Re: [Asterisk-Users] Problem - Adtran TSU 600, t100p
On Tue, 25 May 2004, Bartosz Jozwiak wrote: Hello, I have just received Adtran TSU 600 with 24 FXS ports. I have installed sucessfuly T100P card. Sucessfully? Did you load the module for the card? What does 'ztcfg -v' show? Is asterisk running? Does asterisk see the ports? (zap show channels) Adtran is connected to t100p with crossover T1 cable. On T100P card I have a green light and on Adtran I do not get any errors or alarms. But I do not get dialtone on FXS ports. Adtran is configured: For Network Timing, fxs ports ore fxs_ls on Adtran. In zaptel.conf: snip Good luck, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FYI: Cisco firmware 7.1 released
Cisco has version 7.1 of their SIP firmware for the 79x0 phones. They advertise no new software features, but it does include bugfixes for a number of things. I know there was a discussion about the 0.4sec delay, which is said to be resolved in this firmware (CSCed48311: Media takes 0.4 sec to be set up) Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Channelized T1, SIP phones, HW Echo Canceller
I have a channelized T1 coming in from our telco, terminated onto a TE405. There are three channelbanks serving internal analog extensions, and about 10 Cisco 7960s. I have no reports of echo on the analog extensions (as expected). The 7960 users complain of occasional echo (seems like 1 in 5 calls). Only the SIP user hears the echo, not the caller. I have echocancel=yes, echotraining=yes, echocancelwhenbridged=yes. Changes in the taps of echotraining have made things worse, so I have left it alone. I have backed the txgain down, as audio going out on the telco T1 is really hot. Even at -6dB gain, it is still notably louder from outside than other audio (comparing the ring generated by the telco when calling into asterisk with the ring generated by asterisk calling a station from the auto-attendant). If I drop gain to anything less that -6, I lose all audio. Would a hardware echo canceller deal with this type of echo? My understanding is that it is a result of sip being non-realtime and introducing latency (the latency being half the difference from the original utterance and the echo). Is this correct, or do I have it all wrong? From my studying of the list archives on this subject, it seems that there is no answer for Why is it so intermittent, other than to say that the problem originates somewhere in the two-wire system of the remote party. Is that correct? Has anyone heard of any kind of contraption to use just a single Tellabs card outside of the chassis? If possible, I'd like to avoid the cabling mess of a full tellabs chassis just to use one card. I have looked for a single-card chassis, but with no luck. Any pointers? Many thanks, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] problems with analog interface to PBX
On Wed, 12 May 2004, Dan Fernandez wrote: Folks, For the last few days I've been trying to experiment with a Panasonic PBX and an X100P but have run into quite a few problems which I am not sure if they can be solved with this type of card (how about TDM01B?) 1) I wanted to use *'s IVR capabilities, so I routed the calls to the extension where the x100p was connected to. Asterisk should answer the call, playback a message, dial another PBX extension and if no one answers dial another extension (via IAX). The first problem I ran into was that the Flash application doesn't really work. To get around this I added another x100p to dial the new extension. The problem I ran here was that even though I specified in the Dial app to just dial for 30 seconds, it rang forever as if * cannot recongnize that no one had picked up. Asterisk does seem to detect hangups and busy tones (I have busydetect=yes and busycount=10) For about 6 months, we were using the same logical setup (a channelbank of FXO cards for a Merlin Legend switch, with asterisk doing incoming IVR / autoattendant, then transferring the calls out to the Legend, and handling voicemail). The first problem I encountered that I hadn't expected had to do with asterisk transferring the call back to the Legend. I did a Flash(), a SendDTMF(), and another Flash() - the Legend saw this as an attended transfer, and it caused some oddities. Turns out I needed to Flash(), SendDTMF(), Hangup(). Along the way, I found the Flash times that the legend was expecting to see, and adjusted them in the source code, so as to eliminate occasional flash detection problems. I'd take time to plug an analog set into the extension you have the X100P on, and make sure you can flash/transfer calls like you're expecting asterisk to. There's no reason (that I know of) that your flash can't give you exactly the behavior you're looking for. Good luck to you, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zap channels stuck in 'Rsrvd' state
I have two Adtran 750's connecting our analog phones to asterisk. On occasion, I get a channel that gets stuck off hook. 'show channels' shows: Zap/27-1 (longdistance s 1 ) Rsrvd (None) (None) And will just stay like that until the phone is manually picked up and hung up again (or asterisk is stopped/started). I guess this is a function of an unclean hangup (being read as a flash instead of a hangup?). A 'soft hangup zap/27-1' will not do anything (though it makes an attempt). Does shortening the rxflash time sound like it may help this? (Does anyone have a good explanation, or link to one, of the prewink, wink, preflash, flash, start, rxwink, rxflash, debounce timing functions?) Thanks, as always... Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 SIP Images
I had a completely different experience. The day I decided I wanted to get a contract, I called Cisco, gave them my personal credit card, and three hours later had my CCO access upgraded. I just bought a smartnet for one phone for two years (a whopping $16), there was nothing to it. I've never been contacted by a sales rep (as a result of this purchase). I had an issue with the firmware not functioning properly - inside of two weeks, they had released a new firmware version resolving that problem and a few others. I'm not sure why the experiences would have been so different, but they are. Steve On Mon, 29 Mar 2004, Terence Parker wrote: I think John's said it all - I have absolutely nothing to add! I'm just posting to second his opinion. Terence On 29 Mar 04, at 3:22 AM, John Baker wrote: -- snip -- I finally got ahold of someone at Cisco to sell me the support contract, but it took three weeks and a couple of follow up phone calls for them to process the paperwork and assign me a number. You'd think Cisco would have an easy sign up over the web for this stuff, but no. You've got to send them a check (Why wouldn't you take a credit card???) and answer a barrage of questions before you get the thing. I wondered why a company like Cisco would make you jump through so many hoops. I soon got my answer: one of their sales reps called within days to discuss purchasing more product. I'd be glad to talk to you about it, I told him, but we're a bit premature. I need to evaluate your phone with a current image and I'm getting nowhere with your technical support. Any chance you could speed up the process? It might help you get more business... No chance. After three weeks worth of runaround, I finally got my SIP image. Again the phone was nice, but the service wasn't. The price definitely wasn't. Oh, and let's not forget about the software license requirement and the power cube (purchased separately of course) Add all that up and you're paying alot for what you're getting. I went with the Polycom phones and never looked back. They're every bit as nice as the Cisco phones for a lot less money. John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Steve Creel[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Adtran TA750, any chance of working MWI ?
After some sleep, google gave me some additional information: Wade said before that MWI is done by FSK and voltage-type MWI is not supported: http://lists.digium.com/pipermail/asterisk-users/2003-August/018426.html For the existing MWI to work, voltage on the line needs to drop for a fraction of a second - I think this can happen in asterisk, probably as part of the do_monitor block where vmwi_generate is called. Does that make any sense? We have about 40 2500 sets... If I have to replace the phones themselves, does anyone have a model they'd suggest for replacing an Avaya 2500 set that supports FSK MWI? Steve On Fri, 26 Mar 2004, Steve Creel wrote: I have L36, and Onhook Messaging is enabled. Does anyone have a reference for MWI (other than that stuff that turns up on google)? Make sure that Onhook Messaging is enabled on the Adtran FXS ports. I'd also suggest upgrading to the latest 750 firmware (L36) as it fixes some specific MWI issues. -wade I have a bunch of existing ATT/Lucent/Avaya 2500YMGP sets with LED message waiting lights. Do I stand any chance of getting the Adtrans to light these? What I know so far: When you pick up the telephone, the LED flashes. If you plug two telephones in, picking up one flashes the LED on the other. Hanging the telephone up will flash the LED. Incoming calls flash the LED. Stutter dialtone is there and functioning. Debugs in chan_zap reveal that do_monitor sees there are messages, and tries to update MWI status. The Adtran is putting out 50 volts when on hook (either with or without a message waiting). If 'On Hook Messaging' is enabled (I think this is for ADSI ?), the voltage is closer to 40v. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Adtran TA750, any chance of working MWI ?
I have a bunch of existing ATT/Lucent/Avaya 2500YMGP sets with LED message waiting lights. Do I stand any chance of getting the Adtrans to light these? What I know so far: When you pick up the telephone, the LED flashes. If you plug two telephones in, picking up one flashes the LED on the other. Hanging the telephone up will flash the LED. Incoming calls flash the LED. Stutter dialtone is there and functioning. Debugs in chan_zap reveal that do_monitor sees there are messages, and tries to update MWI status. The Adtran is putting out 50 volts when on hook (either with or without a message waiting). If 'On Hook Messaging' is enabled (I think this is for ADSI ?), the voltage is closer to 40v. I'm at a loss as to what to do, could anyone offer some pointers? Many thanks, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Adtran TA750, any chance of working MWI ?
I have L36, and Onhook Messaging is enabled. Does anyone have a reference for MWI (other than that stuff that turns up on google)? Sorry for not including the firmware version Steve On Fri, 26 Mar 2004, Wade J. Weppler wrote: Make sure that Onhook Messaging is enabled on the Adtran FXS ports. I'd also suggest upgrading to the latest 750 firmware (L36) as it fixes some specific MWI issues. -wade -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Creel Sent: Friday, March 26, 2004 1:33 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Adtran TA750, any chance of working MWI ? I have a bunch of existing ATT/Lucent/Avaya 2500YMGP sets with LED message waiting lights. Do I stand any chance of getting the Adtrans to light these? What I know so far: When you pick up the telephone, the LED flashes. If you plug two telephones in, picking up one flashes the LED on the other. Hanging the telephone up will flash the LED. Incoming calls flash the LED. Stutter dialtone is there and functioning. Debugs in chan_zap reveal that do_monitor sees there are messages, and tries to update MWI status. The Adtran is putting out 50 volts when on hook (either with or without a message waiting). If 'On Hook Messaging' is enabled (I think this is for ADSI ?), the voltage is closer to 40v. I'm at a loss as to what to do, could anyone offer some pointers? Many thanks, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Steve Creel[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fuse for Adtran 750 PSU
On Fri, 19 Mar 2004, Jacques Leisy wrote: Sorry for a very stupid question, but I cannot find a supplier anywhere. Where can I buy the 3 Amps GMT fuses for the Adtran's PSU. Car fuse don't seems to fit. What is GTM the abbreviation of A good question (that I wish had been in the archives when I went looking). You need a 3 amp GMT fuse. Datasheet: http://www.bussmann.com/library/bifs/5008.pdf I bought a couple (and probably overpaid - $3.18 ea) at: http://www.newark.com/NewarkWebCommerce/newark/en_US/support/catalog/productDetail.jsp?id=02B3398 I think I saw them somewhere else for alot less, just don't remember where. Good luck, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Schools/Districts using asterisk?
On Thu, 18 Mar 2004, Chris Hobbs wrote: I'm investigating asterisk to use as a replacement for an aging Lucent PBX in our district office, as well as replacing the Centrex/intercom based systems at our schools. I'm curious if any other schools/districts are using asterisk? If so, I'd certainly be interested in talking about your setup. Chris, We're in the process (final phase to be completed in the next two weeks) of replacing our ATT/Lucent/Avaya solution at our high school. Over the summer, we hope to integrate/replace the PA/intercom system (Teltrend IV) to extend asterisk dialtone to the classrooms. Contact me off list and I'd be more than happy to talk with you about our progress, plans, and experiences. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] local VoIP in Florida
On Wed, 17 Mar 2004, Tim Sailer wrote: On Wed, Mar 17, 2004 at 08:54:57PM -0600, Matthew Marlowe wrote: 727 or 772? There is 772 in FL available. 727. That's St. Pete/Clearwater. What area is 772? Tim http://www.nanpa.com/area_code_maps/display.html?fl shows the Florida area codes... For the archives: The NANP is the numbering plan for the Public Switched Telephone Network for Canada, the US and its territories, and the Caribbean. For information on North American phone numbering, see NANPA website at http://www.nanpa.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 and short delay before voice starts after ring.
On Wed, 10 Mar 2004, John Fraizer wrote: For what it's worth, I don't have any delay between answer and audio with my asterisk server and 7960G either originating or answering. It doesn't matter if it's a call to/from another SIP/IAX device or to/from PSTN. It's pretty much instant (not detectable by humans at least). So, there may be some truth to the fact that the delay is caused by the Asterisk install in your case. There are so many variables that it is very hard to tell but, since I don't see the delay, I am leaning towards it being an Asterisk implementation issue. Can you test this with an extension that goes into VoiceMailMain(). My 7960 and 7960G phones both get the first couple letters of Commedian Mail cut off (usually ...median Mail). Just trying to quantify the delay we're talking about... Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Phone with large display
On Tue, 9 Mar 2004, Jonathan Moore wrote: I have seen conflicting references in regard to this. Seems like the Cisco site has comparison charts that show this phone doesn't have a SIP image, but after seeing the post I did a little searching and people seem to have a way of running SIP on them. So how is it done? Cisco has a SIP image for it, you can download it directly from them. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Passthrough
On Tue, 2 Mar 2004, Emanuele Laface wrote: My actual pbx is connected to the external world with 2 PRI interface, my idea is to insert asterisk in the middle, I want disconnect the two PRI, connect them to the asterisk and connect the asterisk with old pbx with a cross cable. So, at the first step, my asterisk is simple a passthrough, but in the future I can change smoothly all my office phone and finally I can disconnect the old pbx. You've got two options here - you can use dacs in /etc/zaptel.conf to literally just cross-connect the PRIs. The two telco PRIs would come in on two of your ports, and would turn around and go back out on the other two. This happens in the zaptel module and doesn't make it up into asterisk. Your other option is to terminate the two PRIs into asterisk, and use asterisk to provide two PRIs into your PBX. This gives you access to the actual call routing. Ok, I'm at the first step, I have two problem: - First problem: what is the configuration of asterisk for passthrough? I have a good knowledge about SIP, IAX, and asterisk in general, I have build a working configuration with SIP phones (Grandstream Budget One) and asterisk with a PRI, but I don't know how I can configure the zaptel card for passthrough. - Second problem (is not a real problem): where I can find the diagram for a cross cable for PRI-to-PRI connection. If you're looking for a cable to go from the TE410P to your existing phone switch, you need a T1 crossover. Jared Smith has a good chart: http://www.jaredsmith.net/misc/cables/ Good luck, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Passthrough
On Tue, 2 Mar 2004, Andrew Kohlsmith wrote: You've got two options here - you can use dacs in /etc/zaptel.conf to literally just cross-connect the PRIs. The two telco PRIs would come in on two of your ports, and would turn around and go back out on the other two. This happens in the zaptel module and doesn't make it up into asterisk. Can you elaborate on this? I have no mention of the term 'dacs' in /etc/zaptel.conf. According to asterisk-cvs, on October 30, 2003, dacs support was added: Add DACS functionality to zaptel for cross connecting channels zaptel.conf.sample is appropriately documented: dacs The zaptel driver cross connects the channels starting at the channel number listed at the end, after a colon If I wanted to cross connect the first span to the second: dacs = 1-24:25 If I want to cross connect just channel 3 to channel 27: dacs = 3:27 I imagine (though haven't tried it), you can use: dacs = 1,3-5:25 to take channels 1,3,4,5 and put them on 25,26,27,28 One note: you can only use dacs on T1/E1 spans, not the pci fxs/fxo cards. Hope that helps... Steve ___ Steve Creel[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Passthrough
On Tue, 2 Mar 2004, Emanuele Laface wrote: On Tue, 2 Mar 2004, Steve Creel wrote: Your other option is to terminate the two PRIs into asterisk, and use asterisk to provide two PRIs into your PBX. This gives you access to the actual call routing. Ok, my problem is exactly how I can do that? How can I say to asterisk this port is connected to the world and the other is connected to my office telephon switch (this is my main problem, I see something about groups but I'm not sure about the right configuration...)? Don't try to map port to port - you're making your problem more complex than it needs to be. Let asterisk do some call routing for you. You've got an incoming call with dialed number identification. Write an asterisk extension rule to handle it... Should asterisk send it out on a specific channel? Should it be sent out to one channel out of a group? For example, say your incoming number is 12345. You've connected the telco PRIs to spans 1 and 2, and your PRIs to the existing PBX are spans 3 and 4. The telco channels are all in group 1, the channels to the PBX are in group 2. [incoming] exten = 12345,1,Dial(ZAP/g2) ; Send incoming call for 12345 to the PBX Now you have asterisk switching the call instead of cross connecting the ports. How I can forward a call? It's simply an extension.conf rule? Yes. When I make the forward in this way (with extension.conf rule) asterisk make some work or is a simple passthrough from interfaces? Yes, it's some switching/callsetup work, but no codec translation, which is by far your biggest CPU consumer. I need that calls from PRI to PRI don't load the computer. I want to use all CPU to (future) SIP calls. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Simple Front Gate Intercom
On Thu, 26 Feb 2004, Greg Kedrovsky wrote: On Thu, Feb 26, 2004 at 12:16:06PM -0800, TC wrote: http://www.vikingelectronics.com/products/app-notes/doorboxes.html The W-1000, W-2000A and W-3000 doorboxes are designed to be installed on the unused telephone line input of nearly any phone system or... Key word: input. My telephone line input is my x100p fxo card, and it is a ONE-port card. I have no unused line input on my phone system. Therefore, I'm hosed with these models??? Help me... I've fallen, and I can't get up... The W-1000, W-2000A and W-3000 type devices connect diretly to a phone (or an X101P) The E-10 and E-20A type devices connect to a CO line (provided by telco or TDM400P). It's a question of the application - the latter of the two is just a speakerphone (what you're looking for). ___ Steve Creel[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Simple Front Gate Intercom
On Thu, 26 Feb 2004, Greg Kedrovsky wrote: In a nutshell: Can I use Asterisk to hook up an intercom at my front gate? My wife would like to have one of those simple speaker/microphone intercoms. People show up at our front gate, press the doorbell, it rings in the house. We pick up a phone on my Asterisk system and dial (example) 105 to connect to the intercom and say, Who's there? The dipweed at the gate leans forward, and speaks into the same speaker he heard the voice come from and says, Joe Sixpack. snip Is this configurable in Asterisk? It seems to me that an intercom like that would just be the same thing as an open phone line, no dial tone. Then, we dial 105 (or whatever) to get the gate-com, and it opens the line with the intercom to speak with the person wanting in. Greg, As for hardware, take a look at: http://www.vikingelectronics.com/products/doorentry/product_list.html Sites like: http://www.wantphones.com/v-door-entry.html http://www.globe-mart.com/index/c/communications_intercom_systems_viking_.htm and http://www.phonemerchants.com/secprod.html (from google search results) sell them and have prices listed. These phones come in both FXO and FXS devices. The FXO will be the autodialer type, or those capable of using an automatic ringdown line. Connect these to your TDM400P. The FXS will be those that don't address autodialing or ringdown circuits. They will mention the REN (ringer equivalence number) and require a power supply. I would suggest you setup your door phone as an immediate=yes line. As soon as they push the button on the door phone, it starts ringing phones in the house (maybe w/ a distinctive ring?). You answer the phone and get to talk to them (without dialing). Best of luck to you, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Simple Front Gate Intercom
On Thu, 26 Feb 2004, Rob Fugina wrote: On Thu, Feb 26, 2004 at 10:19:08AM -0600, Greg Kedrovsky wrote: On Thu, Feb 26, 2004 at 10:28:37AM -0500, Steve Creel wrote: As for hardware, take a look at: http://www.vikingelectronics.com/products/doorentry/product_list.html Nice. Thanks. I was unaware of this hardware. It looks like something similar to the Viking W-1000 would work perfectly. Press the Call button on the intercom, and it rings phones inside. I can hook it up to a dedicated line on my Asterisk PBX system, give it a distictive ring, and we're set. Am I getting things backwards, or is the W-1000 what he was talking about as an FXO device. I'm having trouble finding an FXS version on Viking's site at the moment... A bad explanation on my part... The W-1000A provides battery and generates ring (you plug a phone into it). The E-10 receives battery (you plug a phone line into it). Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 SIP image (off-topic)
On Thu, 19 Feb 2004, Chris Hirsch wrote: Bisker, Scott (7805) wrote: Buy SmartNet support for the phone. That grants you access to software images through their website. Try Insight. 1-800-INSIGHT. They sell all quantities. Given than I'm interested in getting a Cisco phone off something like ebay what does it cost to get an image? If only there was some way you could check to see if this had been asked before... http://www.google.com/search?q=site:lists.digium.com+cisco+smartnet+7960 There are varying stories about the contract. My personal experience (http://lists.digium.com/pipermail/asterisk-users/2003-September/020541.html) was with a used 7960 off of eBay. The contract cost me $8/yr. Someone else had a similar experience (http://lists.digium.com/pipermail/asterisk-users/2003-December/031401.html). Others have mentioned various other prices as high as $80 (http://lists.digium.com/pipermail/asterisk-users/2003-December/031240.html). Apparently, your mileage will vary... Best of luck, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Round-robin chan_zap groups...
I've not seen it documented anywhere, but scrolled past it the other day in chan_zap.c. Apparently you can specify a zap group with an 'r' instead of a 'g' to use the group in round-robin. I looked but didn't find anything in the archives on this, so I figured I'd mention it. Steve ___ Steve Creel[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] merlin legend / * as ld gw
On Sun, 15 Feb 2004, Chris Clifton wrote: Can anyone offer adivce for connecting * to a merlin legend ? I'd like to use a t1 interface to connect the two, * will be used as a long distance voip gateway in this scenario. Is this possible using a digium t100p ? Is it possible? Absolutely. You'll find this to be as straightforward as it sounds - if you run into trouble, it will likely be things like remembering (or not) to use a t1 crossover cable, proper timing, and other things of that nature. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_local and variables
On Sat, 14 Feb 2004, Philipp von Klitzing wrote: Hi! We need to implement the following: Call comes in, ring ZAP/1 (6 rings) For the last two rings, also ring ZAP/2 [incoming] exten = s,1,DIAL(Local/[EMAIL PROTECTED] Local/[EMAIL PROTECTED],18) [test1] exten = 123,1,Dial(ZAP/1) exten = 124,1,Wait(12) exten = 124,2,Dial(ZAP/2) Why not simply use this instead: [incoming] exten = s,1,DIAL(ZAP/1,12) exten = s,2,DIAL(ZAP/1ZAP/2,6) Philipp For SIP phones (and analog phones w/ callerid), that would show two missed calls on ZAP/1 for every incoming call. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_local and variables
We need to implement the following:
Call comes in, ring ZAP/1 (6 rings)
For the last two rings, also ring ZAP/2
I have the following (which works as expected):
[incoming]
exten = s,1,DIAL(Local/[EMAIL PROTECTED] Local/[EMAIL PROTECTED],18)
[test1]
exten = 123,1,Dial(ZAP/1)
exten = 124,1,Wait(12)
exten = 124,2,Dial(ZAP/2)
I can't figure out how to back this into a macro. I would like to use the
setup below, but it seems impossible to pass variables down into the local
channel. Can anyone confirm this, or suggest some alternative? (I've
tried the /n on the chan_local, with no success)
[macro-standard-extension-coverage]
exten = s,1,SetVar(PrimaryChannel=${ARG1})
exten = s,2,SetVar(DelayedChannel=${ARG2})
exten = s,3,Dial(Local/[EMAIL PROTECTED] Local/[EMAIL PROTECTED],18)
[delayed]
exten = 1,1,Dial(${PrimaryChannel})
exten = 2,1,Wait(12)
exten = 2,2,Dial(${DelayedChannel})
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[Asterisk-Users] alert-info and Cisco 7960 phones (6.1)
Getting the 7960 to use the Bellcore-dr1 through dr5 was fairly straightforward. The release notes indicate that you can trigger other ringtones on the phone (in the section Support for SIP Alert-Info Header), but I can't get anywhere with it. ...the Alert-Info header consists of a name of an internal tone or ringing pattern that can be played, as shown in the following example: Alert-Info: Bellcore-Busy There is no need to add a file extension (.au, .wav) to these names... Does internal exclude those rings loaded from ringlist.dat? Thanks for any thoughts or pointers, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: DISA
On Thu, 5 Feb 2004, John Todd wrote: So, to boil your problem down to what I think is the problem: When you attach an inbound call to the DISA application, it does not produce a dialtone fast enough. snip [main1] ; ; Take any number, and give it to the DISA. The DISA ; just then takes anything typed in within the (unchangeable) ; timer values, and hands it off to main2 to be post-processed. ; I include the standard i,h,t values for pedantic reasons. ; exten = _X.,1,DISA(no-password,main2) exten = _X.,2,Hangup ; exten = h,1,Hangup exten = i,1,Congestion exten = i,2,Hangup exten = t,1,Congestion exten = t,2,Hangup Not to point out the obvious, but isn't the delay he's seeing caused by the _X. and the digittimeout? Couldn't this be resolved by using a more specific match on the DISA instead of _X. ? Steve [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF recognized improperly?
We are getting reports from the receptionist that callers are having dtmf
problems (131 gets read as 113, 117 read as 111) - first digit recognized
twice, third digit not read at all).
I have tried 6 different analog phones, two cell phones (on two different
networks), and a digital phone behind another pbx. I can't reproduce the
problem at all.
Switch layout:
{Channelized T1} - [Adtran Total Access 624] - {POTS} - [Merlin Legend
Switch] - {POTS} - [Carrier Access AB1 w/ 8 FXO] - {T1} - [TE410P]
I would just dismiss the problem, except that the recpetionist hears about
it atleast twice a day. Most of the callers are either on POTS or cell
phones (we're a school district, it's parents that call/complain). Has
someone been able to reliably reproduce this problem? Is there an analog
phone that will repeatedly do this?
I do have 'relaxdtmf' turned on.
Many thanks,
Steve
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Re: [Asterisk-Users] Cisco 7960 BTXML
On Wed, 24 Dec 2003, Cameron Palmer wrote: Has anyone devloped some perl scripts for sending Directory and Services information to the Cisco 7960? The older thread I googled is giving me fits. My understanding is that the SIP images only use BTXML internally. I think the MGCP can use BTXML-driven services. The directory should be available (An example is http://lists.digium.com/pipermail/asterisk-users/2003-May/013013.html) For 'Services' you have to use a subset of the CMXML (I think maybe v3.0?) with SIP, it doesn't support all the latest stuff. Good luck, Steve ___ Steve Creel[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk freezes, no manager traffic, console functions
I have asterisk running as a voicemail system off of our Merlin Legend switch. We replaced our old Audix Voice Power (when the power supply fan died and burned it up) with asterisk a week ago. Many thanks to those who provided information about integrated VMI on the legend. The Audix system would, after a mailbox was closed, wait a few seconds, then use that line to dial the switch and update the MWI status. I have asterisk mimicking this behavior with a script that watches the manager port for the MessageWaiting information and then issues the same MWI update call. The Audix also used to refresh MWI status when the line was idle for long periods of time. I replicated this behavior by a perl script that gets a list of all of the extensions on the Legend, and then one-by-one updates their MWI status. If I leave this script running (calls take 2 seconds each, then a 30 second interval between mailboxes), eventually, asterisk will freeze - a little. An open connection to the manager port shows that manager information stops flowing. 'show channels' in the CLI will show that one port (of the group used for the MWI updates) is left in 'Dialing'. For asterisk to respond to any telephony events, it must be stopped and started (though 'stop now' from the CLI does do the job). Any suggestions as to where I might look to find the problem? ___ Steve Creel[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7960 Directory - limit?
On both a 7960 and a 7960G, the 'External Directory' will display only up to 32 entries. When pressing the 'More' key, there is a 'Next' option. When you pick 'Next', it does contact the web server, but displays no new information. The complete directory has 99 entries. I took a subset from the beginning and a subset from the end (40 entries each), and both times, the directory would show only the first 32. Is this a limitation others have seen? Should I just expect to deal with it? Thanks, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Looking for recommendations for home office setups
On Fri, 14 Nov 2003, James Harrell wrote: snip We're a small software company, with employees working from home in three different locations: - Atlanta: cable modem connection - Denver: ADSL/PPPOE connection - Oklahoma City: ADSL/PPPOE connection Is this a pipe dream? Here's my goal: - One phone one fax at each location - One telco phone line at each location - Utilize existing phones, though willing to buy new phones - Central asterisk server in Atlanta - Phone line best rate routing, outgoing calls routed through the hard-line at a different location if local, etc. ie: One can originate a call from the Atlanta phone, have it routed through the Denver outgoing line to another location in Denver to achieve a local phone call. As far as I understand, this may involve three hardware interfaces, one at each location (plus a central asterisk box). Each would have: - TCP/IP connection back to the central asterisk server (perhaps via a VPN? Or can we just use straight TCP/IP with some form of authentication. Caveat: we have NAT firewalls at each location. - Local telco phone line input - Analog line output for using existing phone, or potentially go via ethernet to a true IP phone? I believe I'm looking for some form of gateway box at each location, controlled by the Asterisk server. Possible? If so, what hardware is recommended. First, thanks for the very nicely prepared (and well thought-out) message. You are looking for 1 FXO port to bring in the local telco line at each site. You want two FXS ports to provide asterisk dialtone to the existing phone (assuming it's an analog phone) and dialtone to the fax machine. If your solution ends up being an asterisk box at each location, you can do this with 1 X100P and a 2-port TDM400P. Your best-rate routing is absolutely no problem. The 'swich' statement will be your friend. You shouldn't need VPN unless you're concerned about encryption. IAX2 (which, as you mention NAT, you'd be -strongly- encouraged to use) will do authentication for you. Good luck, Steve ___ Steve Creel[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SoftFax question
On Wed, 12 Nov 2003, Freddi Hansen wrote: Hi, I am looking at using the softfax that Steve Underwood has developed. It's very straight forward when you assign an extension for the fax. A function that several pbx's has is that they listen for the 'faxtone' for 5 seconds after 'answer' in the menu where you can enter your local extension number, it's normally done in parallel with the DTMF detection. I think that snip You want a fax extension: exten=fax,1,Blah() A google for 'fax extension' turns up the announcement of this feature here: http://lists.digium.com/pipermail/asterisk-users/2002-October/005414.html ___ Steve Creel[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone got VM2 working with MySQL?
See mbranca's patch at: http://bugs.digium.com/bug_view_page.php?bug_id=441 On Mon, 27 Oct 2003, WipeOut wrote: I guess the subject says it all.. :) I am running the CVS from right now.. +- 12:25 GMT MySQL CDR logging is installed and working.. Anyone got any ideas? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Steve Creel[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] retrieve_?_from_mysql.pl files??
You'll want to #include it. This leaves the burden of the [general] and any static configs on sip.conf but allows the script to blindly write out from the database to sip_additional.conf in sip.conf: #include sip_additional.conf Steve On Tue, 21 Oct 2003, WipeOut wrote: I know I am posting to myself all the time here but as i didnt find any info on this when I was looking it may help others.. I have just been playing with the retrieve_sip_from_mysql.pl.. Some notes.. You must create an entry with the keyword account and the value will be what you want between the [] otherwise it will ignore any other parameters and not create the entry.. It does not create the [general] section of the sip.conf so I guess you will have to add that manually (anyone got any comments on this?).. By default it wants to create a file called sip_additional.conf.. Does Asterisk look for a file named sip_additional.conf when it loads? or do you have to merge the contents of sip.conf and sip_additional.conf?? Later.. WipeOut wrote: Hi, I was just taking a look at the source code and noticed two files.. retrieve_extensions_from_mysql.pl and retrieve_sip_conf_from_mysql.pl Its pretty obvious what these two files do, but info about them is a little scarce.. Is anyone using these scripts and could give me any details on them? Is there a similar script for voicemail.conf floating about? Thanks, Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Steve Creel[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] retrieve_?_from_mysql.pl files??
It's documented somewhere for extensions.conf, and I was delighted to see that it is a function of the config parser, so yes - it's available in the other files. On Tue, 21 Oct 2003, WipeOut wrote: Steve Creel wrote: You'll want to #include it. This leaves the burden of the [general] and any static configs on sip.conf but allows the script to blindly write out from the database to sip_additional.conf in sip.conf: #include sip_additional.conf Steve Excellent, Thanks for that.. I didn't know there was an include command.. Do you know if include is available in other .conf files eg extensions.conf?? Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Steve Creel[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] #include in config /New subject/
It was originally announced by Mark in http://lists.digium.com/pipermail/asterisk-users/2002-March/001766.html It happens as a function of the config file parser (config.c) and will work across all of the config files parsed by it. On Tue, 21 Oct 2003, Olle E. Johansson wrote: Steve Creel wrote: You'll want to #include it. This leaves the burden of the [general] and any static configs on sip.conf but allows the script to blindly write out from the database to sip_additional.conf in sip.conf: #include sip_additional.conf Eureka! ...is this #include construct a general command for all config files? I must have missed it - where is it documented? /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Steve Creel[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] retrieve_?_from_mysql.pl files??
See: http://bugs.digium.com/bug_view_page.php?bug_id=343 What kind of details do you need? Steve On Mon, 20 Oct 2003, WipeOut wrote: Hi, I was just taking a look at the source code and noticed two files.. retrieve_extensions_from_mysql.pl and retrieve_sip_conf_from_mysql.pl Its pretty obvious what these two files do, but info about them is a little scarce.. Is anyone using these scripts and could give me any details on them? Is there a similar script for voicemail.conf floating about? Thanks, Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Steve Creel[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7905G phones
Michael, I've got one here on the bench with the SIP image. I've not done a ton of testing with it, but I have yet to run into a problem with it. Steve On Thu, 16 Oct 2003, Michael Devenijn wrote: I bought a couple of 7905G phones with a Callmanager license but i found on the site these phones can have a SIP image (which i downloaded) but before i upload the image i want to know if anybody tested them ? Michael ___ Steve Creel[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Directory App - excluding users...
Does anyone have any suggestions for excluding certain users from the directory? Can I just leave the 'Name' field empty in voicemail.conf Certain voicemail boxes shouldn't show up in the directory (company president, etc). I assume this can be handled safely by just leaving out the 'name' in voicemail.conf To go a step further, it would be good to allow them to put the company president's name in, and be given to his secretary. I can sort of get this working by putting in duplicate entries in voicemail.conf. The first one is the real mailbox, the second one is the president's name, but his secretary's mailbox. This seems shady. Would it make sense to add another field in the voicemail.conf for 'extension to transfer to' ? 100 = 1234,President's Secretary 101/100 = 1234,Company President Thanks for your thoughts, Steve ___ Steve Creel[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Loop counter variable in dialplan?
Google search: site:lists.digium.com loop counter There are several threads that discuss it, I liked the 5th one down: http://lists.digium.com/pipermail/asterisk-users/2003-September/019760.html On Wed, 8 Oct 2003, Matt Lawson wrote: How can I loop through something x number of times in the dialplan? i.e. if I get an invalid extension I want to re-play the menu, but not forever. Maybe 3 tries or something. I'm pretty sure that I've seen it before, where you can increment a variable and do Gotos based on it. But I've searched the Asterisk handbook, searched the user archives, and Googled for it, and can't find it now. Anyone have a link with an example of this? Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Steve Creel[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie question
Chris, I can't help you with experience with the openswitch12 board. It sounds like you're wanting a combination of physical analog extensions and soft phones. Depending on the number of analog extensions you'll need, look at getting a T400P and use a channel bank or two. No, you don't need any physical controllers to use soft phones. If you want 4 analog extensiosn and 10 soft phones, you only buy hardware for the 4 analog extensions. On Wed, 8 Oct 2003, Chris Mader wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I have the following questions: 1.) Can Asterisk interface with the OpenSwitch12 board? I've read some postings that say yes and some that said no. 2.) If I use a soft phone do I still need to purchase a board like the open switch or TDM400P to handle the internal extensions? General details of our PBX upgrade: Currently we have a comdial PBX system and we are going to be branching our company into two locations. I have found that it will be cheaper to setup two * systems then to upgrade our current system. (Currently we have 32 internal extension system and 16 incoming lines.) I already know that I am going to use the Wildcard T100P with a PRI from the phone company. My only concern are how I handle the extensions at the users desks. Thanks for the help Chris Mader -BEGIN PGP SIGNATURE- Version: PGP 8.0 iQA/AwUBP4RMl/v5JE8nnIbeEQLd6gCeJyC7riwMx5gHfrflCO1be3wbzEYAnRi7 rbIEcxkkVSh+Sm7VG+xPEMhd =l/E5 -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Steve Creel[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Second Send: Using PCI backplane
You are wanting to use a PCI backplane and put a bunch of TDM400P FXS cards instead of a T1 and a channel bank? If that's what you're asking... A T1 card and a channel bank yield 24 extensions. If you figure the TDM400P is $305 for 4 extensions, it would cost $1830 to get enough FXS ports (not to mention the IRQ and other problems you may run into with that many cards). The T1 card is $495, leaving you $1335 to find a 24FXS channel bank (which is more than enough on eBay). A large number of extensions would be handled by way of T1s and channel banks - up to 96 channels on one pci card with the T400P (1 pci card and 4 channel banks). Out of curiosity, why are you reluctant/opposed to a T1 and channel bank? Steve On Tue, 7 Oct 2003, Dennis Gearon wrote: I am wondering if it's possible to use a bunch of cards in a PCI backplane instead of going out to the extensions with T1 and then and adapter. How are people connecting to large amounts of extensions? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Steve Creel[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Source?
for more information. Sometimes I realize the solution when I'm putting all the information together to email someone about a problem. As most of you teachers (past and present) should know, not all of us learn the same. Some people just get written material. Some NEED the spoon to make it to the next level. Some need the hands-on experience and other's just can't learn any more than they have already know(those people are not likely on this list, however). In a group of people, some will prefer to sit down by themselves and read every word written about a topic. Others would prefer to sit down and talk about it with other people. Others would prefer to sit and play solitare on the computer while someone talks at them about it. If you've done alot of reading and want to talk about what you've read and questions you have, use the IRC channel. Until using Asterisk, I had never used IRC, despite being around it (and involved with linux projects) for years. Seriously, look at the IRC channel to talk about things. You do realize that the http://www.asterisk.org/index.php?menu=support lists the mailing list first for support, don't you. In fact, you have to go to the second page before you even see the google reference. More a few people tend to look for the FIRST way to get help not ALL ways to get help... flame suit on On Thu, Sep 18, 2003 at 08:31:59PM +0200, Dave Cotton wrote: ... Absolutely agree with you Steve. I left teachers training college in 1970. I shock some teachers when I said that in all the years since I haven't taught anyone anything. I've just enabled them to learn. The problem is that in most national education systems the teacher is expected to provide the answers to pass some test at the end of the course. Thinking is not part of the curriculum. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Steve Creel[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail to a commercial PBX/key phone system
Martin, We currently are in dire need of a system to replace our Audix Voice Power connected to our Merlin Legend switch. A little while ago, I knew absolutely nothing about how to do this, and inquired on the list. The response I got were extremely helpful to me, as I had no idea where to start. The Legend currently has 8 analog lines in a hunt group to the voicemail/auto-attendant system. I went in after-hours and put a buttset inline to monitor the first line in the hunt group. When I call in from an outside line, there is a series of DTMF codes sent, and the auto-attendant plays. After making a selection, there is a hookflash, more DTMF, then the line is hung up. In hindsight, it makes perfect sense that this is how it is handled. Steve On Sat, 13 Sep 2003, marrandy wrote: Hello. I've seen some mentions of asterisk possibly being used as an inexpensive voicemail attachment to a commercial PBX etc. Does anyone here, have experience of using it in this fashion ? What commercial systems have been successfully attached too ? How is the attachment made ? Analog, digital ? If anyone has successfully accomplished this, I would like to hear the make and model of phone system and how they did the voicemail connection to an asterisk system. On or off list is fine. Regards...Martin -- I have accepted Provolone into my life! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Steve Creel[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail to a commercial PBX/key phone system
On Sat, 13 Sep 2003, John Brown wrote: On Sat, Sep 13, 2003 at 05:21:40PM -0400, Steve Creel wrote: The Legend currently has 8 analog lines in a hunt group to the voicemail/auto-attendant system. I went in after-hours and put a buttset inline to monitor the first line in the hunt group. When I call in from an outside line, there is a series of DTMF codes sent, and the Are those 8 analog lines from the telco, or from the merlin? From the merlin... ___ Steve Creel[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 + SIP
For anyone who's interested, I called Cisco at 800-213-1542 and talked to a Service Contract Sales Representative for a Smartnet SNT 8x5xNBD on my 7960. It was $8/yr, charged to my credit card, and I had my CCO access four hours from the time I'd called. You will need to have the serial number from the phone. Buy the SMARTnet, it just makes life so much easier. Steve On Fri, 12 Sep 2003, David C. Troy wrote: What is the best number to call (in the US) to get setup with this support? I wish Cisco would just make this stuff purchasable over their website. I'd pay $8/yr if they didn't make it so complicated. As it is I have to take my Cisco rep out to breakfast to find anything out... Dave Hi Shaun and anyone else looking for Cisco images, I don't know what the support contract would cost on a 7960 for the Cisco TAC, but for the ATA186 it's a great, big $8/year. This gives you access to the Cisco TAC, images, and support team which do a fantastic job of follow-through. So I would recommend calling them and asking about the support contract for your particular phone. If you're too cheap to pay the fees, you can always find a service provider that supports the device and do a tftp cross-grade, upgrade, etc. of the firmware. iConnect, Nikotel, etc all support the ATAs, but not the 7960s. The only catch is if you bought the device second-hand. Then there is the chance that your device is ineligible for support. On Friday, September 12, 2003, at 08:11 AM, Shaun Ewing wrote: Hello again, After doing some searching of the list archives, I came across a message by John Todd posted back in July () To cut a long story short, to be able to use SIP on my phone, I need to P0S30203.bin image. Is there anyway of getting this image without getting a Cisco SMARTnet agreement? -Shaun - Original Message - From: Shaun Ewing [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, September 12, 2003 3:58 PM Subject: [Asterisk-Users] Cisco 7960 + SIP Hello all, I know this isn't strictly Asterisk, but I'm sure that there are more people here using the Cisco 7960 w/ SIP, so I thought I'd post here. I've just bought a Cisco 7960 phone to use with Asterisk. It came with the CallManager image on it. I've got the 4.4 SIP images (P0S3-04-4-00). If I put P0S3-04-4-00 in the OS79XX.TXT file, the phone downloads this fine (watching TFTP server debug). It then proceeds to request P0S3-04-.bin. I don't know why. Naturally this file isn't found. I tried renaming the file to P0S3-04-.bin. The phone then downloads around 80% before aborting. I hope somebody might be able to shed some light on the situation. Any help would be greatly appreciated. Thanks, Shaun ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Steve Creel[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Manager / Windows Apps / Line Appearances
It just dawned on me as I was playing with the manager interface - it can't be very difficult at all to write an Win32 app that serves as a lamp field. Between 'Newchannel', 'Newstate', and 'Hangup' events, all of the information is there. I've heard several requests for line appearances, but mgcp and sccp channels don't currently include support. I know that in all the instances I'd like to have call appearances, a windows application would be an equally valid solution. My problem is that I know nothing about writing little Win32 apps like that. While I can give it a shot (and I will), I'm sure there is someone far more qualified who could probably write it much better and far more quickly. Just my $0.02 Steve ___ Steve Creel[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ADTRAN TSU 600 VP24 FXO 24 Port Channel Bank
Then yes, it will work and do what you're looking for it to do. On Wed, 20 Aug 2003, Bartosz Jozwiak wrote: I want to connect analog telephone lines only. The analog lines telecom gives you :) - Original Message - From: Steve Meyers [EMAIL PROTECTED] To: Asterisk List [EMAIL PROTECTED] Sent: Wednesday, August 20, 2003 11:34 AM Subject: Re: [Asterisk-Users] ADTRAN TSU 600 VP24 FXO 24 Port Channel Bank On Wed, 2003-08-20 at 07:58, Mark Spencer wrote: The FXO ports will only allow you to connect phone lines, not actual phones, but since FXO ports are more expensive in general than FXS ones, it's likely you could find someone to trade. We probably should have a list dedicated to trading/selling/buying asterisk related hardware, but failing that i would suggest people just contact you off-list. Yeah, but will it work? What if he wants 24 port FXO, not FXS? Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Steve Creel[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail2 - auto fill the dialing extension?
On Fri, 8 Aug 2003, Adams, Gavin wrote:
Now it's back to tweaking the configuration on our SIP phones (7960s).
The message_uri parameter in the phone's configuration file is working
great. Dials comedian mail directly. Is there a way to let voicemail2
know what the incoming extension is, and use it?
Sure, something like:
exten = 85000,1,VoicemailMain2(${CALLERIDNUM})
Also, we decided to go with actual extension numbers on the phones
instead of usernames per extension. On the Cisco phones, is there a way
to change the name/number on the top line (white text on black) to the
user's name, while having the extension number next to each presentation
(line1, line2, etc)?
Use phone_label for that text...
___
Steve Creel[EMAIL PROTECTED]
___
Asterisk-Users mailing list
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[Asterisk-Users] shared line-appearance
shared line-appearance: a visual indication on one phone that another phone is in use, such as a lighted button. I understand that SIP does not support a shared line appearance. I understand that MGCP as a protocol does support shared line appearances in its Business Phone Package (rfc3149), but that chan_mgcp does not support the Business package, and thus does not support shared line appearances. I understand that if I want it bad enough, I need to extend chan_mgcp to support it. The phone systems I am replacing all offer this feature, and users are used to it (Not to mention that any feature you take away instantly becomes their most valued feature that they _need_ to do their job). My question: How are others working without this? We have many instances where an assistant is to answer incoming calls and put them on hold until the supervisor has an opportunity to take the call (to keep the supervisor from putting their current call on hold to answer the second call and put it on hold). I can only assume that this situation comes up for other people as well. I considered using queues, but then the supervisor is not aware of the number of calls waiting for them. I think I have to look at mgcp, but was curious as to what others are doing. Many thanks, Steve ___ Steve Creel[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Vendors for phones
I'm in the process of setting up a test/demonstration system to show that VoIP is realistic and applicable for our needs. We put a 7905 and 7960 on a request for quote that went out the other day (to people like CDW Microwarehouse). All of the vendors returned thier quotes without including the Cisco phones. So my question: where do you buy your phones? We can't buy direct from Cisco (must have 3 quotes). Thanks... ___ Steve Creel[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] grandstream sip phone
I asked [EMAIL PROTECTED] the other day. They wrote back: US list retail price of BudgeTone SIP phones: Model 101 $75/ea (available now) Model 102 $85/ea (available now) US list retail price of HandyTone VoIP analog telephone adaptor: $75/ea (available in late July 2003) Please contact our reseller (Ovislink/dgtimes) regarding your sample purchase. James @ Ovislink/dgtimes can be reached at tel: (626) 854-1805 or fax: 626.854.0835 and [EMAIL PROTECTED] Their web site is at: www.ovislink.com On Wed, 16 Jul 2003, Marian Danisek wrote: hello, i found in list archives some notes about grandstream sip voip phones. Does anybody succesfuly tested those phones with asterisk ? Mark ? What about the prices ? regards Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ A mind is like a parachute... it only works when it's open. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Steve Creel[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXS and PBX Integration
It sounds like you want to use the IAX to provide dialtone to the Panasonic PBX? You'd use FXS cards in the asterisk box to provide signal into a CO port on the Panasonic. On Wed, 16 Jul 2003, Iván Aponte wrote: Hi All, I got a doubt about something I want to do with asterisk. I have this office (site a) with only a Panasonic analog PBX and another office (site b) with an Asterisk Box with an ADIT 600 . I want to interconnect both via IAX. Is it possible to put a new asterisk box in site a without the channel bank and put a card (FXS or FXO???) and connect it to the pbx as a CO line ? What kind of card do I need a FXS or an FXO card? Regards, Iván Aponte -- Iván Aponte email: [EMAIL PROTECTED] Office: +58(212)9524620 Mobile: +58(414)2774713 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Steve Creel[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Migration to Asterisk - Running off of Merlin Legend system
We currently have a Merlin Legend system. The voicemail is falling apart (with the transition to a 10 digit timestamp on Sept. 8, 2001, the system locked up and refused to take calls; the official solution is to change the system time back to a year with a matching calendar). We are in the process of preparing the network infrastructure to support a VoIP system with Asterisk, but won't be there for a few months. We'd like to go ahead and replace the voicemail system with Asterisk now, and as we're ready, drop the Merlin system. My questions: Right now, the voicemail system (and auto-attendant) are connected to the switch by 4 analog lines. Logic says that these are FXS cards in the switch, like any other extension. The switch handles an incoming call and transfers it to the auto-attendant. How would such a call be identified to be dropped in the appropriate context? When the phone switch fails to reach someone at an extension, it transfers them to the voicemail system. How could these calls be identified as different from an incoming call to the auto-attendant? How is the appropriate mailbox or extension identified? Thanks, Steve ___ Steve Creel[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
