similar to: Anybody with experience with Aculab Groomer II

Good Afternoon, Can someone please explain what Y-cords are available out there and how they can be used with Aastra or other VoIP phones? Maybe with or WITHOUT headsets? Isn't a Y-cord traded for soft Barge in these days? Thanks, Bruce -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello list, (Resending this email due to a typo in previous copy) I need to do E1 to T1 conversion for a project, and was wondering if there exists a card with both E1 and T1 on it. Or is it possible to use two separate cards in an asterisk box, one for E1 and one for T1? (Please don't mention aculab or adtran, dealt with them in the past, won't deal again.) I talked to Digium and the

>Freddi Hansen wrote: >> with boards from Aculab, we are replacing Aculab boards with Digium >> boards BUT we would need more >> Digium boards IF we could use both Digium and Aculab cards in the same >> server. The reason being that >> TE410P doesn't support SS7-ISUP so we continue using only Aculab cards >> in the servers that must support >>

> > >On Thu, 2003-11-13 at 16:50, Freddi Hansen wrote: > > >>>> >Freddi Hansen wrote: >>> >>> >>>>> >> with boards from Aculab, we are replacing Aculab boards with Digium >>>>> >> boards BUT we would need more >>>>> >> Digium boards IF we could use both Digium and Aculab cards in

Hello Trevor, I wonder how I can find out for sure what is the H/W version for a PROSODY ACULAB SS7 Card? I dought that I have a ver 1.1 which have may issues with recently made computers. I have a case opened for my problem with aculab but sysdiag shows that I have ver 1.1 and aculab says I have ver1.5! can sysdiag be inaccurate? I tried config summary and I had the same result! wish you

Hello list, I need to do E1 to T1 conversion for a project, and was wondering if there exists a card with both E1 and T1 on it. Or is it possible to use two separate cards in an asterisk box, one for E1 and one for T1? (Please don't mention aculab or adtran) Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) -------------- next part -------------- An HTML attachment was

Hello list, What servers would you suggest for:100 concurrent SIP calls, 4xT1 card, and a not much busy website, i.e. getting 500-1000 hits a day. Thanks, Zeeshan A Zakaria -- www.ilovetovoip.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101020/8ab7ae3e/attachment.htm

Hello list, I have been using Cepstral's 8KHz voices for my text-to-speech service for some time now, and have been noticing that the voice quality is really poor, doesn't matter what phrase I give it to convert. None of the other 8KHz voices I have ever used were this bad. It doesn't seem good enough system to be used in a commercial system. Is there any better quality text-to-voice

I hate to bring this thread back to life, but... > it may be possible to get it supported, do you think the price >point is remotely competitive with Digium hardware? Also as I am not >about to divulge my information to them to look in the downloads >section, what is the licensing of their SDK? What is the licensing of >the driver? > >Steven >On Tue, 2002-11-26 at 14:52,

Hello list, I need to know how to deal with a redundant network with only one asterisk server, which is receiving registrations from the end points on both of its ethernet ports. This means extension 201 is registering both from eth0 and from eth1. Is there a way/software which can act as a middle man between asterisk and the ethernet ports, and by default sends registrations to asterisk only

Hi list, I am planning a migration to virtual machines, and was considering with it to move from 1.4 to one of the later versions. My and my clients' 1.4 setups have been rock solid and I don't want to put myself into any unnecessary trouble. Those of you with solid experience with all these versions, what would you suggest? What new and exciting enhancements would newer versions bring

Hi list, For a client I am setting up a system which will use T1 PRI from Primus, who offer only NI-1 and NI-2 protocols for D-Channels. Previousely I have only used switchtypes euroISDN and National. Although the documentation says Asterisk does support NI-1 ans NI-2, but wanted to get your opinion if you have used these protocols on an Asterisk box and if there were any things to consider. If

Hi list, For a telecom project I need to setup a SCADA solution. I don't have any previous experience in this type of monitoring and automization. I'll be using SNMP data from asterisk servers and endpoints. If anybody has any suggestion which SCADA software can fit in such a VoIP solution, your guidance will be highly appreciated. Thanks, Zeeshan A Zakaria -- www.ilovetovoip.com

My main asterisk server is under unusual heavy attack, and so far Fail2Ban has blocked about 30 IPs, from various different countries. At this time it is blocking about 1 IP address every few minutes. Just wondering if anybody else is also experiencing unusually increased hack attempts today? Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) -------------- next part

Hello list, Slightly off the list topic, but I hope I'll get some help here. Somebody wants me to implement for his project a Cisco based VoIP system. I told him that I specialize in Asterisk based systems, but he is not even aware of Asterisk. The requirement of project is such that chances are slim that this firm will consider Asterisk based system. So I told him that though not experienced

Greetings, Sounds like a simple thing to do, but I was not able to do it on these particular phones. Snom wiki was not helpful. My client wants to keep his phones pointed to UTC time, no DST, no change in timezone, i.e. to stay at 0 hours difference. The phones are provisioned from a tftp server. If I remove 'dst' value from the provisioning file, on bootup phones force users to pickup

Greetings to one and all on this fine list; We have the current system: Meridian 1 Option 11 +-------------------+ | | ISDN/30 (DASS/2) ===> |NTAK79BB (2MB Pri) | | |<-->4x16 port Digital / 1x16 port Analogue ISDN/30 (EUROIDSN) ===> |NTBK50AA (2MB Pri)

Hi List, We're planning to use Asterisk as our backend PBX for our legacy PBX where-in received calls from legacy PBX can be transferred to Asterisk PBX extension, is this possible? Regards, Malvin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100712/1bfe3cd2/attachment.htm

Only 100? We had a single server over 300. From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Zeeshan Zakaria Sent: Saturday, October 30, 2010 9:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Under heavy attack My count has reached 100 for the day. The server serves doesn't serve

Hi, Does anyone know where this is suddenly coming from? -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected Thanks Dan p.s. sorry about the last post. hit the mouse by mistake and it sent the email. -------------- next part