similar to: HardPhone PlanetVIP-150T - Starts music on Hold and i can't get the call again

Hi all, I've asterisk 1.2.5 , and what is happening is this: Sip user agent "A" calls a pstn "phone B" Sip User agent Activates MOH. "B" starts listening. "A" doesn't hangup and just Disconnect Sipoftphone XLITE (exit) "B" stills listenning Music on Hold and "A" has left Asterisk, who hangs the call? only when B hangs...

Hi, My output of smbstatus is bugged somehow in the Locked files section. For every line with a non ascii character, the whole string from the character to the end (included) is removed. Hence the line is attached to the next one. Here is a simplified example Locked files: Pid ... SharePath Name Time --------------------------------- 8910 ... /path

hello I'm trying to figure out if anyone's accomplished putting someone on hold with a hardphone that doesn't have a hold button or multiple lines. I'm thinking transferring the caller to a specific extension or something...is this possible? Has it been done? thanks hank

I wish to know if there is a way to show a graphic file with 640X480 when starting syslinux bootloader withou showing the boot prompt. I want to show the graphic, then, after 4 seconds, just boot the linux, with no prompt. I tryed to make this in the syslinux.conf : prompt 0 DISPLAY boot.msg timeout 40 in the boot.msg: ^Xsyslogo.lss But it doesn't work. The timeout timer don't work.

I sent this last night but it never showed up in the thread list so I'm trying again. Please pardon me if it duplicates. So I've been banging my head against the rack on this one and am now turning to the group for help. I'm in the process of bringing five Asterisk servers (all originally built from source code by myself) from various versions (1.6.2.x,11.6-cert13, and 13.1-cert2) up

Hi all, I have a Cisco ATA186 connected to an Asterisk Server (SIP) The dialplan uses 1XX for local extensions and XXXXXXX for external numbers, where the first digit is always different than 1. In this moment, when I dial 123 for example, ATA waits till timeout before dialing that number. The same for the longer one. How can I do to make it dial imediately when 3 digits starting with 1 are

I am trying to do a classic R -d valgrind --vanilla < mypkg-Ex.R as described in http://cs.swan.ac.uk/~csoliver/ok-sat-library/internet_html/doc/doc/R/2.9.1/doc/manual/R-exts.html#Using-valgrind The problem is valgrind crashes imediately. I am using Ubuntu 10.04 LST (actually Biolinux) on a 64bit Dell Precision laptop with 4 core-i5 Intel processors. I have R-2-15-0 installed from source

Hi, I'd like to have asterisk play a sound file while a caller is waiting to be connected to an extension. I tried using music on hold, but that seems to run in a loop, not playing from the beginning for each caller. Are there any other options? It doesn't have to be an MP3 file. I tried sending a background() command before the dial(), but that doesn't appear to work.

---------- Forwarded message ---------- From: Marco Mouta <marco.mouta@gmail.com> Date: Feb 15, 2006 1:58 PM Subject: Which ATA device do you recommend? To: asterisk-users-request@lists.digium.com Hello, I'm developing a Voip Solution for a client, which ATA SIP do you recommend? there are some ATA devices fully tested with Asterisk? I hope that Asterisk experient users could give me

Hi I have installed samba 2.7 default with RH 9.0 ,authenticated by win 2000 active directory domain ,which is working fine.But Some of windows clients are not able to connect the samba server on next log-on. I am seeing their old connection in active state in samba SWAT tool.If i kill their connection from SWAT ,will be able to login and access the share. Is their any alternative methoid to

From: Rodrigo Barbosa <rodrigob at suespammers.org> > Actually, there is another neat trick for rpm based systems. > You see, rpm, prior to removing anything, will rename that to ${NAME}.OLD. > So, libc.so.6 becomes libc.so.6.OLD, and then removed. > As we all know, if that library is currently open by any running process, > it won't be imediately removed (even tho you

Hey is there a more consistent way of connecting to Rserve. I have code that opens a connection to Rserve by executing explicitly using Java runtime ( "C:\Program Files\R\r20XXX\Rserve.exe" ). However when i run the app and attempt to connect, it does not connect imediately.I have to close the app and open it again for a few times before i get a connection. Am i doing something wrong?

I've got Asterisk up and running nicely using a couple of different softphones. Audio quality is suffering a bit due to the hardware that I am working with. So I tried to use a Polycom hardphone but the politics is enough to give you a headache. Polycom seems to support SIP only if you buy it thought their vendors. So I'm looking at a Cisco phone. Has anyone successfully implemented

Hi, Which is your favorite desktop wifi sip hardphone ? I'm looking for something like http://www.mitel.com/DocController?documentId=19401 which could be easily moved from one meeting room to another. (In this specific case, finding an electrical plug to power a large desktop phone is seen more relevant than finding an PoE Ethernet plug or using a mobile handset.) Which product would you

I have an asterisk Redhat 9 box running 4 hardphone extensions. Inter-extension calls are crystal clear. However when I dial out through my iconnect account I get a lot of jitter. At first I thought it was my nat gateway but after I programmed on of the hardphones (budge tone 100) for direct dial to iconnect I have clear voice transmission. I have no way of explaining this. My asterisk sip.conf

Hello, Can anyone recommend a hardphone that has the following qualities... Both headset and handset ports Headset port has amplification built in Supports SIP using G729 We are switching from a Nortel switch to Asterisk. If anyone is familiar with Nortel phones, the Nortel 2216 phone has the features we're looking for in a SIP phone. Any help would be greatly appreciated. Thank you.

Are there any IAX speaking "hardphones" out there? If so, can anyone offer comment on their quality? Thanks! -Dorn

Hey all, I've setup my asterisk install on a CentOS5 server, I've got a few IAX2 and SIP softphone clients connected on the same subnet and at least 1 external IAX2 softphone. However I'm having some difficulty getting the Polycom hardphone to function correctly. Watching the logs and debug trace it: - Registers correctly - Is able to make calls to other peers However it is not able

Could someone please recommend a reasonably priced IP phone that works well with *, has a decent (full duplex, echo canceling) speakerphone, has at least two line appearances, and can transfer / conference reliably? The Wiki lists 35 brands of hardphone, but: 1. Most seem to be toys. 2. For many, there is no info on e.g. speakerphone characteristics. 3. When one seems technically promising, e.g.

AT-320EE Anyone try these? Do they work? any reviews? I couldn't find jack on google..