joso
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Posts posted by joso
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Joso, if you are not particular about the HTTP provisioning, you can follow these and get the phone provisioned
On the PBX:
1. Put the MAC address of the phone under the extension's "Registration page->Bind to MAC Address" field
2. Admin->Settings->PnP page- Trust MAC address ON
3. Admin->Settings->PnP page- point to the correct version of the firmware, something like http://provisioning.snom.com/download/fw/s....3.14-SIP-f.bin
On the phone, under Advanced->Update page
1. Update Policy: Update automatically
2. Settings URL: tftp://pbx-ip
3. Subscribe config: off
4. PnP Config: off
Hit "Save", then it will ask to reboot. After the reboot is successful (it goes through couple of reboots), you should have the phone registered.
One thing i noticed is when upgrading to 3.3.1, my pnp parameters are missing. the html folder holds all my settings. even internally the phones don't pick up the settings i have.
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same error:
on pbx
pbxnsip:/opt/pbxnsip# tail -f /opt/pbxnsip/pbxnsip.log
[5] 20090331110323: Web Server: File snom370/snom370-firmware.htm not found
on snom
[5]24/12/2001 00:00:44: Conf setup: return code 400; requeueing http://xxx.xxx.xxx.xxx/provisioning/snom370.htm<; attempt: 1, state: 32
[5]24/12/2001 00:00:47: Fetching URL: http://xxx.xxx.xxx.xxx/provisioning/snom370.htm
when i use an standard webserver (http://webserver/snom.xml) with the snom xml config file - it works!
it is not a snom problem.
where is the error?
I am getting the exact same error as novo using the 3.3.1.3171 build.
Joso
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I have created a service flag for the Night service in the hunt group. i have setup the ext 300 to be abe to change this service flag through a button and it lights up the led based on the manual service flag. The weird thing is when the service flag is "Cleared" it goes to night mode, when the service flag is "Set", it goes to the receptionist and the led lights up during the business hours. I am thinking this is the opposite of how it should work shouldn't it? The led should light up after hours in "Set".
Any help would be appreciated.
Joso
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Using the latest firmware of the audiocodes mp-118 fxo and latest copy of pbxnsip i was able to get everything to work as it should except for one glitch. It seems the volume levels are really low even when i up the voice settings in the audiocodes. if i up the settings too much, i get static. What else is unusal is when i dial out, the first ring is extremely loud, but the rings after the 1st one is very low and it also seems that when the second ring starts that is when i get connected. any ideas on what i can do?
Thanks
Joso
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hi there, i seem to be getting this in my event log on exchange when i use the play-on-phone feature. has anyone seen this?
"An outbound call to "4257106472" could not be established. The selected outbound IP gateway "192.168.2.7" returned this error: "404 Not Found". The caller ID for this call was "josog@jngconsulting.com". For help troubleshooting the SIP response error code that was specified in the event description, contact the vendors who support your IP gateway or IP PBX hardware. You can also run diagnostic tests on your IP gateway or IP PBX hardware to make sure that the devices are operating correctly. Additional information: "A SIP request has failed. The current operation is 'Opening'. The session state is 'Connecting'. The remote participant is 'sip:4257106472;phone-context=PBXNSIP.JNGCONSULTING.COM@jngconsulting.com;user=phone'. The response code was '404'. The response text was 'Not Found'."
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does this support OCS 2007?
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i was curious as to how remote call control was going with pbxnsip? I heard that CSTA is in the works.
Thanks
Joso
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thanks for the info, we actually purchased a ucc certificate a few months ago from entrust and use this on our production exchange 2007 server. when i put ocs and pbxnsip in production, is this the cert i can use? or do i have to purchase another one for ocs. the key is, that i will be setting up an echange edge role and ocs edge in the dmz when i delpoy the phone system, this will be on one server with ocs virtualized (with windows 2003 32bit) and exchange edge role on the host windows 2008 hyper-v. Can is set it up this way? or does exchange and ocs have to be usign the same public ip for external access?
Joso
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would a single self signed certificate work? or would i need to get a UCC (SAN) Certificate?
Joso
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actually the play on phone doesn't work. i should be able to hit the Message Button on my sip phone?
I am not using a tanjay type phone.
It still shows ' send to you via Exchange Server".
I will check for some updates for ocs
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on your first post, you quoted: "deploy an OCS Edge Server in the DMZ. Use Office Communicator, Tanjay's (like the Polycom cx700, LG-Nortel IP540) or the Snom OCS Edition."
could i use a jabra headset instead of a hardphone or a snom 320 phone? for the external users with communicator?
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it looks like i got this to work with ocs.
the only glitch i found was when i dual fork my hard phone with office communicator, both ring at the same time, but when i answer the hard phone instead of communicator, i get an email with a missed call notification.
Joso
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Hello joso,
sounds familiar for me I guess what you described is per design. But let's clarify this a bit. Did you integrated Exchange UM to OCS or did you integrated AudioCodes to OCS and to Exchange UM in parallel?
Normally Office Communicator can place you a missed call notification in your outlook. If, you can see this in a underline, saying "send to you via Office Communicator".
Please recheck your Personal Information Manager settings in Office Communicator!
If a call reaches Exchange UM and you hangup or leaf a voice message you will see the underline " send to you via Exchange Server".
Best regards from Barcelona (Tech-Ed EMEA)
Jan
I integrated AudioCodes to OCS and to Exchange UM in parallel.
and it show that the notification was sent by exchange. how do i change this?
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Thanks for the info.
wow, 3000 dollars, quite steep, but my boss might go for it.
what other option do i have in terms of conference phones? Polycom have a good option?
Joso
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With the Audiocodes mp-118 and exchange w/ocs, i noticed that when i call in to the extension, ocs communicator shows the popup with the new call which is good, but if i hang up within the first or second ring before the voicemail prompt comes on it does not send an email with a missed call notification. if i wait until the exchange auto attendant comes on and hang up before the beep it sends the email.
Any idea what could be causing this.
joso
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Hello Joso,
maybe this comparison fit's you expectations: http://www.provu.co.uk/Snom-comparison-table.html
Only the M3 and 820 are missing.
best regards
Jan
Thanks, that's exactly what i was looking for.
Would Microsoft Roundtable work for the boardroom, instead of waiting for snom to come out with theirs?
Joso
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Thanks for the reply.
I can't seem to find the reason for dmz going against best practices within the document. Currently we only have 1 cisco firewall, not 2
What about conference units, i didn't see snom having a unit for boardrooms.
would you happen to have a comparison chart for snom models as i couldn't find one.
Thanks
Joso
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thanks so much, that was exactly what i was looking for.
Quick question, i am looking at a polycom phone for the receptionist with OCS which she is capable of forwrding all calls to different users since we don't have direct numbers (She answers all incoming calls to 1 number). I was thinking of the cx700 for her, or is that overkill for her.
Also, what is the recommended conference phones for the boardroom? Preferably polycom.
Thanks
Joso
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hi there, i was curious to know how someone external (with a polycom phone and headset for ocs) would access ocs and pbxnsip that is internal. in this guide at the bottom http://wiki.pbxnsip.com/index.php/Office_C...ications_Server it has a diagram showing a sample setup, the only thing external is the ocs edge server. we are currently testing the same kind of setup and would like to know how we would go about setting up remote users to access (pbxnsip or OCS?) to the DMZ (What server should be here) first?
Joso
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With our current phone system, while in night mode, calls get forwarded to a couple of phones in our office, but the original caller id from the external user does not show up. We would like to see the caller id after the redirection from the main reception.
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Well, currently that is not possible... We have "vanity" ringback tones somewhere on the todo-list. Maybe if we address this topic we should also consider having them dependent on the time of day.
How about the caller id feature? When in night mode, do we specify which phones to ring and does it show up with the actual caller id of the person?
Joso
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Hi there,
One of the features that was requested by my collegues was when night mode is set, the handets ring a different tone to all users in the office. So the user is able to tell whether the call was forwarded (with the regular ring tone) and the night mode ring (another tone)
Not only that, they want to be able to see the caller id of the original caller when it's in night mode.
Are these features available?
Thanks!
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I think i found the problem. In the audiocodes mp-118 under coders. I had it set at G.711U-Law, when i changed this to G.711A-Law it works fine. Silence SUpression has always been disabled.
What setting should this be set at, since there are many to choose from.
Joso
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It looks like i got it to work for testing.
I believe we will be purchasing polycom phones for delpoyment.
Thanks!
autoprovisioning snom
in snom Phones
Posted
i am referring to Admin->Settings->PnP page
as of now, i have reverted back to 3.2 until 3.3.2 comes out. with 3.2, my polycom conference phones work and my snom's provision great over http (remotely) as well as internally. Although i would like to use 3.3.X because of some nice features, especially customized email messages.
Thanks
Joso