similar to: 12 seat call centre with Asterisk, VoIP only, UK - possible?

anybody recommend a supplier in the UK for a pri/isdn30 line (other than BT) thanx very much __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

Hi :) My employer is looking to move a call centre to a new office, and has been increasingly frustrated with their legacy PBX (call-logging licensing and hardware upgrade costs). So I've stepped forth as the Open Source Pedant and suggested Asterisk so we can do all our own CallerID / call logging / analyses, and make use of IP Phones / teleworking, etc. The problem begins in that I only

A client used to use BT isdn30 and ported the numbers to telewest several years ago. Now, the client moved to adept telecom. I *think* adept resells BT products. We got new numbers from adept (bt?) and the old pbx on the telewest lines forwards the calls to the new numbers. On the adept line I got a digium card in an opteron supermicro server. ztcfg gives me over 99.99 pretty much all the time.

Hi, I've just been made aware of Asterisk through the article on Slashdot. We're a small UK company with a ISDN-30 link to the office and a Meridian (Nortel) PBX. We're expanding and moving offices soon and I'm looking seriously at VoIP technologies. So starter for ten, can Asterisk interface in with ISDN-30 technology? I'm assuming the set-up is a server sat running

Hi i want see if my QoS are good because i am not very sure ... the VoIP quality are not very good when i download. I have on my Linux routeur/Firewall Asterisk .. and i have into my config : ================================================ tcdevices: eth0 2000kbit 2000kbit tcclasses: eth0 1 100kbit 180kbit 1 tos=0x68/0xfc,tos=0xb8/0xfc eth0 2 full/4 full

Hi, Are there any hardware for ISDN30 ? if yes any problem with this ? is i out-of-box like ISDN2 but with 30 linies ? Do I need more than the cable from my teleprowider and a PCI-card ? /HHA _________________________________________________________________ Find high-speed ‘net deals — comparison-shop your local providers here. https://broadband.msn.com

You should be able to get ISDN though. I think BT provide this anywhere. Otherwise your solution will be a Primary rate (ISDN30) with 8 channels. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Matt Sent: 07 June 2004 10:19 To: asterisk-users@lists.digium.com Subject: Re: RE: [Asterisk-Users] Multiple DDI &

I'm looking to expand my bandwidth for my Asterisk PBX. Why should I choose a T1 over DSL for my asterisk server? I found someone offering T1's for $290 a month + Loops or 3 Meg for $561 a month + Loops. Is this a good deal? Thanks Bart -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hey all What VoIP networking monitoring, asterisk monitoring tools would you recommend? I started working with an IT company that insists on using DSL with a Sonicwall router. The problem is that the clients are having sound problems. The owner is convinced that it's the Asterisk box. In the 4 yrs I have been doing this I have not had this bad a sound problem and it always came down to a bad

Hello, I have a Sangoma A101 connected to an ISDN30 (E1 in the UK) with some Snom 300 and Idefisk softphones. I can do SIP and IAX2 calls just fine, however I cant get any audio in either direction on the Zap channels. When I call in or dial out over the ISDN30 (UK E1) I can see the call answered/placed on the CLI and then silence follows. I've been provisioned 25 out of the 31 channels only

Hi There, We currently using iproute2 for load balancing. However we need more upload speed as we load balance over 3 dsl lines. I''ve been looking for a way to combine the upload speed to more faster. Found a site called www.upstreaminter.net where these guys bond the adsl channels to improve uploads, Since downloading is problem as its need to know the ip address of the downloader they

Hi again, I keep posting about my problem with HTB -> http://mailman.ds9a.nl/pipermail/lartc/2005q3/016611.html With a bit of search I recently found the exact same problem I have in the 2004 archives with some graphs that explain it far better than I did -> http://mailman.ds9a.nl/pipermail/lartc/2004q4/014519.html and http://mailman.ds9a.nl/pipermail/lartc/2004q4/014568.html

Quoth Jon Morgan <jon.morgan at motors.co.uk> > >We have a 2 port Digium TE220P card, one span is configured to connect to our ISDN30 provider (British Telecom), the other span connects to our internal PBX. Here's the zaptel.conf snip: > >span=1,1,0,ccs,hdb3,crc4 >bchan=1-15 >dchan=16 >bchan=17-31 > >span=2,0,0,ccs,hdb3,crc4 >bchan=32-46 >dchan=47

I have a couple of numbers that are diverted to a number that is conected to an isdn30 card, running asterisk 1.4. eg. 123456 => 22334455 654321 => 22334455 What I would like to know is the number of the orginal number dialled (123456 or 654321). I thought that RDNIS was the answer, but it is always coming up blank. When I did a debug on the pri span, I saw the following message

On how to even search for this "feature" as I have no idea on what it can be. I've got a meridian linked to * (by EuroISDN) which is linked to a ISDN30. I can make calls from the meridian, and receive calls into the meridian. Great stuff. However, if someone dials an invalid number, then instead of hearing a "three tone", the line just drops and goes dead. The console

Hello, I just found the great howto and started shaping my internet connection. The howto''s last update is a liitle in the past now so I have some questions about how things are done the best way nowadays ;-) To ensure a stable and low latency voip communication I added an HFSC qdisc to device ppp0 (1 Mbit SDSL). There are two classes (by now): One for SIP and RTP and one for the

Our users are experiencing some unacceptable delay when trying to have a conversation. The delay is so noticeable that they keep stepping on each others words and resort to calling the customers via cell phone. Here is the setup SDSL Connection (PPPoA) Speedtouch 610 SDSL Modem 3Com 2224PWR Plus Switch (phones on separate VLAN) 8 Cisco 796 Phones All connecting to a remote Asterisk Server. We

Hello, I'm trying to connect a Digium Wildcard TE110P T1/E1 Card to a 10 channel BT ISDN 30 box, however I'm fairly new to Asterisk and this is the first time I've set-up a telephony interface card can anyone give me some pointers or known working configurations? I'm running Asterisk 1.6 and Dahdi 2.4.1.1. Thanks, Tanuki -------------- next part -------------- An HTML attachment

Hi there Just a quick question, I will be building some servers in a lab utilizing Digium E1 cards. I would like if possible to avoid the expense of installing an e1/ISDN30 in my lab. I have two questions really, first does anybody know of an effective simulation tool I can use to replicate a real world PRI but without the telco line being installed. And secondly, can I have a scenario with 2

My employer wants to use Asterisk, but the E1 circuit providing the current phone system is DASS2 rather than ISDN30. Converting the E1 to ISDN30 is not a practical proposition at this time as it'd stop the legacy phone system from working. Is there any sort of hardware support for DASS2? I speculate that the E100P should be able to deal with the electrical side of it, but I'm unsure of