similar to: FW: SIP echo on LAN

Good Morning, I have a weird situation, I'm testing with Xlite as SIP phone (is it any good ) and dialing an extension (also Xlite on same LAN) and I'm getting a real bad echo on the dialer's side and a not so bad one on the receivers side. Has anyone had something like this ? Aparently one should only get echo when you break out onto a telco network ? -------------- next part

Sorry all, Did that and its going good now. Rgds Nic _____ From: Nic le Roux [mailto:nicl@i-procc.za.net] Sent: 20 January 2005 11:22 AM To: 'asterisk-users@lists.digium.com' Subject: Asterisk 1.0.3 startup Hi All, I've managed to compile make and make install asterisk on Mandrake 9.2. However on startup I get the following message: [cdr_tds.so]Jan 20 11:13:54

Hi All, I'm running Asterisk on the following vendor_id : GenuineIntel model name : Celeron (Coppermine) cpu MHz : 668.202 cache size : 128 KB with 192 MB Ram Audio coming from Asterisk (the demo ) is excellent when using a SIP phone on the LAN to Asterisk, and when dialling in from outside via ISDN to Asterisk. However, when connecting from SIP phone to SIP

Hello, I'm having the following problem when using a headset on XP connected to an on-board Realtek soundcard on an AsusTek M2N68-AM Plus motherboard: - Using any sound recorder (Windows', Audacity, XLite), the level is just too low when speaking at a conversational level, even with the microphone level pumped all the way up (line displayed totally flat in Recorder)

Hi all, I was excited to see the announcement on the list regarding the fix for the echo problems on Digium FXO cards! I have 2 X101P's, TDM400P with 4 FXS modules and couple of XLite softphones. A few months back,I had gone thru the recommendation on the list to remove echo from the SIP phones(I never did have any echo on the TDM400P FXS phones), and had removed about 90% of the echo.

Hi, I have a weird issue that I hope someone can help me with. I have 2 test computers and I've changed each the roles of each one with the same results. I have one xlite client running across a VPN and another connecting directly from the WAN via the external IP. The client connecting directly to the external IP can call the client running across the VPN no problem. However, the client

On Thu, Feb 6, 2020 at 11:16 AM Yvan Roux <yvan.roux at linaro.org> wrote: > > Hi, > > here are the results for ARM targets: > > * 32-bit has the same issue reported in PR44767 > > * same issue with quick-append.text for AArch64 check-all results are: > Testing Time: 4520.30s > ******************** > Failing Tests (1): > LLVM ::

Hi guys i signed up to iconnect a few hours ago to try do some cool stuff. but im having a few problems. Im running asterisk and xp/xlite softphone.. both xp box and asterisk box are on public ips. The problem is that when i ring anyone in the world it'll ring they'll pickup and i can hear them 100% perfectly/clearly.. but they cant hear me.. occasionaly they can hear something like a

Last line of output when running "smbclient //cupsipp/print$ -k -U nicl -d 10" Any Idea what this could be or how to resolve ? Not realy any info on Google I could find [2004/01/14 11:16:51, 10] intl/lang_tdb.c:lang_tdb_init(135) lang_tdb_init: /usr/lib/samba/en_US.UTF-8.msg: No such file or directory

Hello! I bought a virtual IP line to my ISP to use with my asterisk but when I try to connect it to my ISP tells me I can not use and I can only use with a softphone that gives me, xlite ready configured. I use ngrep to see what information sent on xlite for communication, the User-Agent was changed so I change the User-Agent to my asterisk to the same as saying the xlite but still does not work.

Hi, I've got an Asterisk box with grandstream and xlite clients on it. No here's the thing: - I grey out all the codecs on the Xlite except for GSM - I call the Grandstream from the Xlite, the Xlite uses the GSM codec and the Grandstream uses ulaw, with Asterisk doing the conversion, everything fine - I call the Xlite from the Grandstrea, the Xlite ends up using the ulaw codec as

The value of 14800 is correct. I had issues with my TDM400p with 2x FXO's installed and using the Xlite client. I could not get the echo stable at the initial call. Changing to a hard phone made everything work correctly. I still had problems with the off location I called, but mostly worked great. To see the TX gain I created an extension with the Milliwatt command attached to it and called

Hello I have an Asterisk 1.4 server and two XLite softphones, where Asterisk and the local XLite phone are located in a LAN behind a NAT router, and the remote XLite phone is located elsewhere on the Net behind its own NAT router: http://img252.imageshack.us/img252/3749/asterisknat.png I'm having the following issue: When the _local_ XLite calls out the remote XLite, everything works fine;

> > Different brand/model, but similar as they are both el cheapo, > entry-level headsets. I tried using them on a laptop, and I get > marginally better microphone output, even with its volume cranked all > the way up + automatic gain control enabled. > > I guess those on-board soundcards by Realtek aren't as good as a > quality microphones. I'll get a USB headset

My first Asterisk install: Debian sarge with the 2.6 kernel, and two X-Lite soft-phones. I followed the online how-to documents and was calling between the two soft-phones and calling the demo system with no problems and had full audio. I then went on to configure the TDM400P's two FXS modules. I got into that a ways and was having some success, but no dial-tone when I was off the

Hello I thought I had things set OK to have Asterisk play FR files for prompts and MOH, but for some reason, it still can't find them: ============ ll /var/lib/asterisk/sounds/ drwxr-xr-x 2 asterisk asterisk 4096 2011-01-21 16:18 custom/ drwxr-xr-x 10 root root 61440 2011-03-17 14:21 fr/ Note: fr/ contains core + extra + moh as downloaded from here:

Dear all, I have tried to run an asterisk instance together with XLite on a single machine (a PowerBook). The intent is to take advantage of IAX connections to easily cross NATs while traveling. While the IAX setup proved 'easy', just having to fiddle a little with working configs at both sides, I did not succeed so far in getting XLite to connect to the local Asterisk server, AND be

Asterisk version 1.2.14 I use snom190 and xliteV3 as sip phones. asterisk send the rtp stream never to the xlite softphone. Any hits for me? *CLI> rtp debug RTP Debugging Enabled -- Executing Dial("SIP/xlite-007918f0", "SIP/snom") in new stack -- Called snom -- SIP/snom-00797110 is ringing -- SIP/snom-00797110 is ringing -- SIP/snom-00797110 answered

Hello Asteriskians! I have an Asterisk box with a simple HFC card in it and a bunch of people using the Xlite software to connect. The HFC card is connected to an internal extension on our legacy PBX. So far so good. The Xlite clients can call each other, and the internal extensions on the PBX and the Xlites can call each other, no problem. The problem is when using an Xlite to dial an external

I have a bunch of road warriors who I've set up with Xlite clients. Unfortunately the sound quality has been intermittent at best. Sometimes it's great other times completely unusable. When it's bad one usually hears harsh static when the other party speaks or their voice gets "clipped" to static if they speak too loudly. Many of these users have migrated to Skype ? much