similar to: Meetme echo cancellation

Hi there I am using an IAX2 softphone built from the IaxClient library dialing into Meetme conferences. The IaxClient seems to use silence suppression, and not sure if this can be disabled. The client works fine through most routers, but for some it disconnects the client after about 5 minutes and gives this error in the asterisk logs: Chan_iax2.c:1480 attempt_transmit: Max retries exceeded to

Hi there I am using IAX2 softphones dialing into meetme conferences. I also have jitterbuffer=yes, with typical jitterbuffer settings. The problem I am having is that as soon as there is a delay from a participant, then the delay continues until the participant hangs up and dials in again. When dialing in again the delay seems to go. It seems to me as though as soon as the server registers

Hi there I am moving into a production environment. I will mostly be using Meetme, with Ztdummy for timing. I have a question on which of 2 processor setups is favourable. I have the choice between Dual 2.8GHz Xeon Processors and a single Pentium 4 3.06GHz Processor. These will cost me exactly the same amount. Would one of these processor setups be favourable, both in terms of performance and

Hi there I have a question regarding IAX jitter. I have 3 users on a LAN dialing into a Meetme conference on an Asterisk box which is also hosted on the LAN. I have set jitterbuffer = no and tos = lowdelay. Now, for 2 of the users the audio is fine, but for the 3rd user there is intermittent break up in the audio when they are receiving. I have had a look at "iax2 show channels" and

Hi there I have been using Asterisk Meetme with Ztdummy for timing. It seems to work fine and I haven't had any major problems. I am now moving into a production environment and am wondering if it is better to use a Zaptel card? Are there any problems with Ztdummy? I will probably have around 30 concurrent users in different conferences. Many thanks Steven -------------- next part

Hi I have a routing problem with the OpenVPN service running directly on the firewall itself. I have two DSL connections, one with a static IP (and my default route), the other with a dynamic IP. The first is called ISBD in the configs, the second is called SAIX. Connecting to the OpenVPN via ISBD works well, the packets route perfectly. Connecting via SAIX does not. In the attached status.txt,

This is the SMTP Server program at host orange.fr. I''m sorry to have to inform you that your message could not be be delivered to one or more recipients. It''s attached below. For further assistance, please send mail to <postmaster> If you do so, please include this problem report. You can delete your own text from the attached returned message. The SMTP Server

Hello all, Please forgive me if this sounds a little (or a lot) ignorant as I am very new to asterisk. Right now I have two pc's connected back to back through an E100 card running asterisk. I have openh323 running as well and I am able to route calls through the E1 line. Next up I would like to be able to register asterisk with a gatekeeper. On another computer is running openGK. Using

On Mon, Jun 12, 2017 at 7:30 PM, Langley, Robert <Robert.Langley at ventura.org > wrote: > As far as the peer status (and I now remember seeing this earlier) the > issue appears to be that the host name for gsaov07 is attempting to resolve > over the wrong network for gluster "ent...." and not "stor.local". > So, it may be as simple as removing gsaov07 as a

As far as the peer status (and I now remember seeing this earlier) the issue appears to be that the host name for gsaov07 is attempting to resolve over the wrong network for gluster "ent...." and not "stor.local". So, it may be as simple as removing gsaov07 as a peer, then probing over the correct network. I'll follow up with the Engine log. Sent using OWA for iPhone

On Mon, Jun 12, 2017 at 6:41 PM, Atin Mukherjee <amukherj at redhat.com> wrote: > > On Mon, 12 Jun 2017 at 17:40, Langley, Robert <Robert.Langley at ventura.org> > wrote: > >> Thank you for your response. There has been no change of IP addresses. >> And I have tried restarting the glusterd service multiple times. >> I am using fully qualified names with a

Thank you for your response. There has been no change of IP addresses. And I have tried restarting the glusterd service multiple times. I am using fully qualified names with a DNS server that has forward and reverse setup. Something I had noticed is that, with oVirt, communication is being attempted over the VM network, rather than storage network. Which is not how the bricks are defined. Not sure

glusterd failed to resolve addresses of the bricks. This could happen either because of glusterd was trying to resolve the address of the brick before the N/W interface was up or a change in the IP? If it's the latter then a point to note is that it's recommended that cluster should be formed using hostname otherwise if the server goes through IP change we have to change the configuration

Is anyone using the subject phone without the proprietary call setup equipment? How do you configure this phone to use the * pbx? I am hoping to go out and buy a couple of phones but the dealer here says I need to spend another $15K or so for the call manager equipment. Regards, Sean Langley, P.Eng Firmware Engineer General Dynamics Canada (403)730-1482 sean.langley@gdcanada.com

This is in relation to bringing up a new oVirt environment. When I was bringing up oVirt (Hosted-Engine), I had the Glusterd service stopped, since the server (I have its name as GSAoV07) was initially only going to be used to host the oVirt engine as a hypervisor (it will have Gluster volumes in the near future). I have three other servers I am using for storage. One of those three is also going

On Mon, 12 Jun 2017 at 17:40, Langley, Robert <Robert.Langley at ventura.org> wrote: > Thank you for your response. There has been no change of IP addresses. And > I have tried restarting the glusterd service multiple times. > I am using fully qualified names with a DNS server that has forward and > reverse setup. > Something I had noticed is that, with oVirt, communication

Hi, I am implementing and audio recorder where I need to encode to vorbis DURING recording. I need to allow the user to cut and paste segments of encoded audio into a new stream. As such, I need to encode the audio in sub-second pages (the smallest allowable amount for the wave format), so that cut/paste can be implemented simply by re-arranging these encoded pages. How can I calculate, the

Use oh323. Download the openh323 and pwlib tarballs from openh323.org Follow Jeremy's instructions in the /asterisk/channels/h323/ directory EXACTLY! good luck Regards, Sean Langley, P.Eng Firmware Engineer General Dynamics Canada (403)730-1482 sean.langley@gdcanada.com > -----Original Message----- > From: Senad Jordanovic [mailto:senad@cwcom.net] > Sent: Friday, September 12,

Hello All, Eric Langley here with my first post to this list. I am looking to determine if R is suitable for a development project I am working on and if so possibly finding someone proficient in R that would be interested in doing the coding. I would like to preface my inquiry that while I am not a programmer I can communicate in a dialog my objectives. An array of rank ordered data looks like

Hello I am trying to read a large number of WinZip files from inside R (v 1.7.1, options()$unzip = internal). I have unsuccessfully tried to unpack the files and write to an external directory so that I could read the text files using zip.unpack. Reading some of the archives suggests that this is for .ZIP format files only. I have also unsuccessfully tried to use gzfile to open a connection and