similar to: SIP video voicemail problem

Okay, now that I've got NUT up and running with the newhidups driver, I can give it a quick shakedown. Most of the values look good, but are few are off. Here's what I get: elrond@foxstar:~$ upsc foxstarups@localhost battery.charge: 100 battery.charge.low: 10 battery.charge.warning: 50 battery.date: 3150/08/01 battery.mfr.date: 2005/03/22 battery.runtime: 1935 battery.runtime.low: 120

Hello, I am having trouble getting upsd to start. I am using the Debian NUT packages version 2.0.3 (current in Debian/sid). I am using the newhidups driver with an APC Back-UPS BR 800. The init.d script wasn't functioning, so I backtracked to running upsd manually and I get: foxstar:/etc/nut# upsd -u root Network UPS Tools upsd 2.0.3 Can't connect to UPS [foxstarups] (newhidups-auto):

I'm trying to get two Windows Messenger clients to communicate with video and audio though asterisk. I'm running into one of two problems. I get garbled audio under the current config. I had another config where I could get a voice call to work but using video would cause the caller to get music on hold. (very odd) Calling a phone hanging off of an TDM the audio works great. This is

Hello, Upon upgrading to the Debian package of NUT 2.2.0 my APC Back-UPS Pro is no longer recognized. Here is the output if I try to run the driver manually: ----------- # /lib/nut/usbhid-ups -DDD -x vendorid=051d -x productid=0002 -a foxstarups Network UPS Tools: 0.28 USB communication driver 0.28 - core 0.30 (2.2.0-) debug level is '3' No appropriate HID device found No matching HID

Built asterisk 11.8.1 on a Debian VPS. Testing using ekiga. If h261 is checked in ekiga's video format list I have video, and mouse over the video window shows it to be using h261. But then I get the following lines a dozen or more times in the CLI: [Mar 21 16:25:32] WARNING[31818][C-00000010]: file.c:1241 ast_writefile: No such format 'h261' The problem is that I can't seem to

--- listas iPfone <listas@ipfone.com.br> wrote: > Hi all > > I need to know if the video support for h.263 is > active in version stable > 1.0.7 to use with eyeBeam in asterisk it works for me... [2222] type=friend secret=xxxx auth=md5 callerid="myCallerId" <2222> canreinvite=no host=dynamic disallow=all context=default allow=alaw allow=ulaw allow=speex

Hello list, I'm trying to set up a video call from my Ekiga client to a Grandstream GXV3140 IP-phone. The call succeeds but there is no video. I have in sip.conf : videosupport=yes disallow=all allow=alaw allow=g726 allow=g729 allow=gsm allow=h261 allow=h263 allow=h263p allow=h264 The Grandstream peer has codecs (sip.conf) : gsm;alaw;g729;h261;h263;h263p;h264 The Ekiga peer has codecs

i have posted this before but was unable to resolve it. i have some new info so i figured i would try again. the trace from bandwidth.com are below. they are telling me that the ip that is bold should be our ip not bandwidth.com. i have changed every setting that i can see and nothing fixes this. Where would i change this at? they cannot tell me. INVITE sip:+185993133333 at 216.82.224.202

Hi, I've one Sipura SPA2000 at home behind a linuxbox with two network adapters (eth0 for WAN and eth1 for LAN) doing NAT/DHCP: ___________ HOME _______________ ____OFFICE ____ SPA2000 <---> Linux Box <--> Asterisk Box 192.168.0.253 192.168.0.1 eth1 200.93.xxx.a 200.93.xxx.b eth0 My problem is when I try to call to any trunk or extention

how do i troubleshoot no ring tone. It was working and all i added was the lines below now it doesn't ring. Edit sip_nat.conf for proper NAT: localnet=192.168.1.0/255.255.255.0 externhost=pbx.DOMAIN.com (Set your external hostname name here) externrefresh=10 fromdomain=DOMAIN.com (Set your external domain name here) nat=yes qualify=yes canreinvite=no Add extra codecs to

Am 13.06.2020 um 13:47 schrieb Michael Keuter: Hi > Try "sip show peer <peername>" for a phone. So: mobile phone: bpi*CLI> sip show peer 0049177xxxxxxx * Name : 0049177xxxxxxx Description : Secret : <Set> MD5Secret : <Not set> Remote Secret: <Not set> Context : default Record On feature : automon

Hello, I have some problems to use G722, when my client sent an invite request to asterisk using G722/16000 codec asterisk respond with G722/8000 codec. I dont know exactly if Asterisk supports G722/16000 codec?? If yes how can I activate It?? Thanks. Rachid. Below wireshak trace:

Hello, everybody: I'm developing an application using Asterisk and a TDM-400 card. I understand the concept of the difference between GSM and WAV files when using Asterisk, but I'm not happy with the sound quality with the GSM compression. It's merely *acceptable* for a telephone call, but for anything else, it leaves something to be desired. Case in point -- if you compare the

I am having trouble getting Google Chrome to accept a WebRTC call coming from Asterisk, even though Firefox can (now) accept the same call without issue. My setup is as follows: Server: CentOS 7 x86_64 (Elastix 4 RC) with IP: 10.1.0.4 192.168.5.146 asterisk-11.21.0 patched to work around https://issues.asterisk.org/jira/browse/ASTERISK-25659 openssl-1.0.1e-51.el7_2.2.x86_64 [root at elx4 ~]#

I am looking for a SIP Soft Video phone, which I can use with Asterisk. If you have one installed (regardless if free or purchased) please tell me which one, the settings in Asterisk and your experience with it. bye Ronald

Hi List, I have another issue on allowing outgoing calls to PSTN on Asterisk via Avaya Phones, I hope that anyone could help me fix this issue: *When I dial through Avaya phone i just here a "good bye message" reply from asterisk server. And here is the log:* == Starting OOH323/(null)-b7db8aa0 at internal,s,1 failed so falling back to exten 's' == Starting

I am configuring a test Asterisk server (1.8.9.2) to practice setting a single codec globally, to avoid transcoding as much as possible. Since all of my recordings are in gsm format, I am trying to make the SIP clients use gsm everywhere. I am using Ekiga on Fedora 16 x86_64 for my tests. [root at elx2 asterisk]# cat /etc/asterisk/sip_general_additional.conf

Hi list,* *I am not able to make video calls between two sip accounts. below is the information. please help me where I am missing the configuration.* Extensions.conf* exten => 111,1,Answer() same => n,Dial(SIP/2206,60,r) same => n,Hangup() *SIP.conf* [2218] type=friend secret=******* callerid="Virendra" <9172341457> host=dynamic ;

I'm doing some final check-outs before upgrading from 1.4.x to 1.6.x and I've encountered a problem playing back a .wav file to an Ekiga client: My dialplan looks like: exten => 730,1,answer exten => 730,n,playback(/home/phones/common/moh/moha/Sovereign) exten => 730,n,hangup Sovereign.wav is a .wav file that plays nicely on my 1.4 server. Here is what the console displays:

Hello All, I have a media problem while using sip communicator user agent with asterisk behind NAT.I had enabled the debug mode in asterisk and capture the results.I have attached the results with this mail.Can any one help me to fix the problem? Thanks in advance, Partha __________________________________ Do you Yahoo!? Yahoo! Mail is new and improved - Check it out!