search for: rodan

Are IVR and "Auto Attendant" interchangeable terms? They both do the "Press 1 for" thing. Sales is asking me how to word it and I've always used both terms interchangeably.

Anybody have a guide to the Cisco 79xx phones? One that I can give the 7 or 8 ppl in my office so that they can stop asking me questions. I was going to type up a basic guide but then decided I don't want to reinvent the wheel, one of you may already have one. I tried to use Cisco's guide but it's for their own protocol, a lot of options are different or rearranged. I need a basic

Using FireFly, all other codecs but G711 Ulaw is selected. But whenever I place a call, I get: Jan 24 16:07:06 WARNING[30654495]: dsp.c:1468 ast_dsp_process: Inband DTMF is not supported on codec G.711 u-law. Use RFC2833 Umm, wtf? I thought Inband was ONLY supported on G.711 u-law. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I need advanced Asterisk training in the SouthEast area of the U.S; I don't need to know how to install linux and Asterisk and compile the modules and load them and such. I don't need to know what extensions.conf does or sip.conf does; What I do need is a better understanding of what every single little option in sip.conf or iax.conf does, and I need to learn a lot more about all the

...faster, easier, cheaper to just upgrade your existing workstations? Just a thought. Feel free to contact me off list if you would like. Thanks, Dustin -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Paul Rodan Sent: Friday, January 21, 2005 1:17 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] AstTapi - Crashes w/ Windows 2000 - UrgentHelpneeded - May need to hire a developer No word on this post. Please, can anybody help me? Is there a known...

Is there a way to monitor a call mid-stream? I did look on the Wiki and found that AstGUI can do it, but it's a bit of an overkill. What I want is for a customer service rep, sitting in front of a Cisco 7960, to be able to hit a button (either on their phone, or maybe a specific webpage) that will start recording the call from that point on. I'm thinking the services button on the

Pulled Stable. ( CVS-v1-0-01/12/05-11:43:32) This does compile cleanly. Now I just have to weed out the features from CVS-HEAD that I was using and patch them into STABLE. -brian On Wed, 12 Jan 2005 at 11:37 Paul Rodan (asterisk@glitch.cc) wrote: > Problems with CVS-HEAD, but what about CVS-STABLE? > > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Brian S. > Adelson > Sent: Wednesday, January 12, 20...

I heard Sipura had really awesome documentation on the SPA-2000 and SPA-3000, but you have to email them for it. When I did, they said I had to get it from a reseller. It's been a while since I bought my units, I don't even remember where or who they were bought from. Can somebody email me the documentation for these devices? I'm quite interested in knowing what every one of their 200+

...stnames in host allow/deny list. [2005/10/28 15:20:06, 2] lib/access.c:check_access(324) Allowed connection from (192.168.X.X) [2005/10/28 15:20:06, 3] smbd/process.c:process_smb(1114) Transaction 0 of length 137 [2005/10/28 15:20:06, 2] smbd/reply.c:reply_special(448) netbios connect: name1=RODAN name2=ERIC-AMD-4200X2 [2005/10/28 15:20:06, 2] smbd/reply.c:reply_special(455) netbios connect: local=rodan remote=eric-amd-4200x2, name type = 0 [2005/10/28 15:20:06, 3] smbd/process.c:switch_message(900) switch message SMBnegprot (pid 98080) conn 0x0 [2005/10/28 15:20:06, 3] smbd/sec_ctx.c:se...

We have a customer that handles the billing for a rather large company. Anyway, they have their phone system through us, Cisco 79xx phones with Asterisk and such. They want us to build them an IVR system that can interact with their billing system through XML and read back information to the customer. They want their customers to be able to call in, enter account numbers or credit card numbers

The thing I dislike the most about the 79xx phones is that in DHCP mode, they expect the DHCP server to tell them their TFTP server address. They won't let you set it manually. So if I don't have DHCP server that gives TFTP server info, which is most of the DHCP servers at out there, then the phone won't be able to download any updates made to the SIP000*.cnf file. Using dhcpd on

Ok, I usually use the latest stable CVS, with no patches or modifications. If figured if there was a worthwhile patch, Mark would have already included it. However, there was that neat patch about being able to press a certain key and it'd begin recording in mid-stream, that was an awesome feature and I patched my latest features.c file with that patch. But I keep seeing mentions of other

> -----Original Message----- > From: Paul Rodan [mailto:asterisk@glitch.cc] > I'll ask a stupid question, how does a user hit an alpha > letter from his touchtone? > > I know that the Cisco 7960's support entering alpha letters, > and it could > potentially do it (maybe), but how does the average end user > ente...

Does anybody know of a nice web interface or something that can provide some or all of the following? 1. Create agent phone logins. 2. Create real-time report to monitor agent login/logout activity.? Should have the ability to view which agents are currently logged in/out of the system. 3. Create historical report to pull agent activity.? Should display login/logout activity.? Be able to pull

Hello, I want to build a PBX with the following specs : 1. we have two trunk lines 2. we need upto 8 extensions - all analog phones maybe one voip phone What is hardware that I need and where to find it ? Thanks Varun

...E1's to a DMS300 switch. I mean, i configured sip channels (in and oout) to these gateways, i dont have any special hardware in the asterisk server. thanks -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Paul Rodan Sent: Friday, January 21, 2005 12:51 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] three way call using sip The BT100's do support conferencing, most SIP phones do. But how does your Asterisk connect you to the PSTN? Through a Zap inter...

I know FWD uses a SER/Asterisk combo, and I keep hearing about the massive benefits, however, my initial playing around in SER's configuration indicates it's NOTHING like Asterisk at all, and almost 5x as difficult to understand and configure. But that's only after a few hours of playing with it. I'm interested in learning SER more, especially the integration with Asterisk. Is

Ok, My provider is sending a call to me via ULaw but Asterisk isn't picking up on this, I've only allowed ulaw, I disallow=all and then allow=ulaw in my sip.conf and that's the only thing I allow, but when my provider sends me the requests, I get an error about No Compatible Codecs: 17 headers, 8 lines Using latest request as basis request Sending to 67.19.245.213 : 5060

We're encountering a problem with AstTAPI crashing on Windows 2000 Workstations. The program we're using is called Amicus Attorney, it uses a standard TAPI interface to be able to dial our clients, but on the 2 Windows 2000 workstations we've tried it on it has crashed, no errors or anything. When we select the Asterisk TAPI driver, the whole windows just closes/crashes w/ no apparent

I dont have a cisco acount yet can some bady hel me with the ata18x-v2-16-030401a-1.zip file. thanks in advance Rodney Acosta Coya. Dpto. Tecnologia de la Informacion. racosta@moanickel.com.cu Tel:(53)(24) 62 611 -----Mensaje original----- De: Paul Rodan [mailto:asterisk@glitch.cc] Enviado el: Martes, 23 de Noviembre de 2004 11:24 a.m. Para: 'Asterisk Users Mailing List - Non-Commercial Discussion' Asunto: RE: [Asterisk-Users] ATA186 V2.15.ms You probably aren't running SIP firmware. You'll have to upgrade it first. -----Original...