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...gt; > 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 announcements. > > > Chris > > On Sun, Mar 8, 2015 at 10:55 PM, Ryan Wagoner <rswagoner at gmail.com> wrote: > >> 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: >> > >> >> >> >> >> >&g...

...um.com > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Steve Davies > Sent: Tuesday, June 29, 2010 7:51 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Codec negotiation > > On 26 June 2010 22:08, Ryan Wagoner <rswagoner at gmail.com> wrote: > > 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...

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

...rsist between boots, be sure to add this to your /etc/sysctl.conf >>> file. >>> >>> This should prevent the box from listening to any RA announcements. >>> >>> >>> Chris >>> >>> On Sun, Mar 8, 2015 at 10:55 PM, Ryan Wagoner <rswagoner at gmail.com> >>> wrote: >>> >>> 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: >>>>> >&g...

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).

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 announcements. Chris On Sun, Mar 8, 2015 at 10:55 PM, Ryan Wagoner <rswagoner at gmail.com> wrote: > 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 wrot...

...t; > 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 announcements. > > > Chris > > On Sun, Mar 8, 2015 at 10:55 PM, Ryan Wagoner <rswagoner at gmail.com> wrote: > >> 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: >> > >> >> >> >> >&...

...ept_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 announcements. >> >> >> Chris >> >> On Sun, Mar 8, 2015 at 10:55 PM, Ryan Wagoner <rswagoner at gmail.com> wrote: >> >>> 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: >>>> >>>>> >>>>&gt...

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.

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

I updated my phones to the UCS 3.3.1 firmware a few months back. The scenario is I place a call and receive an incoming call. With 3.3.1 the screen will show call 1/2 and I have to press the down arrow to see the caller name / number. Has anybody else noticed this with 3.3.1? I had thought with 3.2.4 it would automatically show call waiting name and number without pressing any keys. It could be

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

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.. > >

Hey Guys! We have two Asterisk with A102D Sangoma cards now i want to connect them back-to-back over PRI line via Cross-cable so what would be the configuration specially timing source and all? anybody did it before like this ? I want to make sure everything before putting in production.. (saving my downtime) -S -------------- next part -------------- An HTML attachment was

...ept_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 announcements. >> >> >> Chris >> >> On Sun, Mar 8, 2015 at 10:55 PM, Ryan Wagoner <rswagoner at gmail.com> wrote: >> >>> 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: >>>> >>>>> >&g...

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

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.

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: