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2020 Oct 30
3
Multiple IP addresses and using same IP for outbound calls as inbound
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 &...
2020 Oct 23
2
Multiple IP addresses and using same IP for outbound calls as inbound
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...
2023 Feb 24
1
Big problems after update to 9.6
...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 ta...
2018 Jul 09
6
How to steal an answered call?
...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 Searching on voip-info.org shows a "BristuffSteal" command but it's very out of date (Asterisk 1.2).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>
2011 Nov 21
2
Continue AGI after Dial() following caller hang up?
..., 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>
2020 Oct 30
0
Multiple IP addresses and using same IP for outbound calls as inbound
...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...
2020 Oct 22
2
Multiple IP addresses and using same IP for outbound calls as inbound
...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 >&gt...
2020 Oct 30
0
Multiple IP addresses and using same IP for outbound calls as inbound
...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: >> >>>...
2023 Apr 18
1
RTP address learning and timing problem
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()...
2023 Apr 17
1
RTP address learning and timing problem
...: > 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 havi...
2015 Mar 12
2
WebRTC demo phones
...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>
2023 Apr 17
1
RTP address learning and timing problem
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...
2023 Feb 22
1
RTP address learning and timing problem
...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>
2020 Oct 22
2
Multiple IP addresses and using same IP for outbound calls as inbound
...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>
2023 Mar 01
2
RTP address learning and timing problem
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 ra...
2023 Apr 17
1
RTP address learning and timing problem
...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 a...
2019 Dec 24
1
GFS performance under heavy traffic
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...
2011 May 12
1
Higher CPU usage on 1.6.1 than 1.4?
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>
2011 Jan 20
1
Introducing easySysAdmin - automated security and telecom fraud protection
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 interes...
2020 Oct 23
0
Multiple IP addresses and using same IP for outbound calls as inbound
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 use...