similar to: 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

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.

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

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 destination > sees the SIP packet as coming from

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 bindaddr in the

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 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

Hi Dovid, We can change the SDP in Kamailio, but Asterisk will still send its RTP from its default address. The remote end is strict about accepting RTP from the specified source and won't accept it. Have you any suggestions to solve that problem? Thank you. On Fri, 30 Oct 2020 at 14:49, Dovid Bender <dovid at telecurve.com> wrote: > Why not use OpenSips/Kamailoo in between?

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

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

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

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

Hello, We have a system that interoperates with an external service, so that the basic call flow is: PSTN origination -> Asterisk A -> External service -> Asterisk B Initially the SDP from the external service tells the two Asterisks to send RTP directly to each other. Part way through the call the external service sends re-INVITEs both Asterisks to change the address for audio to

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: #

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

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

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

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

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.

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

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