search for: dcunningham

Why not use OpenSips/Kamailoo in between? Where you want 1.1.1.1 you pass it along as is. Where you want 2.2.2.2 change the sdp in opensips/kamailio On Thu, Oct 29, 2020 at 20:44 David Cunningham <dcunningham at voisonics.com> wrote: > Hello, > > Does anyone know a way with chan_sip to tell Asterisk to use a specific IP > address for its end of the communication for a specific device? Something > like: > > [device] > type = friend > host = 11.22.11.22 > ouraddress = 33....

OK, thank you George. On Sat, 24 Oct 2020 at 03:16, George Joseph <gjoseph at digium.com> wrote: > > > On Thu, Oct 22, 2020 at 4:13 PM David Cunningham < > dcunningham at voisonics.com> wrote: > >> Hi George, >> >> Thank you for the response. I'm a little unclear on what you mean by a >> transport. We're using chan_sip, not pjsip. >> >> Do you mean a device in sip.conf, using bindaddr to set the address to >&gt...

I don't know in that specific output what happened. Your best course of action is to add further logging or step through the logic with all of the knowledge you have of the RTP streams to understand what is happening. On Mon, Apr 17, 2023 at 8:52 PM David Cunningham <dcunningham at voisonics.com> wrote: > Hi Joshua, > > Thank you for that. From the code it kind of looks like > STRICT_RTP_LEARN_TIMEOUT is a minimum, not a maximum: > > if (!ast_sockaddr_isnull(&rtp->strict_rtp_address) > && STRICT_RTP_LEARN_TIMEOUT < ast_tvdiff_ms(a...

...r related process left, then start gluster on br first and check the `gluster peer status` then start gluster on sg. ( If you can take the downtime ? ) Thanks, Anant ________________________________ From: Gluster-users <gluster-users-bounces at gluster.org> on behalf of David Cunningham <dcunningham at voisonics.com> Sent: 23 February 2023 9:56 PM To: gluster-users <gluster-users at gluster.org> Subject: Re: [Gluster-users] Big problems after update to 9.6 EXTERNAL: Do not click links or open attachments if you do not recognize the sender. Is it possible that version 9.1 and 9.6 ca...

...eout before FUSE addresses the next server. There is a special script for killing gluster processes in '/usr/share/gluster/scripts' which can be used for setting up a systemd service to do that for you on shutdown. Best Regards, Strahil NikolovOn Dec 20, 2019 23:49, David Cunningham <dcunningham at voisonics.com> wrote: > > Hi Stahil, > > Ah, that is an important point. One of the nodes is not accessible from the client, and we assumed that it only needed to reach the GFS node that was mounted so didn't think anything of it. > > We will try making all nodes accessi...

Hello, I'm familiar with Pickup/PickupChan for taking a ringing call, but does anyone know how a phone can "steal" an already answered call from another phone? Our users have decided that call parking is too long-winded and don't want to use that. For example: phone A calls phone B, phone B answers the call, phone C dials something to "steal" the call from B, and

...a.com> wrote: > It's probably best if you read the logic[1]. There's an entire comment > that talks about how it works. > > [1] > https://github.com/asterisk/asterisk/blob/20/res/res_rtp_asterisk.c#L8158 > > On Mon, Apr 17, 2023 at 7:10 PM David Cunningham < > dcunningham at voisonics.com> wrote: > >> Hi Joshua, >> >> Could you confirm if the 5 second period for learning a new audio stream >> is a minimum or a maximum? The unusual call flow in question results in >> Asterisk learning a new audio stream when we don't want it to...

...Fri, 30 Oct 2020 at 14:49, Dovid Bender <dovid at telecurve.com> wrote: > Why not use OpenSips/Kamailoo in between? Where you want 1.1.1.1 you pass > it along as is. Where you want 2.2.2.2 change the sdp in opensips/kamailio > > On Thu, Oct 29, 2020 at 20:44 David Cunningham <dcunningham at voisonics.com> > wrote: > >> Hello, >> >> Does anyone know a way with chan_sip to tell Asterisk to use a specific >> IP address for its end of the communication for a specific device? >> Something like: >> >> [device] >> type = friend >...

It's probably best if you read the logic[1]. There's an entire comment that talks about how it works. [1] https://github.com/asterisk/asterisk/blob/20/res/res_rtp_asterisk.c#L8158 On Mon, Apr 17, 2023 at 7:10 PM David Cunningham <dcunningham at voisonics.com> wrote: > Hi Joshua, > > Could you confirm if the 5 second period for learning a new audio stream > is a minimum or a maximum? The unusual call flow in question results in > Asterisk learning a new audio stream when we don't want it to, and having a > mini...

On Tue, Feb 28, 2023 at 9:51 AM Joshua C. Colp <jcolp at sangoma.com> wrote: > On Tue, Feb 28, 2023 at 9:50 AM David Cunningham < > dcunningham at voisonics.com> wrote: > >> Hello, >> >> Does anyone know if one of the "strictrtp" options disables RTP learning? >> As far as I can tell from the documentation the values "no" and "seqno" are >> more permissive in allowing other...

Hi David, On Dec 24, 2019 02:47, David Cunningham <dcunningham at voisonics.com> wrote: > > Hello, > > In testing we found that actually the GFS client having access to all 3 nodes made no difference to performance. Perhaps that's because the 3rd node that wasn't accessible from the client before was the arbiter node? It makes sense, as...

...seconds of audio would help avoid this. Thank you! On Thu, 2 Mar 2023 at 12:32, Joshua C. Colp <jcolp at sangoma.com> wrote: > On Tue, Feb 28, 2023 at 9:51 AM Joshua C. Colp <jcolp at sangoma.com> wrote: > >> On Tue, Feb 28, 2023 at 9:50 AM David Cunningham < >> dcunningham at voisonics.com> wrote: >> >>> Hello, >>> >>> Does anyone know if one of the "strictrtp" options disables RTP >>> learning? As far as I can tell from the documentation the values "no" and >>> "seqno" are more permi...

...bind for that device? We've only used bindaddr in the [general] section before, but if it will work in a device that could be the answer. On Fri, 23 Oct 2020 at 00:13, George Joseph <gjoseph at digium.com> wrote: > > > On Wed, Oct 21, 2020 at 9:16 PM David Cunningham < > dcunningham at voisonics.com> wrote: > >> Hello, >> >> We have an Asterisk server with two public IP addresses, let's say >> 1.1.1.1 and 2.2.2.2. Normally calls come in to 1.1.1.1 and are bridged with >> a call dialled from Asterisk to an external destination. The extern...

...ss = 33.44.33.44 This is for use on a server with multiple IP addresses. There is the "extenip" setting, but it's really designed for NAT, and can only appear in the [general] section. Any suggestions would be greatly appreciated. On Sat, 24 Oct 2020 at 09:43, David Cunningham <dcunningham at voisonics.com> wrote: > OK, thank you George. > > > On Sat, 24 Oct 2020 at 03:16, George Joseph <gjoseph at digium.com> wrote: > >> >> >> On Thu, Oct 22, 2020 at 4:13 PM David Cunningham < >> dcunningham at voisonics.com> wrote: >> &gt...

...<Original Volume>?? group custom-group' ,? where? the file is located on every gluster? server in the '/var/lib/gluster/groups' directory. > Last ,? get rid of the sample volume. > > Best Regards, > Strahil Nikolov > > On Dec 27, 2019 03:22, David Cunningham <dcunningham at voisonics.com> wrote: >> >> Hi Strahil, >> >> Our volume options are as below. Thanks for the suggestion to upgrade to version 6 or 7. We could do that be simply removing the current installation and installing the new one (since it's not live right now). We might...

Hello, Can anyone recommend a particular online WebRTC phone for testing with Asterisk? We tried: - JsSIP, but even with the "enable video" checkbox disabled it sends video options in the INVITE SDP and Asterisk rejects it with "Rejecting secure video stream without encryption details". - sipML5, but it won't register, perhaps something to do with not using the Asterisk

...y via 'gluster volume set <Original Volume> group custom-group' , where the file is located on every gluster server in the '/var/lib/gluster/groups' directory. Last , get rid of the sample volume. Best Regards, Strahil NikolovOn Dec 27, 2019 03:22, David Cunningham <dcunningham at voisonics.com> wrote: > > Hi Strahil, > > Our volume options are as below. Thanks for the suggestion to upgrade to version 6 or 7. We could do that be simply removing the current installation and installing the new one (since it's not live right now). We might have to convince...

On Thu, Oct 22, 2020 at 4:13 PM David Cunningham <dcunningham at voisonics.com> wrote: > Hi George, > > Thank you for the response. I'm a little unclear on what you mean by a > transport. We're using chan_sip, not pjsip. > > Do you mean a device in sip.conf, using bindaddr to set the address to > bind for that device? We'v...

Hello, We have an Asterisk server with two public IP addresses, let's say 1.1.1.1 and 2.2.2.2. Normally calls come in to 1.1.1.1 and are bridged with a call dialled from Asterisk to an external destination. The external destination sees the SIP packet as coming from 1.1.1.1 and the media address in the SDP is 1.1.1.1, which is great. However if we receive a call in to 2.2.2.2 then the call

Sipml5 works. You need to have TLS enabled on asterisk web socket. Mitul On 12-Mar-2015 12:47 PM, "David Cunningham" <dcunningham at voisonics.com> wrote: > Hello, > > Can anyone recommend a particular online WebRTC phone for testing with > Asterisk? > > We tried: > > - JsSIP, but even with the "enable video" checkbox disabled it sends video > options in the INVITE SDP and Asterisk rej...