search for: sparco

...t as to why it is implemented the way it is, please enlighten this pitiful soul. Thanks, Chris [forgive me if i've been posting incomplete thoughts lately, my various project deadlines have been getting to me, i'm mentally exhausted.] -- Christopher L. Wade Unistar-Sparco Computers, Inc. Senior Systems Administrator dba Sparco.com Email: clwade@sparco.com 7089 Ryburn Drive Phone: (901) 872 2272 / (800) 840 8400 Millington, TN 38053 Fax: (901) 872 8482 USA

Hi, I'm am currently in the process of trying to integrate an * box with an NEC Electra Elite IPK. Currently, we have 7 POTS lines coming into our building. These lines are plugged into our NEC using the appropriate analog line interface card from NEC. The NEC effectively has NO configuration done to it, other than to make all the internal phones ring when a call comes in. We also

Hi people, question one i see that asterisk is now in 1.x release. having tried it in the past i want to know if i can use a voice modem as an outgoing line. i know in the past that was not possible/supported so im just asking in case the option is now available. question two im planing to use asterisk as a pure voip solution with sip phones and h323 phones no need for digium/dialogic hardware

...tension (internal, vm824, 3) exited non-zero on 'SIP/824-d2f2' I have logger.conf setup to dump absolutely everything to one file, this also affects the verbose output to that file as well. Any help would be appreciated. Thanks, Chris -- Christopher L. Wade Unistar-Sparco Computers, Inc. Senior Systems Administrator dba Sparco.com Email: clwade@sparco.com 7089 Ryburn Drive Phone: (901) 872 2272 / (800) 840 8400 Millington, TN 38053 Fax: (901) 872 8482 USA

...m table for example. :) I know this solution, thanks to the fact that it is a 'hack', is far from the 99.999% I just stated, but it also seems to be the only low-end solution for poe. Am I right, or just plain blind? Thanks, Chris -- Christopher L. Wade Unistar-Sparco Computers, Inc. Senior Systems Administrator dba Sparco.com Email: clwade@sparco.com 7089 Ryburn Drive Phone: (901) 872 2272 / (800) 840 8400 Millington, TN 38053 Fax: (901) 872 8482 USA

...For background check out bug 0002894 [http://bugs.digium.com/bug_view_page.php?bug_id=0002894]. Enjoy, Chris PS. my C is very rusty, almost gone even, so I give no warranty that this will not crash your entire system. Use at your own risk. -- Christopher L. Wade Unistar-Sparco Computers, Inc. Senior Systems Administrator dba Sparco.com Email: clwade@sparco.com 7089 Ryburn Drive Phone: (901) 872 2272 / (800) 840 8400 Millington, TN 38053 Fax: (901) 872 8482 USA

...700 results in 'From' being the 'Channel' that performed the blind transfer. Any ideas. (writing this just before going home for day, so it'll be tomorrow before I can respond with further details if needed) Thanks, Chris -- Christopher L. Wade Unistar-Sparco Computers, Inc. Senior Systems Administrator dba Sparco.com Email: clwade@sparco.com 7089 Ryburn Drive Phone: (901) 872 2272 / (800) 840 8400 Millington, TN 38053 Fax: (901) 872 8482 USA

Is there a way to ring selective line on multi-line phone. For example if I'm on the phone talking internally on line 1 and the calls comes-in the line 2 will automatically ring. The phone P104 allow extension to be assign each line. Is there a way to call certain line (example line 3) on multi-line phone instead of line 1 when the phone is not busy? For example the Sip phone P104 has

Hi, now I've got caller id working on my BT line in the UK, I'd like to play a different message to those pesky sort who with hold their outgoing number. How can I do this in my extensions.conf for my [incoming-analog] context? I realise some people may call who are unable to change the way that their system withholds the outbound number, so I'll give them chance to leave a voice

Sorry if this is the wrong list... I need a toll-free number to be delivered to me on IAX. (This is NOT an existing number need to buy the whole service.) Anyone know of a service provider offering this? -Mark 707-735-1038

If I have different opening hours on different days, can I accomodate that in a single gotoiftime, or will I need to filter them out one by one ? For example, our hours are Mon-Fri 9:00-17:00 and Sat 09:00-13:00 can this be done something like GotoIfTime([9:00-17:00|mon-fri][9:00-13:00|sat]|*|*?open,s,1) or something like that, or do I have to do:

I'm getting these "calls" at 16 and 46 minutes after every hour. The SIP phone rings, and if we pick up, we get a dial tone. If we don't pick up, we get the dial tone in a voicemail message. An analog phone connected to the incoming POTS line doesn't ring (whether or not * remains connected to the line). It's like the horror movie where the babysitter is getting

...asic idea boils down to this, be able to use the recent aliases and convenience features everywhere they might be needed/wanted (namely, all the different goto's). If this already exists, please hit me with the clue stick ;) Thanks, Chris -- Christopher L. Wade Unistar-Sparco Computers, Inc. Senior Systems Administrator dba Sparco.com Email: clwade@sparco.com 7089 Ryburn Drive Phone: (901) 872 2272 / (800) 840 8400 Millington, TN 38053 Fax: (901) 872 8482 USA

I've been banging my head against a brick wall for the last hour and I'm sure this is one of those easy to solve things - just that I can't see the wood for the trees. I'm trying to do: ----------- [some-context] Exten => s,1,Macro(dodial,'SIP/201,15,tT',123456,MOHClass) [macro-dodial] Exten => s,1,SetCallerID(${ARG2}) Exten => s,2,SetMusicOnHold(${ARG3}) Exten

Hi there, I've seen people using SER with Asterisk. I took a look at SER website, and I didn't see the point in using it, since Asterisk already handles SIP very well (apparently, at least). But, as I'm starting, and some of you (more experienced) use it, I know that there's something there... So I would like to know why to use SER. Is it because of scalability, performance,

Hi all, Has anyone gotten custom ring tones to work using ALERT_INFO with the Cisco 7940 SIP phone? I've read the wiki, but just can't get this to work. I'm currently using the 7.2 SIP image. Thanks, Chris

Hi: Just saw your post while trying to solve a similar asterisk problem. Did not see any responses. Was your problem solved and what was the solution? Carey

I have set up an auto attendant and all is working but I am bothered by a long pause when switching between menus. This pause is between 5 and 7 seconds and is quite annoying. Is there anyway to address this. One other thing I find interesting is that when I move from the main menu to the sub menu the delay is there but when I move from the sub menu to the main menu the delay is not there.

I am currently writing a prototype agent monitoring system, which (as most others in question) simply monitors the output from the event system, and displays the relevant information. I would hope to donate this back to the community once it works properly :) However, I feel that it would be more useful for the manager output to be tab delimited, one record per line: i.e. instead of Event:

Hi all, I know this is a stupid question, but it is one I've been trying answer for quite some time. Exactly how many simultaneous calls can the Cisco 7940 have, considering you can be talking to one, and have XXX others on hold? Using SIP, is XXX only 1? I've found documents in various places indicating different values in regard to the max number of calls the phone can handle.