similar to: Sipura 841 and headset

On Tuesday 10 May 2005 9:45 am, David Masure wrote: > 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 ! > ... > > Or....can anyone advise me on headset working with the sipura 841 ? I just use a

Hi all, I'm confused, in digium website, it says: TDM2400P: It supports a combination of up to 6 FXS and/or FXO modules for a total of 24 lines. 6 plus 6 is 12, how come it's 24? if I have 24 PSTN lines, i'll be needing 24 FXOs. Pls. elaborate. thanks. Lito -------------- next part -------------- An HTML attachment was scrubbed... URL:

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'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'm looking at switching VOIP providers, but want to ensure I move to a company with sufficient capacity. Can any Canadian VOIP users post/email me with feedback on their providers? I'll post the results for all to read...... -------------- next part -------------- An HTML attachment was scrubbed... URL:

Sorry, I know it's OT, but does anyone know of a relatively inexpensive headset that is compatible with the Cisco 7960? I've tried the headset off Norstar phones, doesn't seem to work with or without the amp. ------------------------------------------------------------------------ | nate carlson | natecars@natecarlson.com | http://www.natecarlson.com | | depriving some poor

Greetings, I am trying to get either of the above features to work with *, but can't seem to get it quite right. If anyone has them working, I'd sure appreciate an extract from the relevant config files. Or, maybe I'm tilting at windmills, and * doesn't support them - in which case, the underlying business need is to provide the one incoming call on more than one

I am looking for an ATA that has had very reliable results when passing FAX over IP. I was thinking of testing the Cisco (not Linksys) ATA 186 I1, ATA 186 I2, ATA 188 I1. This is what I'm looking for: FAX -> PTSN -> through Asterisk -> ATA -> Fax Machine. I have QoS from PSTN entry to ATA on the network so I can assure precedence. What has everyone out there been using

We have a server with a TE120 on a partial PRI trunk that several times a day declares the PRI trunk down and stops handling calls until the asterisk is stopped, the zaptel/te120 modules reloaded, and asterisk started. Just before things go down, the log shows the following error: ERROR[9424] chan_zap.c: Write to 28 failed: Unknown error 500 at which point a "show pri spans"

I am looking for some Linksys and GrandStream ATAs in Canada. I am looking for places that ship from Canada so I don't have to deal with the clearing of customs and tax remittance. Any suggestion? -- Thanks

From c|net News: "On Monday,Microsoft and nine leading phone manufacturers--Asustek Computer, GN, LG-Nortel, NEC, Plantronics, Plycom, Samsung, Tatung, and Vitelix--announced the public beta program for Microsoft Office Communications Server 2007 and Microsoft Office Communicator 2007." http://news.com.com/8301-10784_3-9719931-7.html?part=rss&subj=news&tag=2547-1_3-0-20 --

Hello! In my relentless quest for knowledge, I pose this question: who's got the biggest dialplans, and how big are these monsters? What's the biggest dialplan in use right now? If you feel you are a competitor, let me know how many contexts/extensions/priorities you are dealing with. Maybe the context with the most extensions, the extension with the most priorities would be

I need to accept an incoming call, make a series of outgoing calls, and once I find someone willing to accept the call, bridge the original incoming call to the outgoing call. Using Dial from an AGI script isn't enough because once the Dial'ed number connects, the call is immediately bridged and I need to ask the called party if they will accept the call. I can see a couple of

I've been scratching my head trying to think of a way to do this, but without success yet. I'm using the Manager API. If I have two channels linked to each other (i.e. direct connection), or even if they are independent channels, I can transfer them both to the same extension by using Action: Redirect and using Channel: for one and ExtraChannel: for the other. This is most useful for

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

2008/9/29 Alex Balashov <abalashov at evaristesys.com> > Try this: > > exten => _XXXX,1,Set(THISTECH=${CUT(CHANNEL,/,1)}) > exten => _XXXX,n,NoOp(Technology is ${THISTECH}) > exten => _XXXX,n,Set(THISCHANNEL=${CUT(CHANNEL,/,2)}) > exten => _XXXX,n,NoOp(Channel is ${THISCHANNEL}) Hi, I don't have any spare zaptel enabled system I could try this on, but I

I would like the ability for a sipura sip device to instantly connect to an asterisk server as soon as the sipura sip device goes offhook and before any digits are pressed. This way asterisk can provide the dialtone and the dialplan. This also allows me to play a greeting to the phone before giving them a dialtone. Is there any way to do this, like possibly having the sipura device dial a

Just wondering how your experience is with HPEC, Is it just for analog interfaces or we can use it on TE122 as well? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080214/3f0580f1/attachment.htm

Has anyone done interop testing with Level 3 and Asterisk. If so, would you be willing to share your experiences. Gene -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050308/061aea35/attachment.htm

Hello List, I am curious how the ordering of the extensions are determined for an ARA dial-plan. For example, if I have these: _9X. _9011. Which is selected first? Any number dialed starting with 9011 is matched by either rule here and I don't remember seeing any ORDER BY clauses when I had debugged the ARA queries. I'm sure I just missed some critical documentation here.