Friedom-Tech
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Posts posted by Friedom-Tech
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Well, that is the default file which is already included in the 3.1 build. If you want to have a different behavior than now, then you need to edit it.
to do what i want...be able to transfer to voicemail of extensions beginning with 1, what needs to be edited?
pls advise
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It is not clear what is causing this. But one potential reason has been identified. When you are logging to the file system, that write/append operation may block the RTP thread. Unless you are tracking something down you should not write log files to the file system. Internal logging to the web interface is fine.
ok i have turned that off but possibly in the future it should be an option to be able to write the log files...
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what OS?
Just curious how many calls do they have up? is it a call center?
on average upto 15 calls simultaneous
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This is very true, I tell customers to make the ring value 30 seconds.
pbxnsip, personally i think you should NOT read back the callerID info. other phone systems that do this mearly play a recording that says "you are receiving a forwarded call press 1 to accept" customer can forward the cell phone ANI if they need that info (i assume).
i think it is a nice feature because not always do you want to pick up calls from the office but if you dont know who it is you feel obligated to pick it up.
i would say try doing two things:
1) give the option to turn that on or off ( the users who are pushing ANI it is a waste)
2) speed up the playback a little its currently at snails pace!
let us know!
Thank you for all the great additions!
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Hi;
in the latest version you added a nice feature with the option of requiring the cell phone user to hit one as a confirmation that they received the call.
this feature also adds the option of reading out the caller id of the person calling, the only problem is that if the redirect to voicemail is set for 15 seconds and you have it ringing to cell phone after 5 seconds then by the time it finishes reading out the number the caller has been redirected to vm!
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i got this message today only once but this is a big company so i want to figure out what would cause this.
how can i find out? please advise
CPU meter shows 76, which is over the threshold of 75. Therefore, a new call has been rejected
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You can put the attached file into the html directory and edit it. This requires a restart of the system.
do i need to edit it or is it good to go?
if i need to edit it what fields do i need to edit in it?
pls advise
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That looks like what I would expect. Looking at your original post it becomes clear why she cannot dial 8121: There is no pattern for that. The point is that extensions must start with 2-7 for the Polycom dialplan. Otherwise the above dialplan is not usable.
Is it an option to move the 1xx extensions to another location?
can you modify the dial plan at all? when they call the extension directly they have no problem so i would assume that all that needs to be done is set the dial plan to allow four digit extensions beginning with 8?
pls advise
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It is in polycom_sip.xml. Search for "dialplan.digitmap", there you should see the dial plan that your phone has. You should see something like "[2-7]xx|8[2-7]xx|[2-9]11|1xxxxxxxxxx|011x.|*x.".
here is the dialplan:
dialplan.impossibleMatchHandling="0" dialplan.removeEndOfDial="1" dialplan.applyToUserSend="1" dialplan.applyToUserDial="1" dialplan.applyToCallListDial="0" dialplan.applyToDirectoryDial="0">
<digitmap dialplan.digitmap="[2-7]xx|8[2-7]xx|[2-9]11|1xxxxxxxxxx|011x.|*x." dialplan.digitmap.timeOut="3|3|3|3|3|3"/>
<routing>
<server dialplan.routing.server.1.address="" dialplan.routing.server.1.port="5060"/>
<emergency dialplan.routing.emergency.1.value="911" dialplan.routing.emergency.1.server.1="1"/>
</routing>
</dialplan>
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BTW what phones are you using? Did you chech what dialplan the PBX sends to the phone in the generated directory?
i am using a Polycom 650. i have about 80 phones (mostly Polycom 320's) registered through PNP...
the four files that the system is showing in the generated folder are
1) Polycom_Sip.xml
2)Polycom_Phone.xml
3)Polycom_master.xml
4) Polycom_Addrbook.xml
is the dial plan included in one of these?
pls advise
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HI;
when the receptionist (using a polycom650) picks up her phone to make an outbound call and has entered the first 7 digits of the number (for example) and then a incoming call comes in, it automatically picks up the call as she has the handset in her hand and has a "dial tone"
is there anyway to set the phone/system to allow her to continue dialing the number even though a call is coming in?
pls advise.
ty
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Well, as said before if you want to use the PnP dialplan on the phone you must use accounts and extensions starting with the digits 4-7. Starting with "1" means that the number will be a 11-digit telephone number in NANP area.
is there any config i can make to the dial plan to take this problem away?
besides in this case the employee is not beginning with a 1 they are beginning with an 8 as they are transferring to a mailbox...they are entering 8 followed by the three digit extension beginning with a 1!
pls advise
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The question here is if this is a problem with the dial plan on the phone or a problem on the PBX. From what you write above, it sounds like a problem with the phone. What you can do is choose that the user has to press enter; alternatively the user must choose extension numbers that start with 4-7. This what most "good old" TDM PBX did as well. The alternative would have been that the user must enter "9" to get a dial tone, which requires one more key press and messes up the address book dialling. Therefore, for the PnP we chose "1" for "dial a NANP" number (10 digits).
Using the enter key has also one big advantage compared to the "get dial tone" method: You can edit the key, especially the last digit. And you have no problem dialling international numbers.
HI;
in this case the user does not want to have to hit enter OR dial 9!
even in enviornments that use 9 for outbound calls i have never seen a system that requires that for internal transfers.
it is very possible that this is an issue with the dial plan on the phone but mu question is how do i get around it?
pls advise asap
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Not yet - still eating Christmas cookies...
Lol! Please let me know when you are done with that as this is an urgent issue for them!
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Hmm. You should also upgrade to 3110 (the latest release), but honestly I am nore sure if that will fix the problem. I guess we have to re-test this in the lab.
Hi;
hope you had a nice Christmas....any update on this issue??
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Did you switch to the TCP transport layer (see http://wiki.pbxnsip.com/index.php/Polycom, "Switching the transport layer")?
Es. As I said this was working great until I made two upgrades. One was the upgrade to .3107 and the other was the firmware upgrade of the phone to 3.1.1.
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I don't know what I did wrong, but I can't have the cs410 to the right date or time. dns servers are in there, pool.ntp.org as ntp server, it pings ok, yet this cs410 box insists the date and time is 2008/01/01 07:20:28 when as far as I can tell it is actually 22 of dec at 8:56 am.
any advice would be appreciated. Season greetings to all.
dlb
Hi. I have had this problem with a CS410 before..have you tried restarting the system? That always did the trick for me.
Let me know.
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I had the same problem happen to my 650 with two side cards and I opened up a ticket with Polycom and they couldn't figure it out. The BLF's would stay on and then not respond. After a reboot of the phone it would work again for a bit. I sent them a trace where we sent them a notify and they sent back a 503 message. Can you open up a ticket with Polycom and I will open up the ticket again with them to put some pressure on them.
i can definately do that however the strange thing is that it was working fine until last week when i upgraded firmware on phone in order to fix a different issue!
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You would put in the same address and port, that you put in the PBX multicast. see http://wiki.pbxnsip.com/index.php/Paging
Actually i spoke to Polycom who advised me that Multicast paging is not currently offered with Polycom phones. they say that it is in the plans though as many people have apparently requested it.
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What polycom phone are you using ie. 550 or 650 and how many side cars? What version of s/w is on the polycom.
Two receptionists with Polycom 650 with two sidecars each.
the issue only began last week when i upgraded to version 3.1.1 of Polycom firmware.
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i am using Polycom phones....so in the address field what ip address do i put there? sorry but the concept is new to me...
can someone please help me set this up...this client needs the paging urgently but i cannot use unicast as there are too many users.
please advise asap. i am using Polycom phones and need to know which multicast address to put into the system.
PLEASE HELP.
thank you;
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HI;
i have a client who who works with extensions beginning with 1XX and 2XX. when the receptionist tries to tranfer directly to voicemail for the users beginning with 2 IE 8201 she has no problem but when she tries to transfer to users begininng with 1 it never works...i beleive that after the third digit it is automatically doing the dialing ie she enters 8121 it only recognises 812 so it says invalid extension...i would imagine this may have to do with the PNP plan which is set to three digit extensions but then why is there only a problem with extensions beginning with 1??
please advise asap.
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Hi;
i am using version .3107 with the latest phone firmware...however the receptionist is complaining that the call monitoring is not working....the lights sometimes work and sometimes dont work.
this is a biog issue for them as it is an office with 80 users.
pls advise asap
PLEASE NOTE: until we upgraded to version .3107 and firmware 3.1.1 the sidecars were working perfectly.
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Whow. Any more insight? What operating system? Is the system clock okay?
Windows Sercer 2003 Web Edition...the time is correct.
Require to Restart PBXnSIP server
in General Setup
Posted
i have had this issue in the past and it fixed the issue when i unchecked the SIP ALG protocol on the router.