similar to: 401 Unathorized

On Mon, Mar 9, 2015 at 9:40 AM, David Wessell <david at ringfree.biz> wrote: > Welcome to our hell. > > We ran into this on VVX 300 and 400 phones running UCS 5.2.x. We finally > got Polycom to issue a hotfix firmware version. I'll be happy to share it > with you offlist, just email me. > > Officially Polycom will fix the issue in 5.3 in a few months.. > >

Hi Guys, We are getting a strange issue on certain polycom phones, sometimes when a call comes in it just "flashes" to show there is a call but does not play any sound. This problem is very intermittent and happens to maybe 2 out of 10 calls. Has any else experienced this before? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Greetings- As many of your are Polycom "experienced", I was hoping some kind soul could provide direction on a specific issue. On a system running Asterisk 11.11.0 (and latest FreePBX), I'm finding an instance where, using intercom/paging functionality of FreePBX, I need to override an end user's 'Do Not Disturb' selection on the handset. By default, DND simply

Welcome to our hell. We ran into this on VVX 300 and 400 phones running UCS 5.2.x. We finally got Polycom to issue a hotfix firmware version. I'll be happy to share it with you offlist, just email me. Officially Polycom will fix the issue in 5.3 in a few months.. Thanks David On Mon, Mar 9, 2015 at 9:34 AM, Andrew Colin <andrew at convergedgroup.net> wrote: > Hi Guys, > >

Matt, At some point you need to consider how much is too much... I run a call center with more then 125 commissioned phone sales reps and more than 60 customer service reps. We run dual servers, fiber from one provider and 6 bonded T1's from another provider. We purchase our so trunks from a wholesale company who is a major provider to resellers. Being so, their network is extremely

Hello; To be able to send and receive faxes through asterisk and to be able to have trunk with google voice and to be able to have integration with those that support Jaber .. What are the channels and libs that I have to be sure that they are existed? Regards Bilal

We have licensed both products and sent a support request on 6/11, with zero reply or any activity on it at all so far. No replies to subsequent ticket updates or e-mails. -- Carlos Alvarez TelEvolve 602-889-3003 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130617/c0e347d9/attachment.htm>

Hi people I'm just mailing to see what people are using for CTI solutions with asterisk. Aslos, has anyone managed to integrate asterisk with Salesforce? Thanks in Advance Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: ish at pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2,

Getting "no service" display on aastra 480i. Sip debug shows an "unathorized" blub when the aastra tries to register. Some reading indicates that 1.4 firmware wants aastra.cfg and mac.cfg in /tftpboot/. There are none. Anyone have basic config files? Or can point me to a good link? All links I have tried, that purport to have config files, are either dead or error out.

Is there any integration for ACT! and asterisk? I've googled for hours and haven't been able to find anything. Thanks David [cid:image001.png at 01CBE88E.66E8E450] -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110322/ecc019cc/attachment.htm> -------------- next part --------------

hello i am a new user for this group. i am working at a ISP. here i want to made a bridge firewall i am using fedora core 3. i want to block a serirs of ip address 192.16.18.0/255.255.255.0 and want to give the accesss only 172.16.18.0/255.255.255.0. but iptables not be able to block ip;s its passes all the ip series. i made my machine as bridge. i think my bridge passes all the

A point of clarification. The "get started video" on asterisk.org says that running asterisk on a virtual platform is a popular option. "Go for it!" the presenter says. However, the 2011 third edition of Asterisk, The Definitive Guide says they don't recommend it for production use, though many do so successfully. I also don't find may recent discussions of this

Hello H. Peter, we use pxelinux to start linux on desktop pcs. Now we have a problem with the interaction of dhcp. If the ip address is not fix, we cannot use the ip address as name for the configfile. A better solution would be, first to check for the mac and then for the ip address. Is it possible? TIA, Tobias

Hi all I have a problem to register a cisco 7960 to an asterisk 1.2.2 I defined in sip.conf the next : ["phonenumber"] type=friend username="username" secret="password" host=dynamic context=work I am trying to catch the register requests with sip debug with no success (empty screen). I can only catch the register messages with ngrep on host it's comming

Hi all I have a problem to register a cisco 7960 to an asterisk 1.2.2 I defined in sip.conf the next : ["phonenumber"] type=friend username="username" secret="password" host=dynamic context=work I am trying to catch the register requests with sip debug with no success (empty screen). I can only catch the register messages with ngrep on

It seems that /etc/locale.conf is ignored in Centos 7. As a traditionalist who prefers things sorted lexicographically rather than indiscriminately with case ignored and dates to be displayed in the form "Sep 11 2008", I have always added lines to this file: $ cat /etc/locale.conf LANG="en_US.UTF-8" # Fix collating sequence for sort and ls export LC_COLLATE=C #

Is it possible to change error messages for Validation helpers? I am writing an app against a existing database (so no control over column names), but when there is validation error (e.g. with validate_presence_of) I would like to customize the field name. For example for telephone whose field name is PhoneNumber I would like to chnage it to "Telephone Number cannot be empty" rather

Hi I am using asterisk version 1.6.0.5 I have build up one utility that will fire Originate Action on Manager... In which, i have define number to call eg. 919912312345 (MobileNumber) How can i know that this number format is true for Indian Number... In originate action, user can enter any international number.. How can I came to know this number format is right for that country...?? IS there

Today, We have added INVITE Authentication. This seems to bring a large amount of problems to people in the way since they can't make outbound calls. Here's what needs to be done. You need to add three variables to your peers or friends, username, authuser, and secret. username=<phonenumber> authuser=<phonenumber> secret=<registration password> Dan

Hello, Can anyone suggest us how to got the phonenumber of the caller. In the environment variable, I just see the ip of the gateway. Environment: 'agi_callerid' is 'XXX.xxx.XXX.xx' Should I do some changes in the GW conf, or is it just not possible when we got a call from PSTN ?!? Thx in advance, Ares