Hello,
I'm trying to find a way to import (global) address books via command line
scripting or using an API. As far as I can see the current API
<http://wiki.sipfoundry.org/display/sipXecs/Configuration+Web+Service> only
allows to edit user phone books, but not global phonebooks.
list. The answer was something like: "the
wiki and download pages are community driven, so that's why there are not
up to date".
As far as I understand sipXecs 4.6 is released. Just the information is not
up to date.
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Claas Hil
ybe the old content should be removed or at least a link from wiki page
to issue tracker release notes pages should be added.
So the question is: Is there a new up-to-date site with contains all
information or is there just no spare time to update wiki/download pages?
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Cl
we completely disabled BLF and this is
somewhat worse if the BLF function is a top argument to use PolyCom phones
with two sidecars ;)
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Claas Hilbrecht
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ages a further enhancement or is this
impossible at all?
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n (TransactionPlugin) that allows
> modification of both request and responses. The primary goal of this
> plugin is to provide a compatibility layer between sipX and
> misbehaving phones by providing a means for a plugin to
> modify/remove/add sip headers to ensure proper interop.
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l valid? I'm thinking about to reinstall sipXecs from scratch and if
4.6 is just a few days away ...
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a real local reference clock. Since
there are so many different ones I think it could be sufficient to allow
some text from a textbox to include in a generated ntp.conf. This makes it
possible to use most of the magic from sipXecs but allows to use a local
reference clock as well.
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283)
at
java.util.concurrent.ThreadPoolExecutor$Worker.runTask(ThreadPoolExecutor.java:886)
at
java.util.concurrent.ThreadPoolExecutor$Worker.run(ThreadPoolExecutor.java:908)
at
org.apache.mina.util.NamePreservingRunnable.run(NamePreservingRunnable.java:51)
at java.lang.
anyone using sipXecs + IM with success? See my previous msg about the IM
state with "On the phone". I think something is seriously wrong with my
setup or something is broken inside sipxecs.
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Claas Hilbrecht
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__
(Linum)" should be display.
<http://www.jucs-kramkiste.de/fileadmin/user_upload/claas/sipxecs-im-state.png>
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iew, both show the same (different to On the phone)
message. But pidgin always show "On the phone". I'm running 4.4.0-
2012-04-13EDT09:33:36 ip-10-72-10-163.
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one".
What system needs to be restarted to get the setting change active?
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this too :).
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> You add it the the poly com firmware bundle. There is a section. For
> licenses. Its not an efk. Undo that.
Under phone settings there's a license tab but I can only see one option
there: polling.time
Where is the configuration option to add these license files?
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Claas Hilb
phone. How can I
add the free polycom site license to sipXecs?
http://www.polycom.com/products/voice/applications/index.html
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ompletely. You will even notice this with
iftop on our VPN link, there is no extensive usage anymore.
You say that you setup a remote sipXecs site to solve this issue. What have
you done exactly to make monitoring (a lot) speed dials work realiable?
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Claas Hilbrecht
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to calculate the BLF
traffic.
>
>
> mike
>
>
> On Tue, Apr 24, 2012 at 6:29 AM, Claas Hilbrecht
> wrote:
>
>
>> Have you tried capturing the traffic between sites and measuring it
>> rather than just assuming what's going on?
>
> Only with iftop
> Have you tried capturing the traffic between sites and measuring it
> rather than just assuming what's going on?
Only with iftop so far and that shows the VPN link is saturated to 100%. I
can use tcpdump next to create a capture and load this into wireshark.
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ly increase bandwidth consumption?
Is there any way to calculate the required bandwith?
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> Change the order of the lines on the phone and see if the problem flips
> lines.
Good idea, I will try this.
And just to make sure. Is sending a busy on DND something that is supported
by sipxecs? Reading the help text on the DND setting I would think so.
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Claas Hilbrecht
es for
> differences.
But what config files could I compare besides the one that sipxecs
generates that I'm able to download in the "phone view"?
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sipx-u
fault, using every entry under "reset to
defaults", even formatted the filesystem. But the problem remains the same.
On the same (!) polycom phone the extension "222" sends a busy on DND or if
the line is in use while the extension "219" doesn't.
There must
sy signal if I call the line 222 (from outside) and a ring tone
on line 219. The voicemail is disabled for both lines.
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sername, password and so on). I
even downloaded the phone configuration from sipxecs and compared both
lines with diff. Even there I can see any difference besides
username+password. So there must be some other settings somewhere inside
sipXecs, but I don't know where to search now.
Has someo
> yes, it depends on what you want to use it for though. we use a cti
> app on the pc that integrates with the polycom directly and laso
> support several 3rd party crm apps.
Which app you're using?
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Hello,
just for the records, upgrading/downgrading firmware to
> First off, the recommended polycom firmware at present is 3.2.4b and
> bootrom 4.3.1.
did the trick. The headset has no delay while connecting to an incoming
call.
Thanks @Michael and as everytime @Tony
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described it. The only thing that is different
is our PolyCom firmware. I'm just using the version Michael recommended. I
will come back and report.
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tip.
sipxecs config: analogHeadsetOption is on/on
The headset is set to Jabra mode on the phone, echo cancelling is on.
PolyCom firmware:
spip_ssip_vvx_3_2_5_release_sig_split.zip
spip_ssip_BootROM_4_3_0_release_sig.zip
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Claas Hilbrecht
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anything
else that I need to know?
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. Your curl should be able to dial on the command line
successfully before you try to get Telify working.
5. Click on any telified url and enjoy :)
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vice instructs
you to do so.
I thought "cluster" is aka "HA system" in the sipxecs world ...
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e
/etc/yum.repos.d/sipxecs.repo has to be removed or not.
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e even ssh could be used.
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x27;s not like you're going to make configuration changes if the
> master is off-line.
A read only "mirror" is sufficent I think. Even the rsync approch would
work, be maybe it's better to use some "built-in" replication for this
700) include the 0.
Can someone confirm my settings or explain how to do it?
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on.
But meanwhile I found another very simple solution that will solve the
problem meanwhile. Just use branches to set the preferred gateway. Since
each user has up to four different sip accounts registred on a polycom
phone I can setup the user B to belong to branch B that uses gateway B.
-
7;ve read the patton documentation "call
router configuration" but putting all things together is a different thing.
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right, I already noticed that but I just want to make sure that I
understand everything correctly.
Has anyone done this before with a patton gateway? I'm somewhat stuck now
how I can route a call from one patton gateway to another.
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Claas Hilbrecht
efault to 30
mapping-table calling-e164 to called-e164 reject-call
map 2..$ to 1
complex-function test
execute 1 reject-call
execute 2 map-caller-id
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on available somewhere how to setup such a system?
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ackup" tftp server on the secondary systems would be useful.
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> authentication name should be "202"not "202@xxx"
Hmm, what should I say other than: "I'm so stupid" ...
Thanks a lot Tony.
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ot; request.
You can see all posible settings e.g. at
<http://en.wiki.edpnet.be/index.php/Siemens_Gigaset_C470IP>. I tried every
combination but without success.
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Gigaset-C470IP.pcap
Description: Binary data
__
it before the agent takes the call.
---
This sound exactly like the information I wish to display. With the plugin
in XX-5004 maybe I can get this working even as a newbie.
Now I'm looking for more information about writing plugins, maybe there's a
VM somewhere with a current d
umber 5678 -> company B
From: 123456-7890
To: company A
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german translation should be more meaningful?
And again, big thanks.
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three other common private networks as
defined in RFC1918 are used more or less and I would like to avoid using
them.
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t; trace...
I will report after the sipXecs 4.2.1 reinstall in a few minutes.
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nother location
with the same problems. And at this site I never ever use the Web GUI,
upgrade the firmware or have done anything else what I can imagine that
could create such problems. As soon as I get the problems solved the plan
is to connect both sites.
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Claas Hilbrecht
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