similar to: Polycom phones and ring no answer/302 Moved Temporarily

Anyone have some recommended equipment for alerting people to calls in a noisy environment? I have Polycom IP550 phones set up in some really noisy environments - our mine hoists - and they tend to drown out the ringers. I'm using Clarity WR100s now. They're analog devices, attached to Linksys PAP2T ATAs as part of a call group to get a loud (advertised as 95dB) ring out there, but it

Has anyone successfully integrated Asterisk with an Inter-tel Axxess phone system via a SIP trunk using the IPRC card? -- Marvin Horst -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101006/5dbe435a/attachment.htm

We're running Asterisk 1.8 on a 32-bit Debian machine, and it has been fine for some time now. But! We've got such a incoming call volume over the few weeks that we'll have Asterisk occasionally restart itself. My hunch is that it is in part memory pressure. I can't add RAM and have it help, because it's 32-bit. I intend to move to a 64-bit machine, but I was hoping to

We are having the strangest issue that I have seen for some time. A customer of ours with Polycom phones (4x ip335, 2x ip550) will occasionally (maybe 1 in 50 calls) hear ringing on the line along with the other party. It has happened on both incoming and outgoing calls across apparently all of the phones. We use ip550 in our office with Asterisk and have never had such a problem (we run the same

Does Asterisk 1.0.2 support 302 redirects? With 1.0.1 I get: Got SIP response 302 "Moved Temporarily" When forwarding the call to other SIP server. This is a "bug": http://lists.digium.com/pipermail/asterisk-users/2004-May/045774.html --- Jan Baggen - jbaggen@ip2.nl IP2 Internet BV / http://www.ip2.nl

Hello, I 'm looking for a way to pass the '302 moved temporarily' received from the SIP device back to the SIP provider. Here is the setup: Some SIP phones are connected to an Asterisk System version 1.8. External connection to the public network is also done via SIP to a VoIP provider. Phone A has a CFW all calls to a phone number in public network (Mobile Phone) incoming call to

I have some Cisco 7905 phones with the SIP load 1.02.00(040406A). When the phone is off-hook but no call has been placed, or when the Do Not Disturb is activated, the phone returns a 302 "Moved Temporarily" message back to asterisk as follows: ----------- -- Executing Dial("SIP/5060-0811bb00", "SIP/9871234|20|Ttr") in new stack -- Called 9871234 -- Got SIP response

Hi folks, this does look like a bug to me: Asterisk replaces the @63.214.186.6 by @context which obviously leads to a failure. Any comments, do I have a configuration issue on my side that I missed? Cheers, Philipp -- Executing Dial("SIP/philipp-bd5f", "SIP/992365264680@nikotel- out|90") in new stack -- Called 99xxxxxxxxxx@nikotel-out -- Got SIP response 302

Hi, With a 1.4.35 or 1.6.1.19, I'm facing this behaviour : - extension 7002 is a SIP hard phone currently configured to forward incoming calls to extension 7003, when a call is unanswered within a 10s time frame - when extension 7001 is calling extension 7002 with a Dial(SIP/7002,20) statement and no one answers, then : - after 10s, Asterisk receives "SIP 302 Moved temporarily"

Hello when setting a local forward (in this case to extension 23) on a SIP phone, I see the following on the Asterisk CLI : [Aug 31 14:59:34] -- Got SIP response 302 "Moved Temporarily" back from 11.22.33.44:40670 [Aug 31 14:59:34] -- Now forwarding Local/myaccount184 at CallFromQueue-000007f4;2 to 'Local/23 at from-internal' (thanks to SIP/myaccount184-00003729)

I encountered something strange. A local business has an ACD that, when I call it using a Polycom 550 connected through an Asterisk system, will respond to button presses only if they are short. Calling this business with our old (non-Asterisk) phone system or with my cell phone works because they only generates short tones when pressing the keypad during a call. The Polycom, however, generates

<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> </head> <body bgcolor="#ffffff" text="#000000"> Hello,<br> <br> I have an * installation that sometimes receives&nbsp; a 302 "Moved Temporarily" response to an INVITE. * sends the ACK but takes about 30 seconds to start the new INVITE

I tried really quickly the other day to send an image to these phones from the dialplan like this: exten => 2821,n,SendImage(/var/lib/asterisk/images/asterisk-intro) or exten => 2821,n,SendImage(asterisk-intro) It didn't work for me. Should this work? Is anyone else using this with Polycom Phones? Bob

Hi, I have the following problem that when asterisk receives SIP response 302 it cannot forward the call I get such debug: [Jan 4 10:43:27] WARNING[18671]: channel.c:3281 ast_request: No channel type registered for 'Local' [Jan 4 10:43:27] NOTICE[18671]: app_dial.c:505 wait_for_answer: Unable to create local channel for call forward to 'Local/poczta at routing-sip' (cause = 66)

I'm trying to annotate a density plot and I'm using bquote to paste the sigma symbol next to the numeric text of the standard deviation calculation that I am performing. I have been able to successfully turn the sigma symbol and numeric output the color blue, but when I try to change the font of the text to bold, R doesn't seem to recognize the "font=" command in the same way

Dear all, In my Asterisk 1.2.17 architecture different levels of permissions are established using different contexts that hierarchically include more permissive contexts until default context is reached. In default context there are only local extensions, only in more restricted contexts there are the PSTN access. So, if some user dials some number, Asterisk looks which context that user

When a call is transferred to another extension using a blind transfer, asterisk keeps traces of who is transferring in the BLINDTRANSFER variable. If instead the call is "forwarded" using most phone call forward feature, a 302 Moved Temporarily is sent back to asterisk -- Called SIP/104-DEVEL -- Got SIP response 302 "Moved Temporarily" back from 83.211.***.***:5063

Justin Sherrill wrote: > I would love to run Asterisk on a BSD system. I do not know of any > developers actively working on Asterisk on a BSD platform, though my > knowledge isn't comprehensive. I'm also unaware of anyone developing on BSD like that. Linux of course and a smattering of folks on OSX doing the odd thing. Cheers, -- Joshua Colp Digium, Inc. | Senior Software

Gday all, Tried creating a new rails app with 3.2.0 on MRI 1.9.3 running under RVM 1.10.2, and it crashes on bundle with the following trace: brentona@brails:~/rails_3.2.0$ bundle Fetching source index for https://rubygems.org/ /home/brentona/.rvm/rubies/ruby-1.9.3-p0/lib/ruby/1.9.1/net/http.rb: 2632:in `error!'': 302 "Moved Temporarily" (Net::HTTPRetriableError) from

Paul, You solution worked out really well when I ran my code in R. However, when I try to turn the plot into a pdf, the unicode string no longer seems to function and instead of the sigma symbol there are just two periods (See example code below). The following is the code working in the R environment just like I want it to look: set.seed(1) Data=rnorm(100,sd=10000) plot(density(Data))