similar to: TE110P without router ???

I am going to buy some IP phones from them but I sent them an email couple of weeks ago and got no reply. Has anyone ordered anything from them? Any other places that I can buy from? Sorry if it's a wrong post. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050517/ba256d25/attachment.htm

My router (1605R) currently does not support QoS. Is there any open source software available so that I can set one up before the router? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050118/38b6e3f2/attachment.htm

Hi, I am thinking of signing up with voice pulse connect to connect to my asterisk server and using it as a regular line. Is it good? Or should I go with vonage or others ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050124/16792f10/attachment.htm

I am thinking of dumping broadvoice so I need some other VoIP providers that have a las vegas DID and a service better than broadvoice. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050131/bf4b1c31/attachment.htm

I finally got my incoming and outgoing to work on Broadvoice with a configuration file that is no where close to the one given by them. Here it Is (sip.conf). For others who have a working config could u please share so that I could compare. Thank You [broadvoice] type=friend username=[number] fromuser=[number] secret=[password] host=sip.broadvoice.com fromdomain=sip.broadvoice.com

I have an Asterisk server that was running fine until Sunday. Monday there was a power outage and the server was off most of the day. This server has a TE110P, two TDM04B and an Astribank 32. Today I noticed that the TE110P started having IRQ missess. Before today it only had about two or three per month. Today, this is the miss count after ten minutes on: IRQ Misses: 1443

Is the Broadvoice service up? I just signed up with them and started receiving calls in no time but could not make calls. And after a few minutes I cannot even place calls. register => [number]:[password]@sip.broadvoice.com [broadvoice] type=peer fromuser=[number] host=proxy.lax.broadvoice.com fromdomain=sip.broadvoice.com context=from-broadvoice dtmfmode=inband any help would be

Hi, We are looking towards a 8+ CO line setup (20 extensions) in our office but we do not want an IVR(auto-attendant) feature. All incoming will be answered by a receptionist. I have read the multi-line configuration for cisco 7960 thread in this list but that way I believe we could only display 6 incoming lines. What will happen to the rest? Does the expansion module for the cisco 7960 work

I believe the 911 is a serious issue if one does an asterisk installation in an office. How do you test 911? Won't they arrest you or something for dialing 911 for no reason and talking to one of their agents who could have taken a more important call? On the other hand what an emergency comes up (like someone got seriously injured) and on top of that asterisk crashed all of a sudden

Dear users. My current setup uses a euroISDN E1 with 8 cahnnels for incoming and outgoing calls from a digium te110p. Currently all phones use SIP. However, I need to add some faxes lines and some POS credit card machines. These will require POTS lines with a fixed DDI. I have purchased the tdm400p and 4 FXS modules. My problem is with the zaptel.conf and zapata.conf. I am a little confused as

Which would you recommend as far and quality and pricing to connect to asterisk (including DTMF issues)/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050125/bf7bf081/attachment.htm

Is there any way to change the outgoing caller id on BroadVoice I have tried SetCallerID(Name <Number>) but that does not work Thanx -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050126/bbd95fcf/attachment.htm

I have been looking around for Outlook Integration for Asterisk. Saw the Asterisk TAPI wiki page and also ran across this: http://www.fonality.com/pop.cgi?page=pop_pbxtray.tt (PBXtray) It looks like Fonality has managed to make an app that does screen pops and allows click to dial. Has anyone else been able to get this all to work successfully? Looks pretty slick.

Hi list, I have a terrible noise issue with Dell SC1430 + Digium TE110P. The digium card is not sharing interrupts with any other device, as I saw in Dell's BIOS and also with "lspci -vb" command. After changing coax wire, UTP, balum, digium card ... I have found that the problem is in Dell box, so now I'm running the same Asterisk config in other server with the same

Guys. I have an issue with a te110p card and also some tdm04b cards on the same system: Zttest returns this for the tdm04b cards: [root@mollendo ~]# /usr/src/zaptel-1.2.4/zttest 38 -v Opened pseudo zap interface, measuring accuracy... 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8192 sample intervals 100.000000% 8192 samples in 8192 sample intervals 100.000000% 8192 samples

Hello Everyone, I checked with zttool that sometimes after the machine boots the order of the boards is changed like this: ??? Alarms Span ??? OK Digium Wildcard TE110P T1/E1 Card 0 ??? OK Digium Wildcard TE110P

Just got a headset for testing asterisk and am using X-Lite. I plugged in the headset into the headset jack and is there any way to configure X-lite to use the headset instead of the speakers? Or will I have to plug the headset in the speaker jack ? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, I have configure my Asterisk as follow (using Asterisk@home): [zaptel.conf] span=1,1,0,ccs,hdb3,yellow bchan=1-15,17-31 dchan=16 loadzone = uk defaultzone=uk [zapata.conf] [channels] switchtype=euroisdn pridialplan=local prilocaldialplan=local internationalprefix=00 nationalprefix=0 usecallingpres=yes busydetect=no ; not need on pri callprogress=no ; was yes but wiki says experimatley

Hi ! Did anyone had issues/managed to solve issues with DISA over Zap channels on * 1.0.X (STABLE) ? I have a situatuion where DTMFs that should be recognized in DISA work over SIP channels and do not work over ZAP channels (Zap channels are on TE110P) I have in default context: exten=> 299,1,DISA(no-password|default) and I have SIP extension 200 in [default] and I have Zap trunk which

Hi folks, I've already searched the mailing list but no one else seems to have my same problem. I'm using Asterisk with the following configuration: Fedora Core 4 (but I also tried Fedora 3) 1 Digium TE110P 1 TDM40B 1 HFC-S 'Cologne' bristuff 0.2.0-RC8o (zaptel 1.0.9.2) I compiled right, I can load kernel modules but when I try to unload wcte11xp module (the one