similar to: Again Asterisk and VMWare - it works now!

I've spent the past week experimenting with Asterisk@Home 2.6, and then Asterisk 1.2.6 individually, on VMWare Workstation 5.5. I have an entirely IP (hard & soft)phone setup (IAX and SIP) so I have no requirements to support any Digium PCI cards, etc. All in Asterisk works extremely well except for one thing: Playback of sounds (GSM format) such as an ivr greetings, sound terrible.

I have had no success in getting the voicemail working on Asterisk 1.2.11 on CENTOS4(2.6 kernel) guest on vmware server 1.0.1. I tried with or without ztdummy device, renice -20 on asterisk process and even real-time priority on the host Windows XP box for the vmware process. I am running on an AMD Athlon 64 X2 4600+. The behaviour is when the voicemail answer, the voice sound ok but when

Hi, There is any experience using Asterisk with VMWare? I think about installing a virtual linux box over VMWare and then Asterisk over it. Thanks, Dan

Hi Folks, I'm trying to test out Asterisk overall. I'm having some problems with DTMF. Currently I'm playing with DISA, but I'm worried this will happen when I get to implementing AAs etc. I have a free SIP trunk from IPKall that I'm trying to make work. I'm able to receive calls, and I've now setup and extension with DISA and a password. I connect ok from the

How come FireFly doesn't give me an Inband DTMF option? Only RFC2833 and Info. RFC2833 is the default, so I left it that way. I also unchecked all the codecs except g711ulaw to force that codecs usage. However, when I go to place a call, I get this: Nov 3 13:18:44 WARNING[53641241]: dsp.c:1468 ast_dsp_process: Inband DTMF is not supported on codec G.711 u-law. Use RFC2833 Nov 3 13:18:44

Hi All, I've just setup a test Asterisk system that allows incoming/outgoing calls via an ISDN card (l4i) and incoming/outgoing calls via SIP (iconnecthere). I have two SIP Softphones (Xten X-Lite) for making and receiving calls. When receiving an incoming call via the ISDN interface the sound quality is fine for the Softphone user (i can hear the caller perfectly), but the person

hi, i have connected a SNOM 200 to the asterisk. here are my settings, Codecs ------- Default codec - g.711u/g.711a Packet size - 20ms Negotiation - Interoperable Type - 160 DTMF ---- Inband - negotiate Outband - negotiate Payload Type - 101 when a call comes to the SNOM or when making an outdial, following warning messages are coming on asteisk, WARNING[1209214400]: File dsp.c, Line 1198

On 07/30/2017 02:07 PM, Walter H. wrote: > On 30.07.2017 20:22, Johnny Hughes wrote: >> On 07/30/2017 09:41 AM, Walter H. wrote: >>> On 30.07.2017 14:29, Johnny Hughes wrote: >>>> I personally have a Fedora machine that I keep updated and do some work >>>> on all the time learning/testing. I just seamlessly upgraded it from >>>> Fedora 25 to

Hi all, I have my asterisk setup as following: IP 2 x E1 x-lite <-------> Asterisk -------> PSTN When I place a call from x-lite to PSTN, the quality of the sound in the direction x-lite -> PSTN is very bad. That is, the voice of the x-lite user, heard by the PSTN user is choppy and makes communication not very pleasant. The sound is choppy as if bits of data

Greetings! I have made great progress thanks to this group. My Asterisk seems to be working for the most part. I am using the following equipment/software: * HP Vectra VL - Pentium Pro CPU - 256MB RAM * Redhat Linux 8 - Loaded straight from distro CDs as Developer Workstation - latest updates from RHN * Asterisk (latest as of two weeks ago when I used CVS checkout) * X-Lite SIP Client on a

On 07/30/2017 09:41 AM, Walter H. wrote: > On 30.07.2017 14:29, Johnny Hughes wrote: >> I personally have a Fedora machine that I keep updated and do some work >> on all the time learning/testing. I just seamlessly upgraded it from >> Fedora 25 to Fedora 26 using a couple of dnf commands .. awesome >> experience actually. > because of this feature to upgrade from one

I have configured my GS HT-486 for "send dtmf" in audio, and on the asterisk box, sip.conf has dtmfmode set to inband. Everything seems to be working fine, however, I see my console get flooded with the following warning: "dsp.c:1467 ast_dsp_process: Unable to process inband DTMF on 256 frames" Should I be cautious about them or just ignore? Better still, what should I do

Hi, there were some postings a few weeks ago telling that the GSM codec problem with snom100 will be fixed. But it still seems to be very quality. Will be any change in this subject? THX -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030516/2fa9d206/attachment.htm

Hi folks, I'm experiencing some problems with sound through phpAGI ... What I'm trying to do is a menu, doing some database lookups and so ... But sometimes the sound become too choppy ... just sometimes .. like 1 of 5 calls ... but is a big percentage ... And I have my current menu on the dialplan that I have no problems with it ... I'm using .gsm for both but different

Hi All, I'm stumped on this and I looking for some clues to fix this. This is a new install of Slackware 12.1 onto an IBM x330 Server. Asterisk 1.4.20.1 plays the wav files and the Cepstral_Allison Swift just fine, but when I play the gsm files the audio quite choppy. And, the files produced from the MixMonitor don't even record any audio other than noise. I have a hard drive from

On 31.07.2017 13:23, Mark Haney wrote: > Uh, I run VMWare workstation just fine on my F26 upgraded machine. No, > it didn't work when I upgraded, but it's trivial to fix. > > http://rglinuxtech.com/?p=1939 > > This link gets you a running workstation in about 5 minutes. not really, with this I only get the additional network interfaces listed with 'ifconfig',

Hello, Is there an utility for asterisk for codec conversion? I tried google but i haven' got anything. I am trying to initiate a call with G711 codec to asterisk and i would like asterisk to call a gateway with an g729 codec, therefore making a codec conversion from g711 to g729. I know chan_oh323 does it by specifying the OUT_CODEC variable, but chan_h323 does not. And i was wondering is

My ADSL connection speed is 512Kb up and 128Kb down. When making calls from Asterisk to IAX and back to the Asterisk, the sound is choppy and 20% of voice messages was lost. What is the production bandwidth requirement per internet call. I understand there is no guarantee of QoS but at least a benchmark to follow. -- David Kwok Iaxtel/FWD # 17001813482 -------------- next part

Dear Guys, I?ve test Xten Lite softphone to connect to my Asterisk Box but it registers all the three lines at the same time and if I try to dial an extension it tries to reach 3 Ext. at the same time, can somebody haved this trouble? and how can I fix it. Also, I ?ll like to have the Xten LITE or PRO Softphone (Lite is free and PRO about $50.00 USD) it can hanle 3 lines (lite) and 6 lines

Hi All, i dont what changes i made recently but i am unable to hold the call for more 10 secs between MSN and GS. PSTN to MSN and PSTN to GS and vice versa works fine. i am not behind NAT.Also MSN to MSN works fine too. my SIP details [general] port = 5060 bindaddr = 0.0.0.0 context = bogon-calls ;context = default disallow=all allow=ulaw allow=alaw allow=ilbc allow=gsm ;My SIP phone - GS