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...y's they are interested in selling? I am looking to pick one up cheap for a proof of concept before going all out on them. Also does any one have any real life practical experience with how well (or not so well) that these devices have worked for them? you can reply to me off list at asterisk@matraex.com Thanks Michael Blood -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040513/f1507e19/attachment.htm

I have an installation which is currenly using 14 DID Trunk Lines. I need to be able to use Caller ID information and currently it is not available on these lines. Is there another way to access this information? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040105/62559e22/attachment.htm

Anybody ever run into a case where the Sipura Dial Plan will not work with the S0 option to immediately connect? My Dial plan reads (*xx|[3469]11S0|0|00|[2-9]xxxxxxS0|1xxx[2-9]xxxxxxS0) and I can dial ONLY then numbers in the dial plan so I know that it works. For some reason when I dial 5551212 1212121212 It does not dial for a while and then it dials 555 1212 Anyone have any ideas?

Does anyone know of a resource for extensions in which the server (whether asterisk or custom scripts) can trigger the phone to be answered? So for example an operator can have a headset and when a call comes through the call is automatically (through a script) connected to the headset instead of the operator having to manually answer the call. Any responses, help or ideas of a type of supplier

I have benn having trouble with the Monitor Command. Basically any time that I send a call into a MeetMe room I am only able to monitor half of the conversation. File-in is recorded with the incoming voice but file-out does NOT record anything. I have tried this with both the b and m option as well as without any options to the MeetMe command. Also the Monitor correctly records both sides of the

Does anyone know of a piece of hardware that can allow multiple earpices to be connected directly to a server running Asterisk. I hope I am not being to vague but basically I am looking to allow a call center to user the server to do all of the "Pickup" and "Hangup" functions. The operators will merely have to have th earpiece in their ear. I have seen serial pieces of

Hi, Is there IAX client in Applet JAVA which can be embeded in a web page ? Best regards Rattana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031208/c388ef61/attachment.htm

Hello all, I am building a software based on asterisk to handle incoming answering service calls. I have one problem that I have not been able to figure out a reasonably priced solution to: The goal of this software is to allow the agent to be able to do their entire job from the desktop. The only thing that seems to be a problem is getting the operator (agents) headset logged on to the

...e currently use asterisk in an answering service? What kind of phone/headset/call connection system do you use? Thanks Michael Blood -----Original Message----- From: Philipp von Klitzing [mailto:klitzing@pool.informatik.rwth-aachen.de] Sent: Wednesday, June 30, 2004 4:58 PM To: Michael Blood, Matraex, Inc. Subject: Re: [Asterisk-Users] Answering Service Agent Auto Login Hi! > So... a phone with auto answer COULD work if we could find an > inexpensive enough one (less than $150 would be okay) any suggestions > would be great. The Grandstream phones now have that feature. You'd...

...ple configurations or would any one be able offer consulting services to assist me in setting this up? We have a short timeframe so we are wanting to get this implemented quickly and we are willing to pay someone to help. Please contact me off list if you can help Thanks Michael Blood Michael@matraex.com 208.344.1115 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041203/ad010fc3/attachment.htm

I am writing a call center application. I do not want to use Queues to manage my incoming calls and connect them to the operators for a few reasons which I wont go into here. The option I come up with is to create a context that the call goes to which runs background() and just loops to play it again and again forever. The background() will have options to dial 1 to leave voicemail Then, when

I have been trying to figure a way to SendDTMF into a MeetMe room using the Manager API. I can't redirect everyone into another context and then bring them back because that would mess up my logic. I am trying to use local channels and the originate Action to accomplish this. Exten: 3441115 Priority: 1 ActionID: actid-00000001 Context: senddtmftones Action: Originate Channel:

...bling APCI in my kernel (also enabling my motherboard chipset in the kernel). After doing this I got 20x the Hard Drive write speed, no interrupts are shared, and the error is gone. span=1,0,0,esf,b8zs Message: 1 Date: Wed, 29 Jun 2005 07:13:29 -0600 From: "Michael Blood" <Michael@Matraex.com> Subject: [Asterisk-Users] PRI got event: HDLC Abort (6) on Primary D-channel of span 1 To: <asterisk-users@lists.digium.com> Message-ID: <018a01c57cac$5a6959c0$0f00a8c0@cyprus> Content-Type: text/plain; charset="us-ascii" I receive this error on the asterisk console...

I receive this error on the asterisk console and it is pretty much ALWAYS coming up. Sometimes there will be a break where it does not display. We had our PRI provider test the lines and they claim that there is no signalling problem. It doesn't matter if there are no calls or if there are 10 calls in progress the error is still displayed. I also get an annoying popping or clicking sound

Does anyone know of an ATA which has a modifieable (open source) firmware AND that supports faxing. Granted most of them don't do faxing well but that is more of a problem with the network side. Michael Blood -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050831/8c13db6b/attachment.htm

I have been trying to get rsync with ssh to work and have found that even though I have a daemon running it spawns another when I connect (running as the user I sshed as). This was in the enhancements for 2.5.6. I am trying to figure out how i can use the ssh for security but still connect to the daemon that I have running as root so i can back up all of my files that arn't readable by anyone