similar to: Polycom 501: takes calls, but fast busy on dial out?

Polycom IP501 Assembly: 2345-11500-040 Rev: B Bootrom: 3.1.0.0269 SIP Ver: 1.6.5.0043 I can dial other extensions internally, and can get to voicemail, but when I try an outside number, I hear dial tone, the digits dialed, yet nothing happens when I press "Send". Nothing appears on the Asterisk CLI screen. This phone worked briefly before upgrading from br:2.6.1 and sip:1.6.2

Hello, Here is a breakdown of the issue I am experiencing. I have three remote employees, in various states, who have Polycom 501 phones. They are unable to receive incoming calls after a few minutes of the phones being plugged in. They work immediately after being plugged in, but they lose the ability shortly thereafter. They can always make outbound calls, but only to real phone numbers, not

Has anyone successfully made this scenario work in 1.4. I found info at http://www.voip-info.org/wiki/view/Asterisk+presence indicating that this does not work with 1.4 implementations. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100903/13ed20ba/attachment.htm

I am having a strange problem with polycom 501 and dailing. I've read the archives and no one there specifically mentions this problem, so I thought I'd ask here. The problem is that when the user picks up the receiver or pressed new call, sometimes they will enter a number (for example 95072091234) and in the middle of the number the cursor might jump back one digit. So the call above,

This is not necessarily Asterisk specific but if I have Polycom 301/501 and 601s and want to dial a missed call back, how do I prepend a 9 - can I do this via the polycom config? I can't find anything in the docs. Bill -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I'm trying to see if I can get the hint priority working with my polycom 500. So far I have 2 </reg> entries with the same sip registration, one is labeled as private, the other as shared. I have set the hint priority before anything else in my dialplan for my extensions. As it stands, I have two registrations on the phone, one has a half greyed out phone icon, the other

...even when I do the "factory reset" (4-6-8-* then 456). I tried using FTP and TFTP, but even though the phone uploads the log, I get these errors: 0927211350|app1 |3|00|Time has been set from 0.us.pool.ntp.org(69.60.124.59). 0927211350|cfg |4|00|Could not get all 512 bytes of the header. 0927211351|cfg |4|00|Could not get all 512 bytes of the header. 0927211422|app1 |4|00|Loaded

When I want to pick up a ringing line, I dial *8 and hit New Call softkey on my Poly 501. For some reason, if I pick up the hand set and dial *8, it seems to ignore or drop the 8 digit. I've confirmed that this happens with all of my 12 Polycom 501s. Does anyone know what would cause this or how to fix it? Craig Bruenderman Network Advocates, Inc. 300 Envoy Circle Suite 300 Louisville, KY

I'm having some difficulty setting up a Polycom 501 with two different extensions. I've set up in the phone's xml file the proper reg.1withlineKeys = 2, and reg.3 with lineKeys = 1. I have also set up the sip.conf file on the Asterisk box correctly. However, the phone is not making any sort of registration request to the Asterisk server. Is there something else that needs to be

Howdy - > The transer button on the polycom phone does not seem to transfer/park the > call properly. I have to use the # -> 700 to park the call. If I recall, using the Polycom transfer, you have to make sure it is done as a blind transfer. The Polycom attended transfer (default) option does not work. > Furthermore the # -> 700 only works on incomming calls. If I

Hi Andrew - > On Monday 13 March 2006 10:20, Noah Miller wrote: >>> The transer button on the polycom phone does not seem to transfer/park >>> the call properly. I have to use the # -> 700 to park the call. >> >> If I recall, using the Polycom transfer, you have to make sure it is done >> as a blind transfer. The Polycom attended transfer

Hi - > I am not sure what I did but blind transfers do not work. The Polycom does > not allow me to dial the extension of the person I want to transfer to after > I hit: > > transfer -> blind I would strongly suggest getting the latest firmware, and using the sample configuration files with that firmware to set up your phone. This SHOULD work. If it still does not work

I've been noticing that my group of Polycom IP 501 phones seems to randomly reset themselves nearly every night (I guess it usually happens at night, since I've never seen it happen while I've been at work during the day).. When I say "reset", I mean, the hands free volume and ring volume are set to the default and the call logs (received calls, missed calls, placed calls)

I have sent a Polycom IP500 phone to an overseas remote user who has a Speedtouch adsl modem/router. The phone is connected to the 4 port router which performs NAT for the network. The phone rboots, find the server, downloads it's config and tried to register. Then it immediately becomes UNREACHABLE. Jul 26 17:44:41 NOTICE[21026]: chan_sip.c:9138 handle_request_register: Registration from

> Hmmm... I have this aweful feeling that I'm choosing the > exact wrong time to ask a "newbie question" :) Oh well, here > it goes. > > The quick question is : "How do I dial an extension?" > (answer is probably - "you don't" in which case:) "How do I > dial my asterisk box?" - I have no outside line, I just want >

This is a dialpaln issue. I solved the same problem recently. For 4 digit extensions you need to append the dialplan statement in the sip.cfg configuration file as follows <digitmap dialplan.digitmap="[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2 -9]xxxT|1xxxT" dialplan.digitmap.timeOut="3"/> Michael > I am not sure what I did but blind transfers do not

I have a soundpoint 501 phone that has locked up twice now. You can make a call but when a call is sent to it, it responds with sip busy messages. You get the same message when the phone is in do not disturb. I reset to defaults the first time and it worked for a week or so and then stopped. The incoming calls are ringing three phones ( dial(sip/1&sip/2&sip/3 ), often two of them are in do

Hi List; Can someone advise me if Polycom support H323 that work fine with Asterisk? And wether this H323 Polcyom devices more costly than SIP Polycom. Also, I am not able to know if new Polycom come with PoE adaptor so no need for PoE Switch (can use normal switch that does not support PoE)? Do I need any special cable for Polycom or normal Ethernet cable? Regards Bilal

Anyone know how (or if it's possible) to get a logo on the screen of a Polycom 501? I've been looking around for hints on how to do it but so far nothing would indicate Polycom supports doing it. -bill

Hi We've setup an Asterisk PBX recently and I encountered the following problem: When [mac address]-registration.cfg file includes the FQDN of the Asterisk PBX for the Polycom SoundPoint 501 phones it will not (even try to) register with the Asterisk PBX unless the DNS (it asks) successfully resolves the name: _sip._udp.[Asterisk FQDN]. Did this happen to anyone else? PS - The