similar to: Polycom IP335 3.3.1 Call Waiting

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.. > >

I am posting here since you guys are my last hope. I am trying to configure a Polycom Soundpoint IP 335 with MWI. Is there any way to eliminate the scrolling messages and Msgs softkey? I am trying to get it where it's just the light that indicates the new messages. I don't know if Asterisk has to send a different notification or what have you. Thanks, --Eric -------------- next

Sorry - that should be sysctl -w net.ipv6.conf.all.accept_ra=0 to disable that, not 1. Chris On Sun, Mar 8, 2015 at 11:14 PM, Chris Stone <axisml at gmail.com> wrote: > Try: > > sysctl -w net.ipv6.conf.all.accept_ra=1 > > to persist between boots, be sure to add this to your /etc/sysctl.conf > file. > > This should prevent the box from listening to any RA

On Fri, Mar 6, 2015 at 11:52 AM, Robert Moskowitz <rgm at htt-consult.com> wrote: > > > On 03/06/2015 11:00 AM, Robert Moskowitz wrote: > >> >> >> On 03/06/2015 10:55 AM, Barry Brimer wrote: >> >>> >>> >>> IPV6INIT="no" >>>> >>>> But I am still getting a global IPv6 (and of course local scope).

I am having a problem registering my cisco phones which is exactly like that described in http://lists.digium.com/pipermail/asterisk-users/2011-May/262306.html except that I am on Asterisk 1.8.3.3 and using sip level POS3-07-4-00 The symptoms are: o 7960 lines show [X] o Outbound calls can be made from the phone, including call pickup of inbound calls, but not to it. o Trace shows REGISTER

I have Polycom phones that support the g722 codec. Adding allow=g722 to the [general] section of sip.conf works great and I can make calls between the phones using g722. However Asterisk is negotiating g722 for calls going out my voip provider and transcoding these to ulaw. In sip.conf for the provider I have deny=all and allow=ulaw. This can cause potential audio degrading and wastes cpu cycles.

Hello all, I have a system running AsteriskNOW with asterisk asterisk-1.8.8.1-1_centos5 from AsteriskNOW repository. I just changed our Polycom 335 sip.conf so that blindpreferred=1 (all transfers default as blind transfers). If a customer calls in & we answer & transfer, everything works fine. But if we call out to a customer & then transfer to another internal extension, that

Let's say I have two devices configured and the follow call scenarios occur. [100] disallow=all allow=g722&ulaw Polycom phone with g722,ulaw,alaw,g729 [101] disallow=all allow=ulaw Polycom phone with g722,ulaw,alaw,g729 101 dials 100 -> ulaw to ulaw is chosen 100 dials 101 -> g722 to ulaw is chosen Ideally when 100 dials 101 ulaw would be chosen since it is the common format.

With previous Asterisk versions when running asterisk -r a welcome message is displayed with the version. I just upgraded to 1.8 and noticed it is not appearing. All I get is Verbosity is at least 3 and the console prompt. I looked at main/asterisk.c and still see the welcome message code. Any idea why it is not being shown? Ryan

Has there been a change in the way certified Asterisk is being packaged? Starting with certified Asterisk 11.6 has all the extended options are checked by default in menuslect? Certified Asterisk 11.2 does not have them checked and neither does certified Asterisk 1.8.15? Thanks, Ryan -------------- next part -------------- An HTML attachment was scrubbed... URL:

I saw the following lines in the log this morning. From my router logs I see that the connection went down as my ISP was doing maintenance for a few minutes last night. I can understand the external registrations timing out, but why do the phones become unreachable. They are on the internal lan within the same subnet as the Asterisk server. Internal DHCP and DNS was functional. If I had a PRI card

How about, in your /etc/sysconfig/network file adding or editing the line for IPV6 to be: NETWORKING_IPV6=no and then try a 'service network restart' and see what you get. Chris On Mon, Mar 9, 2015 at 11:52 AM, Robert Moskowitz <rgm at htt-consult.com> wrote: > No change after running this and trying both: > > system network restart > > ifdown eth0; ifup eth0

I haven't been successful in getting this to work. The issue looks to be that Asterisk is wanting peer authentication for the invite request as it sends back 401 Unauthorized. I am using FreePBX 2.7 and have tested both Asterisk 1.6.1.18 and 1.6.2.9. My trunk settings are type=peer transport=tcp qualify=yes insecure=port,invite host=10.10.1.31 context=from-internal Here is snippets of the

What the minimum Server Specifications do I need to run 200 concurrent channels at the time with .WAV recording (MixMonitor)? It will be connected via VOIP sip account. Codec will be ulaw. Which UK dedicated server provider do you recommend and how much bandwidth do I need? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

When you perform an attended transfer, the extension of the person transferring is displayed to the co-worker. Can I override the caller ID to display the caller's callerID during a blind transfer? Thanks, --E -------------- next part -------------- An HTML attachment was scrubbed... URL:

Lukasz Sokol wrote: > but have you considered a web-managed config-builder such as FreePBX? > Instead of building your dialplan from scratch ? I've never used FreePBX, but, after having looked at its website, I think I have a general understanding of what it can do. What I don't understand is how FreePBX answers my question about the Linksys SPA3102 being good for a mission

Hi, My phones stopped auto-answering when being paged, since I moved on to Polycom firmware 3.3.0 (3.3.1 is the same, I tried). That is with Asterisk 1.6.2.16. I looked at the wiki but nothing I try there works, even if I cut and paste the same setup. Any one has any idea of what I should change from my old 3.2.3 setup? My older phone (501) still using 3.1.6 still auto-answer

Hi, I''m a new puppet user (thanks !) and I''m just looking at moving my homegrown manifest structure into something resembling the best practise guide at : http://www.reductivelabs.com/trac/puppet/wiki/PuppetBestPractice. It seems as if the structure here doesn''t map to the default puppet layout (and thus doesn''t work with the standard puppet config

I have a Polycom Soundpoint IP335. There are no inbound routes set to the phones yet. However, the phones are getting phantom rings. What is the legitimacy of these calls? Is there something I need to block to stop it? I believe its people trying to hack the phones/phone system but I cannot find where I read that before. Thanks, --E -------------- next part

Hello, I tried this week to upgrade my samba 3.2.4 (2 PDCs one trusting the other) to samba 3.3.0 then samba 3.3.1, and apart from the problem with winbindd and trusted domain, my users are not able to modify any ms word document (excel does the same). You can open the file correctly, modify it, and when saving it, it pops up "Access denied" If you try to save the file in the same