search for: marnock

Hi, You can try ChanSpy http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanSpy. Idris _____ From: phil.dawson@marnock.com [mailto:phil.dawson@marnock.com] Sent: Wednesday, June 21, 2006 12:23 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Monitor a particular SIP call for training purposes Hi, I've been asked if it is possible to allow a user to listen in on another users call for tra...

...PBX. I have done several implementations on IBM x305 and x306 servers and they work great. I cannot comment on the x100 though, since I have no experience with it. Thanks, Steve Totaro www.asteriskhelpdesk.com <http://www.asteriskhelpdesk.com/> _____ From: phil.dawson@marnock.com [mailto:phil.dawson@marnock.com] Sent: Wednesday, January 25, 2006 6:34 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] ISDN / Analog Hi, We have 12 lines via ISDN30 but this seems to provide us with 6 Analog lines. We use the analog lines but what is confusing is ther...

Hi List, Not sure if this is the place for this so here goes ... We have a number of Polycom 501's connected to our * box and they work great. Some of our users have added a few entries into the directory on the phone. The problem is on those particular phones they now sometimes get "resource full" on the phone when accessing the directory. No central directory was configured.

I have a Linksys PAP2 connected to Asterisk. Have one of the FXS ports working fine. I am unable to get the other to work. Does anybody have an example configuration to make both work. Both are registering fine but there's just no dialtone on the non working port. TIA

Is it possible to give a caller three goes at an extension number then hangup? exten => s,1,Zapateller(answer|nocallerid) exten => s,2,PrivacyManager exten => s,3,Ringing(1) exten => s,4,NoOp(${CALLERID}) exten => s,5,SetMusicOnHold(random) exten => s,6,Background(silence/1) exten => s,7,Background(thank-you-for-calling) exten => s,8,Background(silence/1) exten =>

Maybe not the place for this but thought I'd post the info for others. I purchased a cisco 7960 off ebay and needed to convert to SIP for *. I know * supports SCCP but I wont go into that here. I'd read on voip-info.org that a contract could be purchased for approx $8 to allow me to download the firmware. I though, being in the UK, i'd get one through a reseller in the UK.

Hi, I'm currently looking to run Asterisk in the office to replace an old PBX and would appreciate a little help. We are moving offices and will have 8 digital lines. My questions are: As there are 8 digital lines is this known as PRI? Which Digium card would be the best fit? Would you recommend looking at the echo cancellation cards? We are UK based: is caller id supported by Asterisk

Trying to find my feet here. If I wanted to connect Asterisk to a PRI and throw in a T1 Adtran channel bank into the mix for fax machines would the following work? Connect PRI line from telco to Port 1 on the Digium Wildcard TE205P. Connect Adtran TA-624-T1 to Post 2 on the Digium Wildcard TE205P also, would I need a crossover to the channelbank or is it a patch lead like the connection to the

Does anyone know if version 3.1 is Call Manager or SIP? Thanks.

Hi, Can anyone tell me if UK CallerID support has been added to the CVS for the x100p card ??? I'm new to * so please excuse me if this question has been asked a million times already. Thanks in advance Phil.

...They support modem until v.90 speeds and faxes for g3. They are expensive, and again, work great and configure very easy joash ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of phil.dawson@marnock.com Sent: Friday, January 27, 2006 12:01 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] ATA's ??? Hi, I'm currently in the process of building Asterisk for our new office and have hit a snag. We need two internal Analog lines for a modem and fax machine. Am I righ...

what is a typical server / processor / memory configuration for approx 50 user install? phone -> asterisk -> pri thanks Phil.

how do these phones perform? ok for office use? work well with asterisk? any info would be appreciated.

Hi, I'm looking for IBM hardware to support: 100 SIP hard phone users 10 fax machines on SIP ata's maybe later an additional 100 sip soft phones. Initially, all calls will be through PRI. Some conferencing. Don't know yet if this will even get used. Using 1.4 + ( probably business edition ) I'm looking for anyone who some experience / gotchas. I've google'd and

Still finding my feet here. I need a server which can take approx 100 sip users accessing 24 channels through pri. Approx 20 concurrent calls. 6000 calls a month. Are there any "rules of thumb" when it comes to sizing hardware. I've checked the wiki but nothing close to what I need plus some of the information is really old now and may not be relevant. Also, what server

I'm in the UK and have had terrible trouble getting the right contract. After looking on voip-info I set off looking for a vendor only to find that no-one sells the $9 contract. Cisco retracted that one :-( I asked for the equivalent and they said I needed a ?20 contract. I said fine but after 20 minutes or so got a call saying I couldn't have that one as it didn't give me access

<font size="2">I've got an odd situation where callerid is only picked up every other call.&nbsp; Is there anything I can do so callerid works on all calls?<br><br>I'm seeing the channel hangs up during a call<br><br>&nbsp; == Starting post polarity CID detection on channel 4<br>&nbsp;&nbsp;&nbsp; -- Starting simple switch

Hi All, I have tried everything to get callerid to work reliably but to no avail. I have configured zapata.conf as per documentation but still only get 50% of callerid's through. As a test I called our system with my mobile a number of times and only 50% get through. I do get warnings about polarity. I am in the UK. Anyone have ideas what to check? TIA Phil

anyone have experience with switches with QOS. recommend makes / models? I've experience of cisco, polycom and snom phones but how do they compare to cheaper phones namely: Aastra 9112I Budgetone 200 any insight would be appreciated Phil.

single server dual xeon 3.2 raid 1 2 gig mem be overkill. i've looked at the server dimensioning but would like comments / opinions from anyone who has deployed a system for this many sip users. TIA