similar to: Huge ten second audio delay on SIP channel

I have had the same problem when calling across Asterisk from Diax to a SIP phone. If Asterisk "Answers" the call before the "Dial" to the SIP phone there is no delay. Otherwise there is a 10-20 second delay in the Voice path! Peter -----Original Message----- From: Dan [mailto:danto@rdslink.ro] Sent: 14 January 2005 15:57 To: Denis Galv?o - iSolve; Asterisk Users Mailing

On Thu, Jul 20, 2023 at 1:49?PM Johnnie W Adams <jxadams at ualr.edu> wrote: > > Hi, folks, > > We're experiencing an odd ten-second delay intermittently when logging > into any of our Linux boxes which authenticate against LDAP. Here's where > it happens: > > Jul 13 11:54:23 console2 sshd[1853]: debug1: temporarily_use_uid: <my > uid\gid>

Hi, folks, We're experiencing an odd ten-second delay intermittently when logging into any of our Linux boxes which authenticate against LDAP. Here's where it happens: Jul 13 11:54:23 console2 sshd[1853]: debug1: temporarily_use_uid: <my uid\gid> (e=0/0) Jul 13 11:54:35 console2 sshd[1853]: debug1: trying public key file <my key file> My assumption is there's

Nico Kadel-Garcia wrote: > On Thu, Jul 20, 2023 at 1:49?PM Johnnie W Adams <jxadams at ualr.edu> wrote: >> >> Hi, folks, >> >> We're experiencing an odd ten-second delay intermittently when logging >> into any of our Linux boxes which authenticate against LDAP. Here's where >> it happens: >> >> Jul 13 11:54:23 console2

Has anyone seen this problem before? I have a server with a single X100P card. The audio level is a low, but if I raise the gain to more than -2db (Rx + Tx) it starts to oscillate in an echo test. Not at a high frequency but with a noise that is best described as a steam engine starting up. It then starts to clip and crackle. If I bring the gain down to Rx=-2.0 and Tx=0.0 or lower then it settles

Did anybody tried MozIAX extension? It is Mozilla IAX2 soft-phone. http://moziax.mozdev.org/ I tried it yesterday on eee pc, connected to asterisk on local LAN and the performance is terrible! The delay is about 2sec or 3sec. and very bad echo. I think it is the implementation of their IAX2 in their add on, as I have tried external mic. and the same delay problem. As a comparison I've tried

Hi all! Somebody knows something to do with a high delay using Asterisk + DIAX!? When I used IAXComm(Linux) in both sides(peer and me) no problems. Whan I used DIAX099f(WinXP) in both sides(peer and me) I have a delay in the voice coming from the person that I called. I don't have delay in my voice to the peer phone. CODEC: u-law (I tried with all available codecs) Thanks for your help!

Hello all! Has anyone expirienced the following:? With an IAXclient softphone (like diax/iaxcomm/etc) Dialing to the PSTN (zap) or a SIP device has no problems .. but when I make calls between 2 softphones I have weird problems.... in about 4 out of 10 IAX-2-IAX softphone calls I get a big delay .. in the beginning of the call it's all okay... (delay < 0.5 sec) but the longer the call

Hi All! I am facing some delay in conferencing. Using DIAX for Voip calls ,no hardware used yet I am using MeetMe to achieve conferencing and am having a lot of delays. Can anyone tell me how to reduce the delay Regards, Amna Saleem -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, Diax version 0.9.9f is ready to be tested by the interested people. You can download it for the moment from the following location only: http://www.geocities.com/tdanro/diax/diax099f.zip Please do not use older config files with 0.9.9f !!! You have some command line options now for diax.exe: /d - start with debug mode enabled /u - start with ATCOM USB phone support enabled

Hi All, I have installed Asterisk and able to create Users and get them connected to Asterisk after authentication. My question is how can I make calls to different DIAX clients through my Asterisk server. I also have vonage softphone account, using that I tried calling 18882255322 -- Registered 'manoj' (AUTHENTICATED) at 59.93.73.0:4569 -- Registered 'diax'

Building on my last message, I am trying to get CHANNEL data using getvar (through the AMI). And although I'm getting responses, some values returned seem illogical. For example, phone 111 calls phone 222 via the PBX. Here's the data I get back Channel A: "1688509741.112" , name: "PJSIP/111-00000064" , is originator: Y , call-Id: "u.l6kcou25cax60 at

Hello veryone, I have a strange problem. I have an asterisk (latest from CVS) with latest oh323 channel driver. I place calls with DIAX. The H323 gateways only support G711A De DIAX only supports GSM When I perform an inbound call: H323 -> asterisk -> DIAX :: sound is ok. When I perform an outbound call: DIAX -> Asterisk -> h323 :: sound is terrible and CPU load is 80% When I

Hello,everyone, I encoutered some difficult with IAX when I run the asterisk. <---->internet <--> asterisk + NAT <--> DIAX my * box and NAT are at the same linux box which connecting to the internet using ADSL. The box has two network cards and two IP address,such as public IP:211.11.11.11 private IP:192.168.1.10 the windows box running DIAX lies behind the

On Tue, Jul 4, 2023 at 7:52 PM TTT <lists at telium.io> wrote: > Building on my last message, I am trying to get CHANNEL data using getvar > (through the AMI). And although I'm getting responses, some values > returned seem illogical. For example, phone 111 calls phone 222 via the > PBX. Here's the data I get back > > > > > > Channel A:

Channel A: "1688509741.112" , name: "PJSIP/111-00000064" , is originator: Y , call-Id: "u.l6kcou25cax60 at mydomain.com <mailto:u.l6kcou25cax60 at mydomain.com> " , local_uri: "<sip:222 at mydomain.com <mailto:sip%3A222 at mydomain.com> ;user=phone>" , local_tag: "1734d973-c4da-4ae8-a37d-5f7065f1fe54" , local_addr:

Hello, just for my information, what happens if you append "-D_HPUX_SOURCE" ? Some (long) time ago (HP-UX 10.x), it would help compiling foreign software on HP-UX. But I had only at hand the HP compiler. Regards, J?r?me -------- Message d'origine-------- De: Hughes, Martin [mailto:Martin.Hughes@uk.thalesgroup.com] Date: jeu. 9/18/2003 13:05 ?:

Hi, There is someone (Tony?) with disconnection problems (after about 15s) when calling between two DIAX phones? I have a voicemessage regarding this issue, without any contact address. If yes, please send me more details about configuration (iax.conf and extensions.conf files, IAX mode, etc.). As another DIAX user requested that, I'll put on my site some sample configurations files to be

Thanks for your answer. So for my PDF printer, I switched back to BSD printing (calling cups-lpd), and keeped the local (to PDF printer) print command definition. It works now. I still had to add to cups (which automatically generates the /etc/printcap file) a fake printer with the name of the pdf printer (to avoid the loss of the manual addendum to /etc/printcap for the pdf printer). Best

Hi all, A new version of DIAX (0.9.6.d) can be downloaded from the following address: http://www.laser.com/dante/diax/diax096d.zip or http://www.geocities.com/tdanro/diax/diax096d.zip Take care that the link is not available from the web page. It has debug in a file support for IAX2 only. In order to activate debugging functionality, right click on the first box from the third row (in the right