track.sipfoundry.org/browse/XX-5121
Thanks,
Akshata
Tim Byng wrote:
> Yep, I noticed it too.
>
>
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I am able to use the system without issue except I cannot upload Polycom
soundpoint files. I created a 'deactivated' placeholder for my files, then
tried to upload the bootrom (smallest file) through sipxconfig. i was unable to
do so (not getting any progress bar in firefox), I switched to IE an
Yep, I noticed it too.
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Has anyone noticed this?
The SipX in question is the standard ("pub") SipX 4.0.0. The test
scenario is very simple: I call an extension that has no endpoint
registered from a desk phone. The voicemail triggers and then I leave a
message and immediately after the message I hang up (so there is no
s
When a sipx user calls a pstn line (via internal sbc) and then transfers this call to another pstn line (via internel sbc) does this call still exists in sipx after the transfer ?Is there a way to get the cdr (or just duration) of this 'pstn to pstn' call ?MdM
__
> I'm new to sipXecs and I'm having problems getting it to register with
> either of my ITSPs (Vitelity and Les.net). I'm not sure if the problem
> is firewall/NAT related or if I simply have something configured wrong.
>
> When I configuring the Devices->Gateway->Vitelity->SIP Trunk->ITSP
> Accou
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