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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 = 33.44.33.44 >...

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 >> bind for th...

...ss 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 can't talk...

...s the call, phone C dials something to "steal" the call from B, and finally A and C are talking. Searching on voip-info.org shows a "BristuffSteal" command but it's very out of date (Asterisk 1.2). Thanks in advance for any suggestions. Kind regards, -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20180709/495bd0c1/attachment.html>

..., and not using a new AGI from the 'h' extension. It works fine when the called party hangs up and the 'g' option is used, but not for caller hangup. Is this possible? If not a confirmation that this is the case would be very helpful. Thanks for any advice! -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111121/476e9edf/attachment.htm>

...20 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 >> host = 1...

...evice? 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 external >> d...

...4 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: >> >>> Hi...

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(ast_tvnow(), &...

...: > 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...

...sk Websocket server (which I don't see an option to choose) - Janus, but the INVITE SDP contains "RTP/AVP" not "RTP/SAVP, and Asterisk rejects it with "We are requesting SRTP for audio, but they responded without it!" Thanks for any suggestions. -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150312/2a15c775/attachment.html>

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 > minimum of say 2...

...g RTP from it. Note we have "canreinvite = no" in sip.conf, but I don't think that's relevant to the problem. Can anyone suggest how to prevent this problem? Is it possible to turn off learning the media address per call or per peer? Thanks for your help. -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20230223/b77b7173/attachment.html>

...at termination.com -> INVITE sent from 1.1.1.1:5060 to termination.com INVITE received to 2.2.2.2:5060 -> Asterisk dials destination at pstn.com -> INVITE sent from 2.2.2.2:5060 to pstn.com Does anyone know how this can be achieved? Thanks in advance for your help, -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20201022/61ad95c5/attachment.html>

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 sources rathe...

...o 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 permissive in allo...

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 no data is be...

Hello, We have a customer who upgraded from Asterisk 1.4 to 1.6.1.22 and is now experiencing higher CPU utilization on their server. I can't see anything wrong, so is this just expected with 1.6? Can anyone help explain it? Thanks for any advice. -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110512/99300b2d/attachment.htm>

Hello all, Voisonics is pleased to introduce easySysAdmin, an automated support/security platform, designed to save your engineer's time and prevent hacking attempts and telecom fraud. It comprises of an online service run by us, and a lightweight and easy-to-install client on your side. Specifically of interest t...

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've only used b...