similar to: RTP address learning and timing problem

Displaying 20 results from an estimated 4000 matches similar to: "RTP address learning and timing problem"

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
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
2023 Apr 17
1
RTP address learning and timing problem
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(), rtp->rtp_source_learn.start)) { ast_verb(4, "%p -- Strict RTP learning complete - Locking on source address %s\n", Our call shows: #
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
2023 Apr 17
1
RTP address learning and timing problem
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 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
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
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.
2023 Feb 24
1
Big problems after update to 9.6
Hi David, It seems like a network issue to me, As it's unable to connect the other node and getting timeout. Few things you can check- * Check the /etc/hosts file on both the servers and make sure it has the correct IP of the other node. * Are you binding gluster on any specific IP, which is changed after your update. * Check if you can access port 24007 from the other host. If
2020 Oct 22
2
Multiple IP addresses and using same IP for outbound calls as inbound
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 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
2018 Jul 09
6
How to steal an answered call?
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
2020 Oct 22
2
Multiple IP addresses and using same IP for outbound calls as inbound
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
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 being generated towards
2019 Jun 06
2
Find out which key ended recording?
Hello, We have a need to record audio and allow the user to press any DTMF key to end the recording. Currently we're using the AGI command "record file" which does allow us to specify which DTMF keys can end the recording. However we also need to know *which* key actually ended the recording. Note that only allowing # or * to end the recording won't work for us. Does anyone
2019 Jun 07
4
Find out which key ended recording?
Hi Steve, What language is that please? We're using Perl and so far I haven't found an equivalent there. Thanks for your help. On Fri, 7 Jun 2019 at 12:10, Steve Edwards <asterisk.org at sedwards.com> wrote: > On Fri, 7 Jun 2019, David Cunningham wrote: > > > We have a need to record audio and allow the user to press any DTMF key > > to end the recording.
2019 Dec 28
1
GFS performance under heavy traffic
Hi David, It seems that I have misread your quorum options, so just ignore that from my previous e-mail. Best Regards, Strahil NikolovOn Dec 27, 2019 15:38, Strahil <hunter86_bg at yahoo.com> wrote: > > Hi David, > > Gluster supports live rolling upgrade, so there is no need to redeploy at all - but the migration notes should be checked as some features must be disabled first.
2023 Feb 23
1
Big problems after update to 9.6
Hello, We have a cluster with two nodes, "sg" and "br", which were running GlusterFS 9.1, installed via the Ubuntu package manager. We updated the Ubuntu packages on "sg" to version 9.6, and now have big problems. The "br" node is still on version 9.1. Running "gluster volume status" on either host gives "Error : Request timed out". On
2019 Dec 20
1
GFS performance under heavy traffic
Hi David, Also consider using the mount option to specify backup server via 'backupvolfile-server=server2:server3' (you can define more but I don't thing replica volumes greater that 3 are usefull (maybe in some special cases). In such way, when the primary is lost, your client can reach a backup one without disruption. P.S.: Client may 'hang' - if the primary server got
2015 Mar 12
2
WebRTC demo phones
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
2013 Jan 03
3
faxdetect on/off on the fly?
Hello, We want the ability to choose from an AGI script whether or not to enable faxdetect for calls over SIP or DAHDI. Is this possible, or can anyone suggest a workaround? Thanks for any advice. -- 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
2015 Aug 18
2
Asterisk 13 chan_sip trunk appending @string to dialled number
Yes indeed. Do you have the dialplan, eg from /etc/asterisk/extensions.conf? Something is getting this OUT_3_SUFFIX variable and including it in a Dial to 172.22.4.12. On 18 August 2015 at 16:21, Brendan Ord <bord at staff.onthenet.com.au> wrote: > Starting to make sense when I saw this line: > > > > [2015-08-18 15:01:33] DEBUG[19366][C-00001cfc]: pbx.c:3785 >