Vodia support
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Hi, what type of tftp serve?r Pumpking, tft32?
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Hey Matt check http into the m3 server for some clues. there should be some logs there.
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does it work using HTTP Provisioning?
Plug N Play?
no it does not support pnp, I configure the phones in my lab, they are pretty easy to use, all done on the web portal server.
when you register a kirk it will send out a signal with it registration number, then this number will be display on the web portal
you will configure the extension there.
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Sure, using pbxnsip transfer not the transfer offered by the base.
Consider that we have 4 bases and 4 handsets.
I haven't tried to use more than one call, so I cannot give you any hint on this.. sorry.
I don't know if this could be a good solution for you, anyway I solved my problems with C470IP.
I'm evaluating the snom m3, and I opened a topic asking info about it.
Bye
--
Nicola
Hi Guys we have tested the polycom Kirk phone with the pbxnisip, very reliable and easy to set up
here is a link to there product.
http://www.polycom.com/products/voice/wire...20_handset.html
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I'm having this issue all of the sudden also.
open a ticket at the support site. so I can take a look at it
Hamlet Collado
pbxnsip support
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maybe we can get this working soon!
check this out copy from polycom:
http://knowledgebase.polycom.com/kb/search...1%200%204730251
Description
This white paper provides instructions on how to add a background logo to allSoundPoint IP and SoundStation IP phones in your organization.
This information applies to SoundPoint IP and SoundStation IP phones
running bootROM versions 2.x or later and SIP application version 1.x or later.
Introduction
You can add your company’s logo as the background logo of all SoundPoint IP and SoundStation IP phones in your organization. One bitmap file is required for each model; however, SoundPoint IP 301 phones do not support bitmap logos.
Model Width Height Color Depth
IP 300/301 n/a n/a n/a
IP 430 94 23 monochrome
IP 500/501 114 51 4-bit grayscale or monochrome
IP 600/601 209 109 4-bit grayscale or monochrome
IP 4000 150 33 4-bit grayscale or monochrome
Logos smaller than described in the table above are acceptable, but larger logos may be truncated or interfere with other areas of the user interface.
The SoundPoint IP 500/501/600/601 phones only support the four colors listed below. Any other colors will be approximated.
The SoundStation IP 4000 phone only supports black and white. Any other
colors will be rendered as either black or white.
Color RGB Values (Decimal) RGB Values (Hexadecimal)
Black 0,0,0 00,00,00
Dark Gray 96,96,96 60,60,60
Light Gray 160,160,160 A0,A0,A0
White 255,255,255 FF,FF,FF
Configuration File Changes
Warning: Polycom recommends that you create another file with your organization’s
modifications. If you must change any Polycom templates, back them up first.
For more information, refer to the “Configuration File Management on SoundPoint®
IP Phones” whitepaper at www.polycom.com/support/voip/.
Note: Use an XML editor to edit the configuration file.
In the <bitmaps> section of sip.cfg, find the end of each model's bitmap list and
add your bitmap to the end; do not include the .bmp extension:
<bitmaps>
<IP_300 … />
<IP_500 … bitmap.IP_500.66.name="logo-500" />
<IP_600 … bitmap.IP_600.70.name="logo-600" />
<IP_4000 … bitmap.IP_4000.70.name="logo-4000" />
</bitmaps>
Next, enable the idle display feature and modify the idle display "animation"
for each model to point to your bitmap (again without the .bmp extension):
<indicators ind.idleDisplay.enabled="1">
<Animations>
<IP_300>
…
</IP_300>
<IP_500>
…
<IDLE_DISPLAY ind.anim.IP_500.38.frame.1.bitmap="logo-500"
ind.anim.IP_500.38.frame.1.duration="0"/>
…
</IP_500>
<IP_600>
…
<IDLE_DISPLAY ind.anim.IP_600.38.frame.1.bitmap="logo-600"
ind.anim.IP_600.38.frame.1.duration="0"/>
…
</IP_600>
<IP_4000>
…
<IDLE_DISPLAY ind.anim.IP_4000.38.frame.1.bitmap="logo-4000"
ind.anim.IP_4000.38.frame.1.duration="0"/>
…
</IP_4000>
</Animations>
…
</indicators>
Finally, edit the {MAC}.cfg file to instruct the phone to download the bitmap
files at boot time:
MISC_FILES="logo-500.bmp" [for SPIP 500/501 phones]
MISC_FILES="logo-600.bmp" [for SPIP 600/601 phones]
MISC_FILES="logo-4000.bmp" [for SSIP 4000 phones]
Many configuration-generation systems do not make it easy to customize
the contents of this file based on the model; if you are using one of
these systems, you can have all phones download all the bitmaps:
MISC_FILES="logo-500.bmp, logo-600.bmp, logo-4000.bmp" [for all
phones]
Good Post will Experiment with this.
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This may be perfectly normal and may be associated with the NAND memory and the linus kernel operations. Only the OEM manufacturer of the hardware platform can truly confirm if this is a problem.
Santiago is your Cs410 in Warranty?
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what is the difference between the 'SIP Gateway' and 'Outbound Proxy' ?
Also, in both of these options there is a required 'password' field - what that field should be set to if the other end of the trunk is a simple cisco router sip gateway?
Thanks!
Here is a similar post that might help you.
http://forum.pbxnsip.com/index.php?showtop...art=#entry13274
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Hi
I installed pbxnsip today for testing.
I sucesfully register clients and they can communicate with each other.
The problem is with my trunk.I can use this trunk with my pap2t and no problem with it but when i try to use it with pbx "i got 404 not found" error
The trunk is registering succesfuly and no problem with dial plan.I tried lots of codecs but no way.
You can find log details below.
My dial plan format is : 90 + area code + tel number
Thanks.
[5] 2009/09/22 14:08:02: Dialplan Standard Dialplan: Match 9011902122691555@89.19.4.162 to <sip:902122691555@89.149.244.206;user=phone> on trunk proyturk
[5] 2009/09/22 14:08:02: INVITE Response 404 Not Found: Terminate 56bf2388@pbx
can you set the logfile to 7 so we can see more information. also set the log sip events on
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my friend has a PBX at home and want to use our PBX to terminate his call
how do i set up that he can register to my PBX and not as an extension, but as a sip peer
What type of pbx does your friend have?
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I have place my cell phone on the cell phone field in the redirection tab and selected immediate include. when an incoming call comes in via AA both the phone and cellphone rings. However, when I call in to use pbxnsip as my outbound call, I do not get the prompt for this that should say dial 1 for outbound calls but instead get the standard prompt. I am using the latest version that says on the setting "use only the last 7 digit of the cellphone. I just do not see what I am doing wrong. I have start and restart service but does nothing.
jose T
Hi Jose I was wondering if the cell redirection worked?
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Can we have access to your web portal?
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I have gone through the knowledge base and searched the forums, which shed a little light on this topic, but I'm still having a hard time wrapping my mind around the full concept and setup process.
I currently have 2 CS410's running the latest v4 beta in the testing lab that will be placed in the branch offices. I plan to purchase a windows pbxnsip license for HQ.
The system needs to do the following:
- All users in all 3 locations can intercom each other.
- Calls can be transferred from one office to any of the others.
- Users can dial out on CO lines of other offices.
Things I would like the system to do, but could live without if I had to:
- BLF possible for CO lines of other offices. (without registing phone to other office's pbx)
- Ability to pickup another office's ringing CO line (also without registering phone to other office's pbx... multiple registrations cause too many issues for me).
- BLF for phone extensions of other offices (again, without registering phone to other office's pbx).
Is this possible? Am I crazy for wanting to try it? Can someone point me to any documentation that covers this?
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Hi;
i have an offic with approx 90 users...what is the best way to setup the paging and how do i go about it?
i currently have them set on UNICAST but recently when i hit the extension number it just gives a fast beeping sound...i look in the logfile and it says it cannot find the extension but then says that paging would cause too many active calls so im guessing i have to go to multicast.
is there any easy way to set this up? my phones and server are using an ip address of 10.0.10.XX
Multicast for snom phone
create a paging account
on the destination add 224.1.2.3:12345 this is just a random port I used
save the changes.
on the snom phone go to advance /RTP
1 Scroll down to Mulricast
2 add the multicast range 224.1.2.3:12345 so the phones will listen. This has to be done on all the phones but its also by preference.
reboot snom phone
press feature code *90 and should page phones associated with the above ip.
In 3.4, please check the Extension's "Permissions" tab/page. We have added some controls over there. Also, if you have multiple registrations, then the intercom is not support.
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how would you use multicast on a hosted / multi domain solution?
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is Vista Compatible with pbx software?
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Hello,
Trying to intall PBX on vista. My Vista Platform says that its 32 bit os any clues?
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Hello,
Trying to intall PBX on vista. My Vista Platform says that its 32 bit os any clues?
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Installing pbxnsip does NOT install a softphone.
You download and install that seperately.
tx
matt
thanks Matt
will look into the trunk to set up outbbound call,will post soon.
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great, i checked my voice mail and set it up, know to my understanding u can only make calls with users,using xlite?
and one more question when i downloaded a pbx software, i couldn't see the pbx soft phone on my windows xp any suggestion
thank u again 4 replying
My steps to testing a systtem:-install and make sure you can login to the system admin.
-add 1st extension and make sure it registers (by calling a attendent or voicemail
-now add 2nd extension and make sure you can call the 1st and you have good audio.
-now add trunk and make sure you have a outgoing dial plan, test outbound call.
to setup voip successfully its good to have basic (intermediatte?) networking skills. Ping is just checking basic IP connectivity between to ip devices. ie=Ping xxx.xxx.xxx.xxx (where xxx is an ip address of remote pc you want to check if you can connect to)
matt
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To keep things simple:
#1- lets forget about callcentric for now
#2-Is the pbxnsip install on one computer and the softphone on another? is it windows?
#3-can you ping the server (pbxnsip) server from the pc with the xlite installed?
#4- i think by default there is an extension 40 with password of 40. Is that what you are using to setup xlite? (softphone)
#5-if you dial 840 (to dial voicemail) do you get the voice mail prompt?
some quick ideas
matt
#2 the pbxsip is install on my windows but i don't have the software show up in my programs has a standalone application
the soft phone is the only thing visible.
#3 not sure how to ping unless through a tutorial.
#4 has far as the extension is concern i followed instruction on callcentric.com that lead me to create an extension withing the pbx admin site.
#5 i just dial 840 and i did get the voice prompt
I was trying to make an outside call but, must i dial the whole number or go to pbx admin and add a short cut for the whole number?
Thanks Matt
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HI YA'LL
so this is the link to start
http://wiki.pbxnsip.com/index.php/Setup_the_free_demo_system
know I followed the steps as describe
Once the phone is registered and the trunk is registered (200 OK in status screen) will be stated in the pbx interface,
now, when i try to make a call on the soft phone, it says the call failed, it also says user does not exist, know i followed the call centric tutorial on how to set the pbx extension and trunks, why am i getting this error?
I am nooob so can anyone light sum
light hrr.
light hrr.
Polycom PNP
in Polycom Phones
Posted