similar to: CallTransfer

Any help on this would be great I have 2 TDM400P's, 2 asterisk servers (running on powerful boxes with FC1 and * v CVS 1.0.02), and 4 analogue PSTN lines from BT and whatever I do, I cannot get rid of this damn local echo. Ive tried setting the echoTraining, echoCancel (in phone.conf and Zapata.conf) , echocancelwhenbridged to every possible combination , Ive even tried running the fxotune

Hey everybody, I've been playing around with Asterisk (Current CVS Stable dated: Asterisk CVS-v1-0-11/23/04). I've purchased 2 GS BT102 SIP phones. Both have been upgraded to firmware 1.0.5.18. I've also have installed on my desktop and laptop, X-Lite. I've been using these to learn how to setup Asterisk. I've got a Wildcat X100P on order and will be here next week. My

Same problem here: if call come over ISDN PRI and it is for a SIP phone that equals to strong echo situation, at the SIP end. Interestingly this doesn't happen on all calls but it does on 95% of them. Asterisk load at that moment is insignificant - 1 to 2 calls. I have tried with all possible echo cancellers in zconfig.h, with and without MMX, and with and without CFLAGS+=-march=i686 in

HI ALL; Anybody can explain the difference between "call parking " vs "call transfer" Regards mohammad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040720/2975991b/attachment.htm

FYI for anyone using the dual ethernet ports on a Grandstream BT102. I'm using a BT102 connected to an HP2524 10/100 switch, which has an asterisk box connected directly to it. No VLANs defined or in use. Measured bandwidth: PC -> HP Switch -> Asterisk : actual throughput measured at 94.1 mbps. PC -> BT102 -> HP Switch -> Asterisk : actual measured at 8.86 mbps. The

Following a(n apparently) failed attempt to upgrade a BT102, the phone is now brain-dead. Although it still has enough smarts to get a dhcp address and try to download the firmware and config, it never gets past the blue screen, nor will it respond to pings or port 80. Short of sending it back to Grandstream, is there any way to recover the phone? TIA Bruce Komito High Sierra Networks, Inc.

Hey, two questions regarding setwd and list.files: I try to use the function list.files: >datapath<-'/nfs/home/m/cremer/gaw/data/traws_chrs' >fl<-list.files(datapath, pattern="\\.tped") but I get the following error message: Warning message: In list.files(datapath, pattern = "\\.tped") : list.files: '/nfs/home/m/cremer/gaw/data/traws_chrs/'

Hello. I'm trying to transfer calls from an analog phone (Zap/1, TDM400P card) to some other terminal connected to my Asterisk PBX. If I make a flash hook pressing the phone hangup button quickly it works as expected, I get a new dialtone and the other side is put on hold. But I would like to use my phone's "R" key instead for some different reasons (it's quite easier to use

Hello, This morning, I asked myself if WebRTC could be a viable alternative to softphone deployment. For me, main issue with Softphones is the amount of work needed for installation and configuration. Also, Softphones must be carefully choosen if Deskphone-like quality is expected. Now that WebRTC becomes ubiquitous, it might make sense to trade Softphone features (call history, BLF, ...) for

Sorry Newbie asking everyones option. I am setting up a couple of small asterisk phone systems for my work, I started using some snom 190 and bt102 sip phones (the bt102 works really well with iLBC), but the complaint from my workmates is there is no way to see if other people are on there phone or not, or what lines are being used. The snom 190 only has 5 function keys, the snom 220 seems a bit

Hi, When I use the analog phone connected to Zap/1 how do I transfer & hold the caller ? When I hit the flash key, all that happens is the caller hears a beep (sounds like DTMF). But no stutter dial tone on the Zap/1 Port, just continuing conversation with the caller. What could be wrong here? Cheers Jay

Hey all, Wondering if this is possible.. Incoming call is answered through X100P, then an extension is dialed using the same X100P card. Basically I want to dial in, enter 9 + <phone#> and have it do a flash then have it dial *08 <the same phone number> + # on the same PSTN line to have it transfer my call to another phone number. I realize this isn't very safe, but I would

Hi, I have been doing various testing with asterisk and its been going great. However I am a bit feedup of using vi for editing configs, and would rather do it from any machine on my LAN. I am running debian and * via xorcom rapid on a test PC at the minute. Hence phpconfig would be great, however I am having difficulty getting it to work. I have searched the message boards and the wiki, and

Hai, In addition to Rowlands question. Can you run this script and post it to the list also. It gives a complete overview of what your running. Its basicly what Rowland asked, but with a few extra things. https://raw.githubusercontent.com/thctlo/samba4/master/samba-collect-debug-info.sh And the output of: kinit Administrator klist klist -ket /var/lib/samba/private/secrets.keytab

Hi all, Does anyone know of any providers that can offer local numbers based in the UK via IAX or SIP? We're looking at getting a number based in the UK. Thanks! -- jeremy bogan [ jeremy@segpub.com.au ] segment publishing - design.develop.host

On Wed, Sep 26, 2018 at 9:40 AM Carlos Chavez <cursor at telecomab.mx> wrote: > > On 9/26/2018 4:46 AM, Olivier wrote: > > > Hello, > > > > This morning, I asked myself if WebRTC could be a viable alternative > > to softphone deployment. > > > > For me, main issue with Softphones is the amount of work needed for > > installation and

I have today reported a bug with the latest channel.c (1.134) that affects music-on-hold for the first user in a MeetMe room when calling from a Grandstream BT102. The music is broken up about 5-10 times a second. It doesn't happen when calling from Firefly. It is also fine on both clients with 1.133 of channel.c. I am using the ALAW codec. Mark at Digium can't reproduce the problem,

I've been chasing this problem for several days. I have taken the 3.0.26a Fedora 7 SRPM from the Samba FTP site and rebuilt it on RedHat EL5 and installed it. It runs, but it has exactly the same problem as the original 3.0.23c version. I can join the domain and 'wbinfo -t' returns OK. However, 'wbinfo -g' returns only two groups then gives a failure message and

Hi, Has anyone experienced intermittent echo issues with voipjet lately?

I have installed nut 2.0 Var/log/messages: Aug 14 16:37:26 server kernel: hiddev96: USB HID v1.10 Device [American Power Conversion Back-UPS BF500 FW:814.s3.I USB FW:s3] on usb-0000:00:1d.0-1 Aug 14 16:37:26 server kernel: hub 5-0:1.0: USB hub found Aug 14 16:37:26 server kernel: hub 5-0:1.0: 2 ports detected My ups.conf : [APC350] driver = apcsmart port = /dev/usb/hidddev96 desc =