similar to: Re: Sipura 841 and headset (Josiah Bryan)

Hi folks ! I bought two sipura 841 phones. I used to have GN Netcom headset which I connect instead of the handset. The problem is that I don't have any sound coming out the headset and I can't speak neither ! I'am located in France and I was wondering if the cabling in the sipura and in the headset is the same (I mean the order of the cables) or maybe is there something else to

I was just browsing my local suppliers list of headsets - not a single one with a single 2.5mm jack. Either USB or 1-2 3.5mm jacks. Can anyone recommend a headset that works with e.g. SPA-921 and -941? /Per Jessen, Z?rich

I have been looking for analog phones for my * system that work with our plantronic amplifiers and headsets. The problem I am having with the Aastra phones that I have purchased (PT-390, 9116, 9120, 8009 ), is that they don't seem to stay hung up unless you physically hang up the handset everytime you finish a call. I have even purchased the Aastra 9120 which sais it has on-hook

> > Thanks Josiah Bryan, > > I do not have any dns server running on my asterisk server, we do have an > external DNS server working in the data center; the IP of this dns server is > 10.4.1.5... > > Following you will see my main configuration: > > /etc/resolv.conf: > > domain localdomain > search localdomain > nameserver 10.4.1.5 > nameserver 10.4.1.2

You described it well; I?m afraid that?s what it is. They want documents of an organization that doesn?t fit the description, and only then will they test it. That sounds great. I?ll reach out to Jeremy and see if he can help me get the IDs the USAF wants. It sounds like Joshua predates my time here. That?s a really good idea. Thanks very much! Doug From: Josiah Parry

Hey all - Attached is quick perl script i whipped together this morning to dial from Kontact (well, KAddressBook). You'll need Net::Telnet (perl -MCPAN -e 'install Net::Telnet') for it to work. You'll also need to make sure you've got a valid user/secret in /etc/asterisk/manager.conf on your * server for this to work. Usage: /usr/bin/asteriskdial --from

Hi Folks, I've been experiencing something very strange... When I want to listen to call between a SIP phone and a Zap Channel, I can listen a with a nice audio quality. When it comes to record using the monitor command, I just have a wav file which is completely noisy and I can't hear the conversation. Questions ? 1) Is there a way to specifiy the audio quality using monitor ?

Hi, Question... Is there a way to receive two phone calls on the same phone, or, for example to receive a phone call, put the call in stand-by and then make another call and finally, why not put them all together in conference... Thanks David Masure -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, i know this question is not directly asterisk related - but i have no idea where else to ask. We do have around 50 pieces of LinkSys SPA941 - these phones do have a 2.5mm plug connection - and we do have many many headsets we used with normal PC's before (so 2x3.5mm plug connection). Does anyone here know where i can get an adapter 1x2.5mm -> 2x3.5mm ? Or can anyone here tell me

Hi all, I have an asterisk box set up with a bri card (using zaphfc). I have a bri cellular gateway connected to it beacuse I'd like to route all my cellular calls through that gateway. The probel I encounter is that when trying to dial a phone number, I've the message : unable to create a zap channel. My card is normally well configured because when connected to the NT, It works

> > 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

Hi all, I would like to know if there is a function/setting for extracting the call setup time in asterisk (1.4 or 1.6). I need this value for every call processed by asterisk as specified in (ETSI TR 101 329-1 v3.1.2): Call set-up time is the time elapsed from the end of the user interface command by the caller (keypad dialling, E-mail alias typing, etc.) to the receipt by the caller of

I've been using asterisk for 3+ years now, I love it, but it doesnt love me back. :-) It was crashing frequently and seemingly randomly prior to this latest upgrade. Not sure what version it was running prior to upgrade (it was probably an old CVS HEAD from 2+ years go.) Anyway, currently running 1.4.21.2. == Problem == Problem is that its crashing for seemingly no reason at all, no

I have seen on eBay adapters for Cisco 7940/7960 phones, to use cell phone headsets. They were about 12-15$. I think original manufacturere was at http://www.ciscoheadsetadapter.com. >Started a Wiki page here: > > http://www.voip-info.org/wiki-Cisco+Phone+Headsets > > >Jim > >James H. Thompson >jht at lava.net > > ----- Original Message ----- > From:

Hi, I'm trying to use RODBC without having to set up a DSN, using hte direct connection string in odbcDriverConnect. My connection attempt looks something like: > odbcDriverConnect(connection = "SERVER=localhost;DRIVER={/usr/lib/odbc/libmyodbc.so};DATABASE=myDB;UID=reader;PASSWORD=insecure;") And this returns the message: Warning messages: 1: In odbcDriverConnect(connection =

I have the same problem in a box with 2 HFC-PCI, but i already remove the row in modprobe.conf and load the module manually. Both cards works fine Any idea ? Giordano -----Messaggio originale----- Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Julian J. M. Inviato: luned? 27 giugno 2005 12.04 A:

On 06/05/2015 03:43 PM, John R Pierce wrote: > On 6/5/2015 1:33 PM, Michael Hennebry wrote: >> It's a desktop in an old house. >> The outlets have ground-fault protection, >> but the third prong is ungrounded. > > not sure how GFI would function at all without a valid ground, unless > the GFI is wired to neutral, which is dangerous on its own. very. it defeats

Hi there, Do anyone have any experience with SIP phone that support a headset ? We have Budgetone phones but we need headsets. What would you advise ? Thanks Thibault

Hi, I want to use the cdr to record the call log to my Microsoft SQL Server using unixodbc and freetds.... but when I compile, I've got this message.... Does anyone have the same problem and/or know how to solve it ? Thanks Baste regards David Masure make[1]: Entering directory `/usr/src/asterisk/bristuff-0.2.0-RC7k/asterisk-1.0.6/cdr' gcc -pipe -Wall -Wstrict-prototypes

Hi, I'm facing something strange but maybe I haven't the right solution. What I want ot do is : Someone from outside call my phone number, I check some informations using an IVR script and then I want to transfer the call to an external phone number. The point is that when I'm doing that using the Dial command, I've got a message telling me no is able to answer the call. But