Hey Guys,
First post here so be gentle :)
Does anyone know if it is possible to use the Auto Provisioning tool to manipulate the buttons on the ACTUAL phone itself for the Polycom and Aastra phones? It appears to only add BLF's to the expansion module. I want to be able to make the change to the phone buttons themselves, and then any BLF's that can't fit on the phone buttons be pushed to the expansion module. I can manually do this in each phones cfg file, but that would become very time consuming AND be overwritten.
Thanks for your help!
Eric, Thank you for your
Eric,
Thank you for your quick response. Sorry for my late one. However when I do this, the LINE portion of your how-to works perfectly but no matter what I do, I cannot get the BLF portion to show up. Is there an additional step somewhere that I am missing?
Here is my thirdlane-settings.cfg file:
voIpProt.SIP.alertInfo.1.value="Ring Answer"
voIpProt.SIP.alertInfo.1.class="4"
voIpProt.SIP.alertInfo.2.value="Internal"
voIpProt.SIP.alertInfo.2.class="5"
voIpProt.SIP.alertInfo.3.value="External"
voIpProt.SIP.alertInfo.3.class="6"
se.stutterOnVoiceMail="0"
se.appLocalEnabled="1"
se.rt.enabled="1"
se.rt.modification.enabled="1"
se.rt.4.name="Ring Answer"
se.rt.4.type="ring-answer"
se.rt.4.timeout="500"
se.rt.4.ringer="13"
se.rt.4.callWait="6"
se.rt.4.mod="1"
se.rt.5.ringer="4"
dialplan.digitmap="[2-9]11T|0T|011.T|[0-1][2-9]xx[2-9]xxxxxxT|[2-9]xx[2-9]xxxxxxT|[2-9]xxxxxxT|xxxxxT|xxxxT|xxxT|*xxT"
up.oneTouchVoiceMail="1"
msg.bypassInstantMessage="1"
msg.mwi.1.callBackMode="contact"
msg.mwi.1.callBack="*98"
voice.volume.persist.handset="1"
voice.volume.persist.headset="1"
qos.ip.rtp.dscp="EF"
qos.ip.callControl.dscp="AF31"
feature.1.name="presence"
feature.1.enabled="1"
feature.9.name="url-dialing"
feature.9.enabled="0"
feature.12.name="directed-call-pickup"
feature.12.enabled="1"
feature.16.name="nway-conference"
feature.16.enabled="1"
feature.17.name="call-recording"
feature.17.enabled="1"
feature.18.name="enhanced-feature-keys"
feature.18.enabled="1"
feature.19.name="corporate-directory"
feature.19.enabled="0"
softkey.feature.mystatus="0"
call.directedCallPickupMethod="legacy"
call.directedCallPickupString="*8"
efk.version="2"
efk.efklist.1.mname="callrecord"
efk.efklist.1.status="1"
efk.efklist.1.label="Call Recording"
efk.efklist.1.action.string="#9"
softkey.1.label="Record"
softkey.1.action="!callrecord"
softkey.1.enable="1"
softkey.1.precede="0"
softkey.1.use.active="1"
presence pres.reg="1"
prov.polling.enabled="1"
prov.polling.time="03:00"
se.pat.misc.1.inst.1.type="silence"
Let me know your thoughts. Thank you so much.
I think the problem I am
I think the problem I am having is the 000000000000.cfg file was replaced when I tried to upgrade my phones and now the files that it tells the phone to look for override any thirdlane files. How can I fix this?
NO matter what I set in
NO matter what I set in sip.conf phone_313.conf and thirdlane-settings.conf (my tenant) SOMETHING is overrriding it as the inbound caller id is also still showing number@ipaddress as the inbound caller ID. I have disabled URL dialing in every place I can think of. So both of these issues are the same issue I suppose. Any thoughts?
wait, your using 313? that
wait, your using 313? that features isnt damn supported until 3.2.0 are you using some archaic ass phone or are you using an IP650? you should be using firmware 3.2.6 on an IP650
Actually this particular
Actually this particular phone in question is the VVX 1500. But all my customers use the IP670 phone.
I upgraded the phone to the 3.2.6 firmware last night but I am still unable to get the URL dialing removed. I suppose the phone_313.conf was being used when I had the wrong files uploaded.
according to polycom the
according to polycom the vvx1500 doesnt even work with asterisk.. gone one of those 670s laying around?
Little update to this. And I
Little update to this. And I really feel stupid wasting so much time on this because the answer is so obvious but here it goes:
The reason the URL dialing option wasn't disabling is because I was using the IP670 template for the VVX1500, which did allow the phone to register and work fine for the most part but the feature enable for URL dialing on the 670 isn't the same feature for the VVX so by me choosing feature9enable=1, feature 9 on the 670 isn't the same feature 9 on the VVX so disabling it was turning something else off or on.
if you end up actually
if you end up actually getting the vvx1500 to work let me know. both digium and polycom claimed it wouldnt work and in the 2 yrs that followed, never updated anyone if they, in fact, made it start working.
Eeman, It does work almost
Eeman,
It does work almost 100 percent by using the 670 template. I actually use the VVX everyday as my desk phone. The only thing I have ran into so far is the issue with inbound caller ID being a num@ipaddress. Changing the URL dialing for the feature in sip.cfg isn't changing it on the VVX so it's simply a matter of finding the right feature to change.
I will update you on this.
Yes Video. I have actually
Yes Video. I have actually been doing a lot of testing on the Grandstream line and I use the VVX as one of the two on the Video Call. Haven't had any issues with it. I've been testing with h264
on polycom's yes.. thats what the field 'span' does .. lets say you wanted line keys 1 and 2 to be exten 101, line key 3 to be exten 102, and buttons 4-6 to ber BLF keys
then your setup would look like this..
Button/Registration Sequence : 1
Type: Line
Span: 2
Label: 101
Registration: 101
Button/Registration Sequence : 2
Type: Line
Span: 1
Label: 102
Registration: 102
Button/Registration Sequence : 1
Type: BLF
Span:
Label: 105 (or whatever you want to appear on display)
Extension to watch: 105
Button/Registration Sequence : 2
Type: BLF
Span:
Label: 106
Extension to watch: 106
Button/Registration Sequence : 3
Type: BLF
Span:
Label: 107
Extension to watch: 107
etc etc