Re: [Asterisk-Users] List of Motherboards or Servers that are tested ok with Asterisk and Digium boards
Man, looking back it was a gas - the 16k wobbly rampack. You spent 30 minutes looking at a blank screen whilst loading "Horace goes skiing" (or some other c*appy game you wanted to hack) making incantations to the tapedrive god in the vain hope that you wouldn't get an I/O error. It wasn't a gas then. ;) Always coveted a 48k Spectrum. Got a CPC6128 instead .. Julian. Matt Riddell wrote: Julian Lyndon-Smith wrote: Asterisk is cool. But maybe not that cool. Hey, don't you know that the dev team gets all the cool toys ;) You can tell I started coding on a ZX81. Woohoo go the ZX81!!! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Hung Zap channels
Original Message From: "John Heng" <[EMAIL PROTECTED]> To: Sent: Friday, November 18, 2005 1:56 AM Subject: [Asterisk-Users] Hung Zap channels Hi all, Once in a while, I've found that the zap channel will get stuck (or blocked) even after the call has ended. The way I've fix this is to issue a "soft hangup" command for that zap channel. However, I'm not always aware of this until a user tells (or complains to) me. What I would like to know is if there is a way to reset all the zap channels or re-initialize the drivers without restarting Asterisk. If so, I could set up a cron job to do it once or twice a week, in the middle of the night. Any suggestion guys?? To have a channel blocked for ½-1 week would not be good, I think... Can you determine in a script if a channel is hung? Then do a soft hangup on it. Run this in cron. Or regularly do a soft hangup on any channel which haven't had activity in x minutes. But the best solution is naturally to determine why the channel hangs and fix the problem. Leif Cheers J Heng ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Subject: [Asterisk-Users] Eicon Diva Server query
Hi Avi, > > I've given up on crappy passive ISDN cards and am heading into the wild > world of real, Active Super Dooper Server boards. I have a choice of two > Eicon Diva Server cards: > > Eicon Diva Server 4BRI > Eicon Diva Server V-4BRI > > The V-4BRI is actually cheaper, but I'm guessing its for voice only (which > isn't a problem, going into an Asterisk box). Do these cards play nicely > with Linux (2.6 kernel) and Asterisk? Any tips/tricks/pitfalls? > EICON offers rpm and deb packages for all major distributions and versions. The installation and configuration of the drivers is fairly easy. Just add chan_capi_cm after the installation of the driver and you're done. We are using the 4BRI and the BRI cards and never had any problems. If you need any help or hints you may contact me off the list, too. Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Lösungen ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] List of Motherboards or Servers that are tested ok with Asterisk and Digium boards
Julian Lyndon-Smith wrote: > Asterisk is cool. But maybe not that cool. > > Hey, don't you know that the dev team gets all the cool toys ;) > > You can tell I started coding on a ZX81. Woohoo go the ZX81!!! -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] is there any free pocket pc softphone??
alfa wrote: > hello all, > > > is there any free pocket pc softphone http://www.sineapps.com/news.php?rssid=1089 -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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: Re: SIP - Loop detected (Matt Riddell)
Trond Andersen wrote:
> Thank you, but I have tried that... Then the To is:
Can you do a NoOp(${ARG1}) and then show us the result?
--
Cheers,
Matt Riddell
___
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Re: [Asterisk-Users] Bristuff / Junghanns / Customer Service
Chris Bagnall wrote: >>I bought a quadbri card from junghanns around two years ago. > > > I've never dealt with the company in question, but isn't it a bit much to > expect any company to take a product back after two years of use? Not if he was told to wait for it to work. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Eicon Diva Server query
Armin Schindler wrote: Actually the V-4BRI should be more expensive than the 4BRI. The 'V' does mean Voice, but this card has more Voice-features besides the standard 4BRI DSP features (I think it's G.723). Thanks for that. The quote was AU$400 less for the V-4BRI, though I'll double-check that. :) Any feedback on how well these cards perform with Asterisk? Are there other Active QuadBRI cards easily available in Australia that I should be investigating? Thanks, Avi -- National Manager - Special Projects < Melbourne / Sydney / Canberra / Hobart / London /> 2/340 Gore Street T: +61 (0) 2 6233 0607 Fitzroy, VIC F: +61 (0) 2 6233 0696 3065 W: http://www.squiz.net/ .>> Open Source - Own it - Squiz.net ./> ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] TDM04b on FreeBSD
Jason Becker wrote: Alejandro Mejia Evertsz wrote: Hi list! I successfully installed a Digium TDM04B card on FreeBSD 5.4 using zaptel drivers for FreeBSD (installed with ports). I'm using Asterisk CVS-Head and the card works fine, but when placing or recieving a call on any of the 4 fxo ports, users hear (both sides) a "clicking" noise. I also have a Wildcard X100P installed, and uses the same configuration (on zapata.conf) but that card doesn't make that strange noise during conversations. Please let me know if someone had this problem before me, and what you did to correct it. I don't know what else to try. Could the TDM400P be sharing an interrupt? systat? Regards, use should write it to bsd ML. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Eicon Diva Server query
On Fri, 18 Nov 2005, Avi Miller wrote: > Hello gurus! > > I've given up on crappy passive ISDN cards and am heading into the wild > world of real, Active Super Dooper Server boards. I have a choice of two > Eicon Diva Server cards: > > Eicon Diva Server 4BRI > Eicon Diva Server V-4BRI > > The V-4BRI is actually cheaper, but I'm guessing its for voice only (which > isn't a problem, going into an Asterisk box). Do these cards play nicely > with Linux (2.6 kernel) and Asterisk? Any tips/tricks/pitfalls? Actually the V-4BRI should be more expensive than the 4BRI. The 'V' does mean Voice, but this card has more Voice-features besides the standard 4BRI DSP features (I think it's G.723). But this does not matter when using with Asterisk and chan_capi currently. Both cards support RTP, which can be used to bridge a call directly to a SIP phone where the Diva card is doing all the necessary stuff (codecs, jitter, echo-cancel). But this feature is not yet implemented into chan_capi yet (due to lack of time and the Asterisk API does not really help). Anyway, for Asterisk you can use both cards with the same features. Armin ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] newbie questions
On Fri, 2005-11-18 at 01:09 +0100, [EMAIL PROTECTED] wrote: > You can put analogue phones in series, but I am not sure how many phones > a FSX connection will drag and I don't think the cards are designed for > it - don't know. Sounds to me as you would benefit from downloading > Asterisk and play around with a few softphones first and maybe buy > Digiums starter kit, cause you need to take into mind that Asterisk is > not realy a Plug & Play environment, it do require people who are > comfortable with the command prompt on Linux and have the time to fiddle > around with this (or money to pay someone to do so) Thanks to all. I have a better idea now. Linux is not the issue, I know it quite well. However, I know _nothing_ about telephony (traditional or voIP)... in a process of getting onto that learning curve... > > jan fred > > > Chris Wade wrote: > > > Fred Blaise wrote: > > > >> On Thu, 2005-11-17 at 14:48 -0600, Chris Wade wrote: > >> > >>> [EMAIL PROTECTED] wrote: > >>> > hi, > > > My second question: for a branch office of about 20 people, which E1 > > card do you advise? Would the TE210P be a good choice? (number of > > concurrent calls would be max 10 for now) Why? > > > > > An E1 has 30 lines, so you would be perfect with a TE110P. > Connecting an E1 to a company PABX is however an expensiveoption, > so you might want to compare the prices with 10 analogue lines or > maybe 5 BRI lines. I would not let the price of hardware decide > this because you will need to pay a fixed cost per month for PSTN > lines, so check these prices first. Asterisk is scalable in the > sence that you can add more later if you have an available PCI slot. > >>> > >>> Even though they don't appear to be shipping yet, don't forget the > >>> TDM2400P's from Digium. Up to 24 FXO or FXS ports per full length > >>> PCI card. > >> > >> ok. So, with this in mind, if I was to acquire that 24 ports, 1 FXO > >> modules, the rest FSX modules. I could have 1 public telephone number to > >> the PSTN (3 wasted for this example), 20 analog phones inside the > >> company's branch, each phone having its extension in Asterisk? Or could > >> I have just the smaller card allowing me 1 FSO and 3 FSX and some kind > >> of "hub" to connect some number of analog phones (let's say 20)? If so, > >> does the number of FXS limit the number of my simultaneous telephone > >> calls? > >> > >> Sorry for the dumb questions, but your answers are highly appreciated. > > > > > > Sorry, but there is really no such thing as a "hub" for telephone > > lines. Each analog phone must be plugged into its own FXS port. > > > > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Speech recognition or TTS with Asterisk?
Hello, I am interested in TTS with Asterisk. Anyone implemented this port? Thanks in adbvance, John B While not being true "TTS", there are efforts by LumenVox to incorporate speech recognition into Asterisk. They were at Astricon, and they also were present at IP4IT in the Digium booth on Monday/Tuesday of this week. I have spoken with Gerd Graumann at LumenVox, and he says that they are planning for a Q1 release. While I don't have any written details, there were discussions about making a single-user license "very affordable", which probably means $50 or less I suspect. This would be a native Linux environment for all components. Again, while I have no specific details, I believe that Digium is working with them to develop a dialplan application that would allow specific word-matching rules to be easily built. http://www.LumenVox.com/ JT ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SPA 3000 and MWI
My spa3000 is acting funny and I don't think it's an asterisk issue but thought someone might know what's going on. I reset the unit to factory configs yesterday and that's when it started. Whenever I have message waiting set to yes (in the user 1 config area of the sipura) I get the stuttered dialtone when the receiver is picked up even when no messages are waiting. Asterisk is sending the following to the spa300: Event: message-summary Content-Type: application/simple-message-summary Content-Length: 80 Messages-Waiting: no Message-Account: sip:asterisk@ Voice-Message: 0/0 (0/0) Any ideas what could cause this? Chris ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk voicemail responses - feature?
I am using what I think is a standard out of the box Asteriskathome setup, with voicemail, and digital receptionist etc I have recorded the names for each extensions and they show up as XXXivrrecording.wav in /var/lib/asterisk/sounds where XXX is the extension number. the response to a call say from PSTN is receptionist (user presses # for first name directory) directory (user presses 866 for Tom) Allison responds "T-o-m if this is the person you are looing for please press 1" (user presses 1) Allison responds "The person at extension 221 is unavailable, please leave your message after the tone" etc. I was expecting the 221ivrrecording.wav to kick in at some point but it never does. Of course, I would like to hear the recording rather than the 2-2-1 Am I missing something obvious? Anthony ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] SIP INVITE IP address variable?
On 11/17/05, John Todd <[EMAIL PROTECTED]> wrote: > > Perhaps this was already discussed in the archives, but due to the > generic nature of the keywords I've been unable to find it. My > apologies if it's an easy answer. > > I am looking for the variable or other method that will allow me to > determine (from within the dialplan) the IP address of the origin of > the SIP INVITE message for the current leg. I know about $SIPURI - > this is not sufficient. That describes the SIP URI of the original > requester. I need to find the IP address which actually transmitted > the final INVITE to Asterisk, regardless of the URI or any of the SIP > headers. Headers lie. UDP packets don't. > > Note: this is for any SIP INVITE, regardless of if the INVITE is > received in [general] or if it has a specific peer entry. > JT, On a 1.2 machine, you should be able to use the SIPCHANINFO dialplan function as SIPCHANINFO(recvip) to get what you're looking for. BJ -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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: Can anyone explain reason for this echo
I purchased the following item: http://www.oriontelecom.com/echo_canceller/1u_telnet/e1_1u_19inch_ec.html As you can see not a very highly spec'd product but does the job well. I don't accept the fact that mine is a special case. In fact if anything it should be better than most other scenarios as we are using Tier 1 hardware (all HP), Digium Rev 2 firmware and our rack is about 10 metres from the CO. On 11/18/05, Andrew Kohlsmith <[EMAIL PROTECTED]> wrote: On Thursday 17 November 2005 21:01, Eric Bishop wrote:> I got sick of tweaking and playing with Digium's ridiuculous voodoo so I> just bought a dedicated E1 PRI echo canceller and bingo, problem solved.> Digium make some good IP PBX software and hardware but all their echo > cancellers, hardware and software are complete rubbish.There are many, many of us who disagree. The echo was not solveable on yourparticular installation. You could have a longer tail than the software echo can Asterisk has can handle, and longer than the Digium hardware echo can canmanage. I am interested in the echo can you settled upon, and what its specsare.Would you mind sharing this information? -A.___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 1.2 under OS X?
Has anyone compiled 1.2 on OS X? If so, do all the realtime components compile properly? Thanks, HJ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] CVS v1-2-0 make problems?
Do a search on this list, there is a fix for this. Rare but can happen. Greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kristian Kielhofner Sent: Thursday, November 17, 2005 3:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] CVS v1-2-0 make problems? Asterisk-users, Has anyone else had problems with the v1-2-0 CVS rev? Here's the deal: LATE last night I checkout out 1.2.0 with CVS: rm -rf asterisk zaptel libpri cvs co -r "v1-2-0" zaptel cvs co -r "v1-2-0" libpri cvs co -r "v1-2-0" asterisk zaptel and libpri build fine. Asterisk, however, seems to get stuck in a infinite loop while (guessing) determining version. The loop occurs when using cmp to check version.h and version.h.tmp. It goes on forever, forever, and forever. However, using the 1.2.0 tarballs work perfectly, for libpri, zaptel, and asterisk. Yes, this is for AstLinux and it is using my cross-build environment. (Which has worked very well for tracking CVS HEAD at build.astlinux.org, and as mention before can build using the 1.2.0 tarballs). I'd have more time to dig deep into it, but I am just trying to get a 1.2 build of AstLinux done. I somewhat foolishly promised one by tomorrow :). Anyone else experiencing this? Are my CVS commands wrong? What's up? Thanks in advance, and a HUGE thank you to everyone at Digium for getting 1.2 out! -- Kristian Kielhofner ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Help with shell script for externnotify
On Friday 18 November 2005 15:32, Tom Rymes wrote: > Basically, I have 14 after-hours mailboxes that all have different e- > mail addresses. I want this script to parse the mailbox number from > the original command ($2), and then somehow look that up mailbox's > address and send an e-mail. It then checks every five minutes to see > if the message has been retrieved, and escalates things as necessary. > I don't mind the messy solution of defining all 14 addresses in the > script itself, though it would be nice to look it up from > voicemail.conf or something eventually. I'm not sure I understand what you are trying to do, but this may (or may not) help. You mentioned looking up the email field from voicemail.conf, this should do that: EXTEN=`echo $2 | cut -f 1 -d @` EMAIL=`cat voicemail.conf | grep '^$EXTEN' | cut -d ',' -f 3` The above ignores contexts so if you have more than one it will not work. HTH hads -- "At a recent meeting in Snowmass, Colorado, a participant from Los Angeles fainted from hyperoxygenation, and we had to hold his head under the exhaust of a bus until he revived." ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Eicon Diva Server query
Hello gurus! I've given up on crappy passive ISDN cards and am heading into the wild world of real, Active Super Dooper Server boards. I have a choice of two Eicon Diva Server cards: Eicon Diva Server 4BRI Eicon Diva Server V-4BRI The V-4BRI is actually cheaper, but I'm guessing its for voice only (which isn't a problem, going into an Asterisk box). Do these cards play nicely with Linux (2.6 kernel) and Asterisk? Any tips/tricks/pitfalls? Any input appreciated Thanks, Avi -- National Manager - Special Projects < Melbourne . Sydney . Canberra . Hobart . London /> 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, Victoria F: +61 (0) 3 9486 0611 3202 W: http://www.squiz.net/ .>> Open Source - Own it - Squiz.net ./> ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP INVITE IP address variable?
Perhaps this was already discussed in the archives, but due to the generic nature of the keywords I've been unable to find it. My apologies if it's an easy answer. I am looking for the variable or other method that will allow me to determine (from within the dialplan) the IP address of the origin of the SIP INVITE message for the current leg. I know about $SIPURI - this is not sufficient. That describes the SIP URI of the original requester. I need to find the IP address which actually transmitted the final INVITE to Asterisk, regardless of the URI or any of the SIP headers. Headers lie. UDP packets don't. Note: this is for any SIP INVITE, regardless of if the INVITE is received in [general] or if it has a specific peer entry. JT ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how to originate a call and capture it's DIALSTATUS
Hello, I've been trying to originate calls and capture the DIALSTAUS via the manager API. The problem seems that the API doesn't expose enough data to make a decision of what exactly happened to the call. It results in something like this: Action: Originate Channel: IAX2/switch/1 MaxRetries: 0 WaitTime: 2 Context: reminder Extension: s Priority: 1 Callerid: "Reminder" <555-555-> Event: Hangup Privilege: call,all Channel: IAX2/switch-3 Uniqueid: 1132271784.42 Cause: 0 Cause-txt: Unknown this is far from detailed. Is there a way to extract the actual DIALSTATUS such as ANSWER,BUSY,CONGESION, etc? The Cause doesn't seem to return 0 when the call is terminted thru IAX2 or SIP. It seems that it works on ZAP only. ScriptHead ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Bristuff / Junghanns / Customer Service
> I bought a quadbri card from junghanns around two years ago. I've never dealt with the company in question, but isn't it a bit much to expect any company to take a product back after two years of use? Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help with shell script for externnotify
Hi folks, I am working on a shell script that I can use with the externnotify command in voicemail.conf. All is well and seems ready to rock, except I can't figure out how to tell the script what e-mail address to send the mail messages to. I warn you ahead of time that I am no scripting guru. Basically, I have 14 after-hours mailboxes that all have different e- mail addresses. I want this script to parse the mailbox number from the original command ($2), and then somehow look that up mailbox's address and send an e-mail. It then checks every five minutes to see if the message has been retrieved, and escalates things as necessary. I don't mind the messy solution of defining all 14 addresses in the script itself, though it would be nice to look it up from voicemail.conf or something eventually. I started out using /bin/sh for the scripting, but I assume that this is limiting me with what I can do with variables, etc. The only way I can think of off the top of my head is to use some sort of nested variable, like this subset of my script: Tmpmail=/tmp/dispatch$$.mail [EMAIL PROTECTED] # Send an e-mail to the appropriate address for this mailbox: echo "724-6066 $2" > $Tmpmail echo "New Voicemail Message #$3 in mailbox $2" >>$Tmpmail cat $Tmpmail | mail -s "$mailsubject" $ext$2 now, the "$ext$2" is what I mean by nested variable. Basically, if I can find a way to make this evaluate as $ext108, and then as [EMAIL PROTECTED], I would be a happy camper. Is there any way to do this? I know that this is technically not an Asterisk question, but then again, it is. If anyone has wrestled with this in their own externnotify scripts, please let me know Tom PS: I know that perl and other scripting languages would be better for this, but this seemed simpler to me at the time I started. If it can't be done using sh, then I'll start over in perl, but if there's a way to make this work, it's the only thing standing between me and a script that works exactly as I want. Tom Rymes Cascade Link Systems www.cascadelinksystems.com (603) 375-1414 "Intelligent technology solutions for small businesses." ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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: Can anyone explain reason for this echo
On Thursday 17 November 2005 21:01, Eric Bishop wrote: > I got sick of tweaking and playing with Digium's ridiuculous voodoo so I > just bought a dedicated E1 PRI echo canceller and bingo, problem solved. > Digium make some good IP PBX software and hardware but all their echo > cancellers, hardware and software are complete rubbish. There are many, many of us who disagree. The echo was not solveable on your particular installation. You could have a longer tail than the software echo can Asterisk has can handle, and longer than the Digium hardware echo can can manage. I am interested in the echo can you settled upon, and what its specs are. Would you mind sharing this information? -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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: Can anyone explain reason for this echo
I got sick of tweaking and playing with Digium's ridiuculous voodoo so I just bought a dedicated E1 PRI echo canceller and bingo, problem solved. Digium make some good IP PBX software and hardware but all their echo cancellers, hardware and software are complete rubbish. On 11/18/05, Doug Meredith <[EMAIL PROTECTED]> wrote: Eric Bishop <[EMAIL PROTECTED]> wrote:>If I call our Asterisk box via Disa and then place a call to one of the>problem analogue numbers (native Zap bridge) I don't get any echo. So the >echo seems to occur only when using a SIP handset and making a call to an>analogue number.The echo is probably always there. You only notice it with the SIPphone because of the additional latency that this introduces. Doug--Doug Meredith ([EMAIL PROTECTED])SystemGuard - Oracle remote support877-974-8273 (87-SYSGUARD)506-854-7997 www.systemguard.com___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hung Zap channels
Hi all, I'm running asterisk 1.0.9 (yes I know - 1.2 has just been released) with a TDM400P board that has 4 FXO port. Once in a while, I've found that the zap channel will get stuck (or blocked) even after the call has ended. Sometimes this is when someone has left a voice msg, but not always. The way I've fix this is to issue a "soft hangup" command for that zap channel. However, I'm not always aware of this until a user tells (or complains to) me. What I would like to know is if there is a way to reset all the zap channels or re-initialize the drivers without restarting Asterisk. If so, I could set up a cron job to do it once or twice a week, in the middle of the night. Any suggestion guys?? Cheers J Heng ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] multi tenant with queues
On 11/17/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote: I had the exact same dilemma and switched to using AddQueueMember/RemoveQueueMember instead of using agents. This solved my problem. Thanks!! That looks like a better solution all the way around. Chris ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] multi tenant with queues
I had the exact same dilemma and switched to using AddQueueMember/ RemoveQueueMember instead of using agents. This solved my problem. - Waldo On Nov 17, 2005, at 7:13 PM, snacktime wrote: I'd like some feedback on my solution so far for using queues in a multi tenant configuration. For most of the configuration files I've been able to use a naming scheme for the context names, which works nicely for making multi tenant fairly transparent. However that won't work for everything and queues is one of them. In queues.conf the naming scheme will work for defining a queue. It won't work for the agents though as they all have to have unique names. My thought is to create a pool of available agent numbers, and the web gui for the tenants will let the tenant pick the agent numbers they want to assign out of the pool. As numbers are used they are taken out of the pool, and as they become available they go back into the pool. The downside to this is that a tenant won't get to pick the exact numbers they want, but that doesn't seem like too much of a compromise for a multi tenant system. Anyone have any better ideas? Chris ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Missing smp kernel package in Asterisk 1.2installation...
Have you done this? ln -s /lib/modules/`uname -r`/build /usr/src/linux-2.6 ln -s /lib/modules/`uname -r`/build /usr/src/linux to make sure the sources are linked correctly in the /usr/src directory? John -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Leo Burd Sent: Thursday, November 17, 2005 7:08 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Missing smp kernel package in Asterisk 1.2installation... Hello there, I've just downloaded Asterisk 1.2 into my RedHat Enterprise Linux machine and got the following problem when I tried to compile zaptel: "You do not appear to have the sources for the 2.6.9-22.ELsmp kernel installed." However, according to rpm -qa, I do have the following packages installed in my system: kernel-smp-2.6.9-22.EL kernel-smp-devel-2.6.9-5.EL Am I doing anything wrong? If so what shall I do to fix this problem? In fact, I've never experienced this issue in the previous version of Asterisk. BTW, I'm only planning to use my server to handle voip calls ... do I still need to compile zaptel? Thank you so much for your help, Leo ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] What's the best way to stream and/or convert MP3 and WAV files?
Hello everyone, I'm implementing an audioblog application and have some questions about how to best stream and/or convert MP3 and WAV files to be played by Asterisk. Currently, I first copy the files from the server to my machine, convert them to Wav and play. Unfortunately, this process is not very efficient at all. Here are my questions: a) Is there any way to play MP3 files directly by Asterisk? b) What is the best command line to be used to convert MP3 files into any format that can be played by Asterisk? Shall I use sox or anything like that? If so, what would be the proper parameters to pass? c) Is there any way to stream MP3 files directly into Asterisk and still allow users to forward/pause/rewind the file while playing? d) Does Asterisk play any kind of WAV file? Do I need to convert Wav files before playing them? Thank you so much for your help, Leo ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Missing smp kernel package in Asterisk 1.2 installation...
BTW, I'm only planning to use my server to handle voip calls ... do I still need to compile zaptel? According to the doc yes, and you need to install some dummy driver that connect to some USB timer thing. Jan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] multi tenant with queues
I'd like some feedback on my solution so far for using queues in a multi tenant configuration. For most of the configuration files I've been able to use a naming scheme for the context names, which works nicely for making multi tenant fairly transparent. However that won't work for everything and queues is one of them. In queues.conf the naming scheme will work for defining a queue. It won't work for the agents though as they all have to have unique names. My thought is to create a pool of available agent numbers, and the web gui for the tenants will let the tenant pick the agent numbers they want to assign out of the pool. As numbers are used they are taken out of the pool, and as they become available they go back into the pool. The downside to this is that a tenant won't get to pick the exact numbers they want, but that doesn't seem like too much of a compromise for a multi tenant system. Anyone have any better ideas? Chris ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Mission-Critical Deployments
> We've got a couple dozen BT-101s deployed in an office environment. No > babysitting, no complaints from users, no rebooting. Their sound > quality isn't the same as a Cisco 7920, but neither is their price. . . > > In other words, folks, YMMV. > > I say it's worth a person's while to invest a bit in a prototype lab, > and that way you can form your own opinions and not have to rely on the > often-conflicting opinions of others. . > > B. I think the main lesson here is - test! Decide! Live with it! (or make sure you can live with it...) PaulH ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Mission-Critical Deployments
> > Hm..it's pretty close price wiseI thought the channel banks > > would cost more... > > > > With regards to functionality, I would have to test the two setups side by > > side. > > I know that at a site we setup, the grandstream BT101's came out about the > > same as cheap analogs with regards to quality and functionality... > > This is exactly what I disagree with. The BT101's are not worth > *anything* even if you pay me to take them I will *never* install them > for a client. They need babysitting, rebooting, terrible sound > quality, and are very not userfriendly. Going the analog way (vs > BT101) is not close pricewise it is a *lot* cheaper, since it works. > The BT101's don't, it creates to much trouble for any > office to deal with. Quality wise, an analog phone is the quality > users are looking for, since that's what they are used to. The BT101 > cannot offer that. Functionality: what function does the BT101 have > that you like so much? last time I checked the conf button wasn't even > working, xfers you get with features.conf, and ringers sound much > nicer on analog phones, CallerID works much better on analog phones. > Can you please name me one feature that the BT101 has that is at least > as good as an analog phone (besides for the xfer button, which with > any decent analog phone can be programmed if it has dedicated one > touch speed dial buttons)? I think I already agreed with you - that nice analog phones are better than cheap IP phones. PaulH ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] newbie questions
You can put analogue phones in series, but I am not sure how many phones a FSX connection will drag and I don't think the cards are designed for it - don't know. Sounds to me as you would benefit from downloading Asterisk and play around with a few softphones first and maybe buy Digiums starter kit, cause you need to take into mind that Asterisk is not realy a Plug & Play environment, it do require people who are comfortable with the command prompt on Linux and have the time to fiddle around with this (or money to pay someone to do so) jan Chris Wade wrote: Fred Blaise wrote: On Thu, 2005-11-17 at 14:48 -0600, Chris Wade wrote: [EMAIL PROTECTED] wrote: hi, My second question: for a branch office of about 20 people, which E1 card do you advise? Would the TE210P be a good choice? (number of concurrent calls would be max 10 for now) Why? An E1 has 30 lines, so you would be perfect with a TE110P. Connecting an E1 to a company PABX is however an expensiveoption, so you might want to compare the prices with 10 analogue lines or maybe 5 BRI lines. I would not let the price of hardware decide this because you will need to pay a fixed cost per month for PSTN lines, so check these prices first. Asterisk is scalable in the sence that you can add more later if you have an available PCI slot. Even though they don't appear to be shipping yet, don't forget the TDM2400P's from Digium. Up to 24 FXO or FXS ports per full length PCI card. ok. So, with this in mind, if I was to acquire that 24 ports, 1 FXO modules, the rest FSX modules. I could have 1 public telephone number to the PSTN (3 wasted for this example), 20 analog phones inside the company's branch, each phone having its extension in Asterisk? Or could I have just the smaller card allowing me 1 FSO and 3 FSX and some kind of "hub" to connect some number of analog phones (let's say 20)? If so, does the number of FXS limit the number of my simultaneous telephone calls? Sorry for the dumb questions, but your answers are highly appreciated. Sorry, but there is really no such thing as a "hub" for telephone lines. Each analog phone must be plugged into its own FXS port. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Mission-Critical Deployments
Original Message - From: "Brian Capouch" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Friday, November 18, 2005 10:18 AM Subject: Re: [Asterisk-Users] Mission-Critical Deployments > C F wrote: > > > This is exactly what I disagree with. The BT101's are not worth > > *anything* even if you pay me to take them I will *never* install them > > for a client. They need babysitting, rebooting, terrible sound > > quality, and are very not userfriendly. Going the analog way (vs > > BT101) is not close pricewise it is a *lot* cheaper, since it works. > > The BT101's don't, it creates to much trouble for any > > office to deal with. > > We've got a couple dozen BT-101s deployed in an office environment. No > babysitting, no complaints from users, no rebooting. Their sound > quality isn't the same as a Cisco 7920, but neither is their price. . . > > In other words, folks, YMMV. > > I say it's worth a person's while to invest a bit in a prototype lab, > and that way you can form your own opinions and not have to rely on the > often-conflicting opinions of others. . > Sounds like why I wrote - 'I would have to test them side by side' PaulH ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Missing smp kernel package in Asterisk 1.2 installation...
Hello there, I've just downloaded Asterisk 1.2 into my RedHat Enterprise Linux machine and got the following problem when I tried to compile zaptel: "You do not appear to have the sources for the 2.6.9-22.ELsmp kernel installed." However, according to rpm -qa, I do have the following packages installed in my system: kernel-smp-2.6.9-22.EL kernel-smp-devel-2.6.9-5.EL Am I doing anything wrong? If so what shall I do to fix this problem? In fact, I've never experienced this issue in the previous version of Asterisk. BTW, I'm only planning to use my server to handle voip calls ... do I still need to compile zaptel? Thank you so much for your help, Leo ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk 1.2.0 and memory usage
Hello from the shift to the stable 1.2.0 version, the used memory dropped from 1Gb to 1.3Gb dos anyone have clues on this memory used 1.2.0-rc1 = 960 Mb 1.2.0-rc2 = 1000 Mb 1.2.0 = 1300 Mb best regards Thierry tél: +33 (0)3 90 40 06 75 fax: +33 (0)3 90 40 06 75 email: [EMAIL PROTECTED] web: http://www.widevoip.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] no longer loading all config files?!?!?!?!?!!!!!!...
On Thu, November 17, 2005 23:39, Francesco Peeters said: > I have been testing with the issue I have when using 2 ISDN HFC-BRI cards > with ZapHFC / BRIstuff. (Which *seems* to be working, but cannot test > because of this new problem!) > > Everything was working fine until earlier this evening... > > Now it doesn't seem to execute anything beyond chan_skinny (the next one > normally being pbx_config.so, which loads the dial-plan), which means it > is about as useful for telephony as the paperweight on my desk! :-( > > Reload info: > [EMAIL PROTECTED] root]# asterisk -rx reload > Verbosity is at least 28 > == RTP Allocating from port range 1 -> 2 > -- Reloading module 'res_adsi.so' (ADSI Resource) > -- Reloading module 'res_crypto.so' (Cryptographic Digital Signatures) > -- Reloading module 'res_indications.so' (Indications Configuration) > -- Unregistered indication country 'us' > -- Unregistered indication country 'au' > -- Unregistered indication country 'fr' > -- Unregistered indication country 'de' > -- Unregistered indication country 'nl' > -- Unregistered indication country 'uk' > -- Unregistered indication country 'fi' > -- Unregistered indication country 'no' > -- Unregistered indication country 'br' > -- Unregistered indication country 'za' > -- Unregistered indication country 'it' > -- Registered indication country 'us' > -- Registered indication country 'au' > -- Registered indication country 'fr' > -- Registered indication country 'de' > -- Registered indication country 'nl' > -- Registered indication country 'uk' > -- Registered indication country 'fi' > -- Registered indication country 'no' > -- Registered indication country 'br' > -- Registered indication country 'za' > -- Registered indication country 'it' > -- Setting default indication country to 'us' > -- Reloading module 'chan_sip.so' (Session Initiation Protocol (SIP)) > -- Reloading module 'chan_agent.so' (Agent Proxy Channel) > -- Reloading module 'chan_mgcp.so' (Media Gateway Control Protocol > (MGCP)) > -- Reloading module 'chan_iax2.so' (Inter Asterisk eXchange (Ver 2)) > Reloading SIP > Reloading MGCP > == Loaded firmware 'iaxy.bin' > -- Loaded provisioning template 'default' > -- Reloading module 'chan_local.so' (Local Proxy Channel) > -- Reloading module 'chan_skinny.so' (Skinny Client Control Protocol > (Skinny)) > > > Any ideas/suggestions, apart from starting from scratch? (Which I will do > if necessary, but I *am* trying to prevent rebuilding from scratch for the > 4th time in 2 weeks!) :-( > UPDATE: It *still* has to do with the friggin' ZAPHFC BRIstuff drivers!!! If I have bri_net_ptmp enabled for channels 4-5, it doesn't execute the remainder of the system config, if I have it disabled (and thus revert to bri_cpe_ptmp) for channels 4-5, then all works fine, except that it wouldn't be running NT mode on the second card. (thus defeating the purpose of the second card!) lspci shows: 00:09.0 Network controller: Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] (rev 02) Subsystem: Advanced Integrations Research: Unknown device c101 Flags: bus master, medium devsel, latency 16, IRQ 11 I/O ports at c400 [disabled] [size=8] Memory at e3001000 (32-bit, non-prefetchable) [size=256] Capabilities: [40] Power Management version 1 00:11.0 Network controller: Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] (rev 02) Subsystem: Advanced Integrations Research: Unknown device c101 Flags: bus master, medium devsel, latency 16, IRQ 11 I/O ports at cc00 [disabled] [size=8] Memory at e3002000 (32-bit, non-prefetchable) [size=256] Capabilities: [40] Power Management version 1 Help please?... -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] newbie questions
Hi fred, For the branch office you could consider a 2 (or more) port E1/T1 card. You can utilize one port for an incoming E1 from the telco (BT?) and then the second port run to a T1 Channel Bank such as the Rhino 24 port FXO http://www.myphonecall.co.uk/voip/channelbanks/rhino/default.aspx - this is the closest to the 'hub' you describe. You would need the cabling in your patch panel to break up the Rhino's 50-way 'telco' connector to something you can patch (rj45/rj11) or run to your analogue phones. I don't have any recommendations for this as I'm still looking at this for an installation. This would give you an asterisk setup with ZAP channels 0..31 incomming and 32..56 your extensions. You will have the advantage of reliable faxing (ie with fax machines) with this solution instead of going to sip phones and sip/ata/fax. I'm looking to do this in our office closer to christmas - all mentioned hardware known to have good asterisk support. Chris Fred Blaise wrote: Hi all I am new to this whole field, being it PSTN or voIP. I am currently reading the "Switching to VoIP" and "Asterisk: The Future of Telephony", so hopefully, I will be less clueless soon :) My first question: if I buy a Wildcard TDM400P, with one X100M and three S100M modules, I would be able to have 1 telephone number given out by my company to come in to my asterisk server, and I could plug in 3 analog phones onto that card, am I correct? Hence, do we have a 1-to-1 relationship here for either modules? My second question: for a branch office of about 20 people, which E1 card do you advise? Would the TE210P be a good choice? (number of concurrent calls would be max 10 for now) Why? Thank you all. Cheers fred ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Speech recognition or TTS with Asterisk?
On Thu, 2005-11-17 at 12:10 -0700, John Brookes wrote: > Can this be implemented in Java? sure that can be implemented in Java. Have a look at Asterisk-Java at http://asteriskjava.org. Asterisk-Java is to Java what phpagi is to PHP. =Stefan signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] HFC ISDN card and mISDN driver
> I get the same errors with the install-misdn script from Beronet: Replying to myself to say that I solved this issue by downloading the CVS version of mISDN from isdn4linux's CVS repository and replacing the one from the tarball that gets downloaded by the Makefile. However, I still cannot get Asterisk to startup with mISDN, chan_misdn and an /etc/asterisk/misdn.conf file -- it keeps saying "init_stack: Function not implemented". If I remove the misdn.conf file, * will start, but won't initialise my card (obviously). So, I'm back to CAPI and chan_capi-cm on * 1.0.9 until I can find an alternative ISDN-BRI card that allows multiple instances in a single PC. Darn AVM!Fritz cards. *sigh* cYa, Avi -- National Manager - Special Projects < Melbourne . Sydney . Canberra . Hobart . London /> 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, Victoria F: +61 (0) 3 9486 0611 3202 W: http://www.squiz.net/ .>> Open Source - Own it - Squiz.net ./> ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] call levels
Hi, One solution would be to configure each IP phone (in sip.conf) on a different context. [phone1] context=local [phone2] context=intl Then make 2 different contexts named "local" and "intl" on extensions.conf [local] exten => _9X.,1,yourDialString ; Nine followed by any number is just an example as I don't know the dial plan in Argentina [intl] include => local ; Including local context to be available on intl too exten => _00X.,1,yourDialString This just looks like crap if you hadn't take the time to read a bit about the dialplan. I recommend you to look at: http://www.voip-info.org/wiki-Asterisk+config+extensions.conf voip-info.org has lots of nice documentation ;) Hope it helps a bit... Buena suerte! Alejandro Mejia From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mariano GonzalezSent: Jueves, 17 de Noviembre de 2005 03:52 p.m.To: asterisk-users@lists.digium.comSubject: [Asterisk-Users] call levels Hello all. This is my first time with Asterisk, may be my question is fool. I have a two IP phone. I need that the first phone makes calls to local numbers only and the second phone make calls to all numbers. Somebody know the solution? Thanks a lot. Mariano ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Mission-Critical Deployments
C F wrote: This is exactly what I disagree with. The BT101's are not worth *anything* even if you pay me to take them I will *never* install them for a client. They need babysitting, rebooting, terrible sound quality, and are very not userfriendly. Going the analog way (vs BT101) is not close pricewise it is a *lot* cheaper, since it works. The BT101's don't, it creates to much trouble for any office to deal with. We've got a couple dozen BT-101s deployed in an office environment. No babysitting, no complaints from users, no rebooting. Their sound quality isn't the same as a Cisco 7920, but neither is their price. . . In other words, folks, YMMV. I say it's worth a person's while to invest a bit in a prototype lab, and that way you can form your own opinions and not have to rely on the often-conflicting opinions of others. . B. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SER & Asterisk combination to get around NAT
Has anyone successfully used SER and Asterisk together on the same server to get around NAT traversal issues. I have looked at many of the NAT traversal topics which either involve commercial products and significant costs or solutions such as STUN or proprietary systems such as xten. -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.13.3/174 - Release Date: 17/11/2005 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] newbie questions
On Fri, November 18, 2005 0:02, Chris Wade said: > Fred Blaise wrote: > Sorry, but there is really no such thing as a "hub" for telephone lines. > Each analog phone must be plugged into its own FXS port. > Unless you are willing to put telephones in parallel, like you do when connecting multiple telephones to a single PSTN line without a PBX... The big problems there are that: - you won't be able to call between telephones on the same 'line' - all telephones on the same 'line' can eavesdrop on an already existing conversation on that 'line' - you may exceed the connection rate (power use) on the FXS and blow it up... Good luck! -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] stop asterisk when Idle
[EMAIL PROTECTED] wrote: I need to reboot every day an asterisk box, but I would like to do that only when asterisk is not doing anything. Let me be the third person to ask you to post detailed technical information on the problem you are having so it can be fixed, and you won't have to restart the box. Please stop thinking like a Windows admin. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: call levels
"Mariano Gonzalez" <[EMAIL PROTECTED]> wrote: >I need that the first phone makes calls to local numbers only and the second >phone make calls to all numbers. Research "extensions.conf" and/or "dial plan". Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] newbie questions
Fred Blaise wrote: On Thu, 2005-11-17 at 14:48 -0600, Chris Wade wrote: [EMAIL PROTECTED] wrote: hi, My second question: for a branch office of about 20 people, which E1 card do you advise? Would the TE210P be a good choice? (number of concurrent calls would be max 10 for now) Why? An E1 has 30 lines, so you would be perfect with a TE110P. Connecting an E1 to a company PABX is however an expensiveoption, so you might want to compare the prices with 10 analogue lines or maybe 5 BRI lines. I would not let the price of hardware decide this because you will need to pay a fixed cost per month for PSTN lines, so check these prices first. Asterisk is scalable in the sence that you can add more later if you have an available PCI slot. Even though they don't appear to be shipping yet, don't forget the TDM2400P's from Digium. Up to 24 FXO or FXS ports per full length PCI card. ok. So, with this in mind, if I was to acquire that 24 ports, 1 FXO modules, the rest FSX modules. I could have 1 public telephone number to the PSTN (3 wasted for this example), 20 analog phones inside the company's branch, each phone having its extension in Asterisk? Or could I have just the smaller card allowing me 1 FSO and 3 FSX and some kind of "hub" to connect some number of analog phones (let's say 20)? If so, does the number of FXS limit the number of my simultaneous telephone calls? Sorry for the dumb questions, but your answers are highly appreciated. Sorry, but there is really no such thing as a "hub" for telephone lines. Each analog phone must be plugged into its own FXS port. -- Christopher L. Wade, CCNA, CCDA, CQS-CIPCES, CQS-CWLSS ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Mission-Critical Deployments
On 11/17/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: > > I disagree with PaulH on this one. Cheap IP phones makes for *cheap* > > phone, cheap sound, and cheap features. The cheapest IP phone you can > > get will come to around $60.00 USD, which multiplied by 150 makes > > $9,000.00. While a channel bank (ADIT 600) with 6 FXS cards (48 ports) > > runs around $1200.00 multiplied by 3 (3 * 48 = 144 the closest I can > > get without overbuying) makes for $3600.00, each QuadT1 card runs > > around $1,500.00 or $2,500.00 with echo can, multiplied by 2 makes > > $5,000.00 at the most, Total = $8,600.00 at the most, and you already > > have the phones, and I'm telling you that it will be cheaper. Also, > > you might have to rerun wiring for VoIP, beside the fact that for > > cheap VoIP phones you don't get POE, which also means you need outlets > > where you are going to put phones, as well as in featurewise; you can > > do much more in the DP with ananlog phones (or VoIP since it's in the > > DP), then *any* VoIP phone under $100.00 can do without the DP, and > > even a Cisco or Polycom cannot do much without some fancy programming > > from the phone itself with no DP. > > Hm..it's pretty close price wiseI thought the channel banks > would cost more... > > With regards to functionality, I would have to test the two setups side by > side. > I know that at a site we setup, the grandstream BT101's came out about the > same as cheap analogs with regards to quality and functionality... This is exactly what I disagree with. The BT101's are not worth *anything* even if you pay me to take them I will *never* install them for a client. They need babysitting, rebooting, terrible sound quality, and are very not userfriendly. Going the analog way (vs BT101) is not close pricewise it is a *lot* cheaper, since it works. The BT101's don't, it creates to much trouble for any office to deal with. Quality wise, an analog phone is the quality users are looking for, since that's what they are used to. The BT101 cannot offer that. Functionality: what function does the BT101 have that you like so much? last time I checked the conf button wasn't even working, xfers you get with features.conf, and ringers sound much nicer on analog phones, CallerID works much better on analog phones. Can you please name me one feature that the BT101 has that is at least as good as an analog phone (besides for the xfer button, which with any decent analog phone can be programmed if it has dedicated one touch speed dial buttons)? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] newbie questions
On Thu, 2005-11-17 at 14:48 -0600, Chris Wade wrote: > [EMAIL PROTECTED] wrote: > > hi, > > > >> My second question: for a branch office of about 20 people, which E1 > >> card do you advise? Would the TE210P be a good choice? (number of > >> concurrent calls would be max 10 for now) Why? > >> > >> > > An E1 has 30 lines, so you would be perfect with a TE110P. Connecting an > > E1 to a company PABX is however an expensiveoption, so you might want to > > compare the prices with 10 analogue lines or maybe 5 BRI lines. I would > > not let the price of hardware decide this because you will need to > > pay a fixed cost per month for PSTN lines, so check these prices first. > > Asterisk is scalable in the sence that you can add more later if you > > have an available PCI slot. > > Even though they don't appear to be shipping yet, don't forget the > TDM2400P's from Digium. Up to 24 FXO or FXS ports per full length PCI card. ok. So, with this in mind, if I was to acquire that 24 ports, 1 FXO modules, the rest FSX modules. I could have 1 public telephone number to the PSTN (3 wasted for this example), 20 analog phones inside the company's branch, each phone having its extension in Asterisk? Or could I have just the smaller card allowing me 1 FSO and 3 FSX and some kind of "hub" to connect some number of analog phones (let's say 20)? If so, does the number of FXS limit the number of my simultaneous telephone calls? Sorry for the dumb questions, but your answers are highly appreciated. signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] realtime callerid
Hello, Is there a way to restrict realtime to not set callerid via sippeers table even if there a column callerid in that table.something like restrictcid in sip.conf Thank you, ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] no longer loading all config files?!?!?!?!?!!!!!!...
I have been testing with the issue I have when using 2 ISDN HFC-BRI cards with ZapHFC / BRIstuff. (Which *seems* to be working, but cannot test because of this new problem!) Everything was working fine until earlier this evening... Now it doesn't seem to execute anything beyond chan_skinny (the next one normally being pbx_config.so, which loads the dial-plan), which means it is about as useful for telephony as the paperweight on my desk! :-( Reload info: [EMAIL PROTECTED] root]# asterisk -rx reload Verbosity is at least 28 == RTP Allocating from port range 1 -> 2 -- Reloading module 'res_adsi.so' (ADSI Resource) -- Reloading module 'res_crypto.so' (Cryptographic Digital Signatures) -- Reloading module 'res_indications.so' (Indications Configuration) -- Unregistered indication country 'us' -- Unregistered indication country 'au' -- Unregistered indication country 'fr' -- Unregistered indication country 'de' -- Unregistered indication country 'nl' -- Unregistered indication country 'uk' -- Unregistered indication country 'fi' -- Unregistered indication country 'no' -- Unregistered indication country 'br' -- Unregistered indication country 'za' -- Unregistered indication country 'it' -- Registered indication country 'us' -- Registered indication country 'au' -- Registered indication country 'fr' -- Registered indication country 'de' -- Registered indication country 'nl' -- Registered indication country 'uk' -- Registered indication country 'fi' -- Registered indication country 'no' -- Registered indication country 'br' -- Registered indication country 'za' -- Registered indication country 'it' -- Setting default indication country to 'us' -- Reloading module 'chan_sip.so' (Session Initiation Protocol (SIP)) -- Reloading module 'chan_agent.so' (Agent Proxy Channel) -- Reloading module 'chan_mgcp.so' (Media Gateway Control Protocol (MGCP)) -- Reloading module 'chan_iax2.so' (Inter Asterisk eXchange (Ver 2)) Reloading SIP Reloading MGCP == Loaded firmware 'iaxy.bin' -- Loaded provisioning template 'default' -- Reloading module 'chan_local.so' (Local Proxy Channel) -- Reloading module 'chan_skinny.so' (Skinny Client Control Protocol (Skinny)) Any ideas/suggestions, apart from starting from scratch? (Which I will do if necessary, but I *am* trying to prevent rebuilding from scratch for the 4th time in 2 weeks!) :-( TIA! -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Mission-Critical Deployments
> I disagree with PaulH on this one. Cheap IP phones makes for *cheap* > phone, cheap sound, and cheap features. The cheapest IP phone you can > get will come to around $60.00 USD, which multiplied by 150 makes > $9,000.00. While a channel bank (ADIT 600) with 6 FXS cards (48 ports) > runs around $1200.00 multiplied by 3 (3 * 48 = 144 the closest I can > get without overbuying) makes for $3600.00, each QuadT1 card runs > around $1,500.00 or $2,500.00 with echo can, multiplied by 2 makes > $5,000.00 at the most, Total = $8,600.00 at the most, and you already > have the phones, and I'm telling you that it will be cheaper. Also, > you might have to rerun wiring for VoIP, beside the fact that for > cheap VoIP phones you don't get POE, which also means you need outlets > where you are going to put phones, as well as in featurewise; you can > do much more in the DP with ananlog phones (or VoIP since it's in the > DP), then *any* VoIP phone under $100.00 can do without the DP, and > even a Cisco or Polycom cannot do much without some fancy programming > from the phone itself with no DP. Hm..it's pretty close price wiseI thought the channel banks would cost more... With regards to functionality, I would have to test the two setups side by side. I know that at a site we setup, the grandstream BT101's came out about the same as cheap analogs with regards to quality and functionality... PaulH ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] stop asterisk when Idle
But I found some situations that, after several millions of calls seconds,need to reboot the box and not only restart asterisk. That's really not necessary,and it's almost painful to watch people do this... If you posted some detailed information about your system and the problem you are having maybe someone could help you fix the actual problem. Chris ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Mission-Critical Deployments
On 11/17/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: > - Original Message - > From: "John Goerzen" <[EMAIL PROTECTED]> > To: > Sent: Friday, November 18, 2005 2:37 AM > Subject: [Asterisk-Users] Mission-Critical Deployments > > > > I work for a company that is nearing the end-of-life on its existing > > Nortel Meridian switch and is considering Asterisk. We have > > approximately 200 existing extensions, and probably 150 out of those 200 > > are using basic analog phones and would stay that way. The rest would > > have VOIP phones at the desk. > > > > We're seriously considering switching to Asterisk. I've done quite a > > bit of tinkering with Asterisk for my home, but I'm not certain about a > > few aspects of how we might deploy Asterisk in the enterprise. > > > > Here are my questions: > > > > 1. Where could I look for some resources on server sizing? Is it > >any problem to support this number of users with a single server? > > A decent dual xeon should be fine for that...or 2 or 3 smaller servers... > (depends on the funtionality you need) > > > 2. What do we need to do for our data network to make VOIP reliable? > >QoS, basic traffic prioritization on the switch, vlan, ??? > > If it's doable, a serapate data network for VOIP. > A friends install moved to that after running VOIP on their main network, > and it made a huge difference. > YMMV. > > > 3. What's the best way to integrate these 150 analog extensions? > >I've seen interface boxes that usually come in 24-port sizes. Some > >have an Ethernet/SIP interface to hook up to Asterisk, and others > >have a T1 interface. What sounds best and is the most reliable? > > Here I am going to disagree with you. Buy cheap IP phones. > The hardware, setup and lack of functionality of analog extensions makes > them a second choice for me. > I disagree with PaulH on this one. Cheap IP phones makes for *cheap* phone, cheap sound, and cheap features. The cheapest IP phone you can get will come to around $60.00 USD, which multiplied by 150 makes $9,000.00. While a channel bank (ADIT 600) with 6 FXS cards (48 ports) runs around $1200.00 multiplied by 3 (3 * 48 = 144 the closest I can get without overbuying) makes for $3600.00, each QuadT1 card runs around $1,500.00 or $2,500.00 with echo can, multiplied by 2 makes $5,000.00 at the most, Total = $8,600.00 at the most, and you already have the phones, and I'm telling you that it will be cheaper. Also, you might have to rerun wiring for VoIP, beside the fact that for cheap VoIP phones you don't get POE, which also means you need outlets where you are going to put phones, as well as in featurewise; you can do much more in the DP with ananlog phones (or VoIP since it's in the DP), then *any* VoIP phone under $100.00 can do without the DP, and even a Cisco or Polycom cannot do much without some fancy programming from the phone itself with no DP. > > 4. What is a good company to contract with for emergency support? > >Digium? > > Find a local consultant. There are quite a few around... > > > 5. What are people doing to make VOIP phones resiliant in the face of > >power outages? > > I have found the device called a UPS to be a useful in this regard, when > hooked up to POE. > > > Is there anybody here that would be willing to serve as a reference > > check for Asterisk should we pursue that path? > > I could. But I live in melbourne, australia. > Which is not the same as austria. > > > > > Thanks, > > > > -- John > > All just my 2 cents. > > PaulH > > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Zaptel Compile Error
Goran Donev wrote: The error message is: You do not appear to have the sources for the 2.6.9-22.0.1.EL kernel installed. make: *** [linux26] Error 1 I am installing it on a Cento 4.2 server. Can someone shed some light on this? yum install kernel-devel (or yum install kernel-smp-devel) Regards, -- Jason Becker Director & CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zaptel Compile Error
First Thanks to all who worked hard to release 1.20! I installed asterisk with no problem and when it came to installing the zaptel drivers I am getting the following errors. Can anyone help me? The error message is: You do not appear to have the sources for the 2.6.9-22.0.1.EL kernel installed. make: *** [linux26] Error 1 I am installing it on a Cento 4.2 server. Can someone shed some light on this? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 1.2 won't compile: res_config_odbc.c
Hi there, so far I didn't succeed in getting 1.2 compiled on a RH72 System (with gcc 3.0.4). I'd appreciate any tips... ;-> Cheers, Philipp gcc -shared -Xlinker -x -o res_features.so res_features.o gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing- declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 - march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer - DZAPATA_MOH -DOPENSSL_NO_KRB5 -fPIC -c -o res_config_odbc.o res_config_odbc.c In file included from ../include/asterisk/utils.h:35, from ../include/asterisk/cdr.h:48, from ../include/asterisk/channel.h:114, from ../include/asterisk/file.h:30, from res_config_odbc.c:37: ../include/asterisk/strings.h:34: warning: `always_inline' attribute directive ignored In file included from ../include/asterisk/cdr.h:48, from ../include/asterisk/channel.h:114, from ../include/asterisk/file.h:30, from res_config_odbc.c:37: ../include/asterisk/utils.h:173: warning: `always_inline' attribute directive ignored ../include/asterisk/utils.h:186: warning: `always_inline' attribute directive ignored ../include/asterisk/utils.h:199: warning: `always_inline' attribute directive ignored ../include/asterisk/utils.h:217: warning: `always_inline' attribute directive ignored res_config_odbc.c: In function `realtime_odbc': res_config_odbc.c:68: `SQLULEN' undeclared (first use in this function) res_config_odbc.c:68: (Each undeclared identifier is reported only once res_config_odbc.c:68: for each function it appears in.) res_config_odbc.c:68: parse error before "colsize" res_config_odbc.c:150: `colsize' undeclared (first use in this function) res_config_odbc.c: In function `realtime_multi_odbc': res_config_odbc.c:208: `SQLULEN' undeclared (first use in this function) res_config_odbc.c:208: parse error before "colsize" res_config_odbc.c:304: `colsize' undeclared (first use in this function) res_config_odbc.c: In function `update_odbc': res_config_odbc.c:344: `SQLLEN' undeclared (first use in this function) res_config_odbc.c:344: parse error before "rowcount" res_config_odbc.c:404: `rowcount' undeclared (first use in this function) make[1]: *** [res_config_odbc.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk-1.2.0/res' make: *** [subdirs] Error 1 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Overlapping sounds in asterisk and asterisk-sounds
Hi all, Just installing 1.2.0 and noticed that the following sounds that asterisk-sounds provides are already installed with asterisk: /var/lib/asterisk/sounds/conf-hasleft.gsm /var/lib/asterisk/sounds/conf-thereare.gsm /var/lib/asterisk/sounds/invalid.gsm What's the idea behind this - a bug, intentional, something else? Thanks and regards, Patrick ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Poor sounds on Adtran 750
> I changed from a tdm100B card over to an Adtran 750 as I added more PSTN > lines last week. I have a Sangoma A104u card and 12 channels FXO > connected to PSTN lines. I am experiencing very poor audio quality with > hum on the lines and poor volume. When I connected the Adtrans I > upgraded the firmware to the latest and reset the config to the factory > default. I also upgraded Asterisk to 1.2. > I am at a loss to explain why the quality has become so poor, has anyone > any advice? I'm not useing a 750, but just suggesting a possibility... Check to ensure the fxo lines have proper termination/impenance settings. 600 ohm in the US, etc. There was someone trying to use another channel bank some time ago in the Colorado area (I forgot the exact make/model), and the fxo card specs were actually listed as 15,000 ohms (no 600/900 ohm at all). He had the same 'hum' issue, and that was certainly understandable. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: stop asterisk when Idle
"If, on the other side, asterisk continue accepting incoming call, how can I be sure that I wll reach a "convenient" moment ?" If you are not sure if it will even reach a convenient moment, you will also not get a chance to run "stop now". -- -- Steven May you have the peace and freedom that come from abandoning all hope of having a better past. ---- --- - - - -- - - -- - - - --- - -- - - --- - - -- - -- -- - -- <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED] Yes you are right; I was considering asterisk -rx "stop when convenient" (remember I have to shutdown the box). But the poit is: I don't know what exactly this command does !! Does it stop accepting new calls ? If it is right, then I can stay for several hours (let's say I have 100 calls running, and one of them will continue for 5 hours) for 5 hours no one will be able to place a new call ? If, on the other side, asterisk continue accepting incoming call, how can I be sure that I wll reach a "convenient" moment ? So this is not the solution for me I think that I will issue a asterisk -rx "stop now", then I will check for the pidof asterisk and when I will not found it anymore I will reboot the pc. Of course, running calls will be dropped But I found some situations that, after several millions of calls seconds, need to reboot the box and not only restart asterisk. thank you, Andrea "Anton Krall" <[EMAIL PROTECTED] ruder.com.mx> To Sent by: "'Asterisk Users Mailing List - asterisk-users-bo Non-Commercial Discussion'" [EMAIL PROTECTED] m.com cc Subject 17/11/2005 11.36 RE: [Asterisk-Users] stop asterisk when Idle Please respond to Asterisk Users Mailing List - Non-Commercial Discussion <[EMAIL PROTECTED] ists.digium.com> How about a cron job that does: asterisk -rx "restart when convenient" I do this sometimes and does the trick. |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |[EMAIL PROTECTED] |Sent: Thursday, November 17, 2005 3:21 AM |To: asterisk-users@lists.digium.com |Subject: [Asterisk-Users] stop asterisk when Idle | |Is there a way to detect (via batch) if asterisk is idle i.e. |is there no active channels ? (oh323 show channels via console) | |I need to reboot every day an asterisk box, but I would like |to do that only when asterisk is not doing anything. | |So I would like to schedule a batch at a given time that, |before rebooting the system, checks if é* is idle. | |Is it possible to do that ? Or does it exist another way ? | |thanks in advance, | |Andrea | | |Chi ricevesse questa mail per errore e' gentilmente pregato di |cancellarla. | |Visitate il sito http://www.frameweb.it | |___ |--Bandwidth and Colocation sponsored by Easynews.com -- | |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 drops call when calling other VOIP
Does anyone have any ideas for this???
Tony Davidson wrote:
I don't think that's the issue as it works with 99.9% of people we call.
It's only 2 numbers so far that have had this issue. I'm pretty sure
one uses Asterisk at the other end, but I would have thought it
unlikely the second did. I think it's one of those auto switch boards
though - maybe this is causing the issue?
Tom Vile wrote:
its possible that your provider is not setup to use asterisk for your
account. I know a some providers that need to know if you are using a
regular SIP phone or Asterisk.
On 11/16/05, Tony Davidson <[EMAIL PROTECTED]> wrote:
I'm having an issue when Asterisk calls what I believe to be other VOIP
connections.
I can call the number from a normal sip phone, but when I attempt to
connect via Asterisk the call is dropped immediately. Checking my call
logs I can tell the call has connected but I think Asterisk is trying
something when it connects that immediately causes a dropout. My VOIP
connection is via a SIP account.
Tony
The log of the call is:
-- Called engin/030888
-- SIP/engin-f91f answered SIP/203-add5
== Spawn extension (macro-dialout-trunk, s, 17) exited non-zero on
'SIP/203-add5' in macro 'dialout-trunk'
== Spawn extension (from-internal, 0392210888, 1) exited non-zero on
'SIP/203-add5'
-- Executing Macro("SIP/203-add5", "hangupcall") in new stack
-- Executing ResetCDR("SIP/203-add5", "w") in new stack
-- Executing NoCDR("SIP/203-add5", "") in new stack
-- Executing Wait("SIP/203-add5", "5") in new stack
-- Executing Hangup("SIP/203-add5", "") in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on
'SIP/203-add5' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on
'SIP/203-add5'
asterisk1*CLI>
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--
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Phone: 978-203-3848 x205
Fax: 518-631-2856
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[Asterisk-Users] Re: Mission-Critical Deployments
Note: http://www.citel.com/products/handset_gateways/ sells a SIP handset gateway that will let you still use your Digital phones. We used it for our old NEC phones. -- -- Steven May you have the peace and freedom that come from abandoning all hope of having a better past. ---- --- - - - -- - - -- - - - --- - -- - - --- - - -- - -- -- - -- "John Goerzen" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED] >I work for a company that is nearing the end-of-life on its existing > Nortel Meridian switch and is considering Asterisk. We have > approximately 200 existing extensions, and probably 150 out of those 200 > are using basic analog phones and would stay that way. The rest would > have VOIP phones at the desk. > > We're seriously considering switching to Asterisk. I've done quite a > bit of tinkering with Asterisk for my home, but I'm not certain about a > few aspects of how we might deploy Asterisk in the enterprise. > > Here are my questions: > > 1. Where could I look for some resources on server sizing? Is it > any problem to support this number of users with a single server? > > 2. What do we need to do for our data network to make VOIP reliable? > QoS, basic traffic prioritization on the switch, vlan, ??? > > 3. What's the best way to integrate these 150 analog extensions? > I've seen interface boxes that usually come in 24-port sizes. Some > have an Ethernet/SIP interface to hook up to Asterisk, and others > have a T1 interface. What sounds best and is the most reliable? > > 4. What is a good company to contract with for emergency support? > Digium? > > 5. What are people doing to make VOIP phones resiliant in the face of > power outages? > > Is there anybody here that would be willing to serve as a reference > check for Asterisk should we pursue that path? > > Thanks, > > -- John > > > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call levels
Hello all. This is my first time with Asterisk, may be my question is fool. I have a two IP phone. I need that the first phone makes calls to local numbers only and the second phone make calls to all numbers. Somebody know the solution? Thanks a lot. Mariano ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] VoIP Gateway Providers
IAX.cc is what I use for my DID numbers. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeff Ramsey Sent: Thursday, November 17, 2005 1:23 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] VoIP Gateway Providers Hi, Can anyone recommend a good reputable VoIP gateway service provider that I can use with my Asterisk server in wa.us? All I can seem to find is VoIP service directly to the desk. I'd prefer a provider that can provide DID-type services, because that is my big selling point to the company. Thanks, Jeff Ramsey MIS Administrator Tubafor Mill, Inc. [EMAIL PROTECTED] 360.269.1650 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.13.3/173 - Release Date: 11/16/2005 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.13.3/173 - Release Date: 11/16/2005 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] HFC ISDN card and mISDN driver
> I'd say, go ahead and try the install-misdn script from beronet I get the same errors with the install-misdn script from Beronet: make[1]: Entering directory `/usr/src/linux-2.6.14-gentoo-r2' CC [M] /usr/src/install-misdn/mISDN/drivers/isdn/hardware/mISDN/avm_fritz.o /usr/src/install-misdn/mISDN/drivers/isdn/hardware/mISDN/avm_fritz.c: In function `fritzpci_probe': /usr/src/install-misdn/mISDN/drivers/isdn/hardware/mISDN/avm_fritz.c:1332: error: structure has no member named `slot_name' make[2]: *** [/usr/src/install-misdn/mISDN/drivers/isdn/hardware/mISDN/avm_fritz.o] Error 1 make[1]: *** [_module_/usr/src/install-misdn/mISDN/drivers/isdn/hardware/mISDN] Error 2 make[1]: Leaving directory `/usr/src/linux-2.6.14-gentoo-r2' make: *** [MISDN_MAKE_MODS] Error 2 And I actually have an AVM!Fritz PCI card in the machine. :) Any other ideas? Ta, Avi -- National Manager - Special Projects < Melbourne . Sydney . Canberra . Hobart . London /> 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, Victoria F: +61 (0) 3 9486 0611 3202 W: http://www.squiz.net/ .>> Open Source - Own it - Squiz.net ./> ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 1.2 Released!
On Thu, Nov 17, 2005 at 02:55:45PM +0800, Marcus Deluigi (intern) wrote: > Great! > Is there any chance someone tries to build a debian package for it? > :-D It's in the process of landing into Unstable. RC1 packages (and soon release packages) for Sarge are availble, see http://xorcom-rapid.berlios.de/ : deb http://rapid.dotsrc.org/ experimental/ deb-src http://rapid.dotsrc.org/ experimental/ -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco SIP translation-rule Question
I am dealing with a provider that has requested that whenever we terminate calls that we send a # at the end of the number to them. This is suppose to elmimite the post dial delay. Right now, I already change the numbers with the following to make them international: translation-rule 3 Rule 0 ^0 0 ANY international Rule 1 ^1 0111 ANY international Rule 2 ^2 0112 ANY international Rule 3 ^3 0113 ANY international Rule 4 ^4 0114 ANY international Rule 5 ^5 0115 ANY international Rule 6 ^6 0116 ANY international Rule 7 ^7 0117 ANY international Rule 8 ^8 0118 ANY international Rule 9 ^9 0119 ANY international Is there a way to add a suffix of "#" to every number that is dialed? Thanks! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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: 1.2 chan_modem not installing?
> > Try noload'ing all the chan_modem* modules as well > > > > noload => chan_modem.so > > noload => chan_modem_i4l.so > > noload => chan_modem_bestdata.so > > noload => chan_modem_aopen.so > > Or even better, pay close attention to the message at the end of 'make > install' that warns you that leaving those modules in place in > /usr/lib/asterisk/modules will likely cause problems. > > If the modules are not left there, then the 'noload' lines are not > necessary. Actually, that _is_ what got me in trouble. The warning at the end of make install was my hint to delete those files, and when I did, I could not restart *. I then noloaded chan_modem.so and had the same problem. So, uncommented the makefile entries and built/installed the files, then asterisk would start correctly. Noloading each of the four files shown above was the only thing that corrected it for me. The way these particular files were handled is different then in previous cvs-head changes, where essentially deleting the files was the expected action. Rich ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Sound Choppy
Hmm... I've also had some issues with choppy sounds, but my situation is somewhat weird. I've disabled APIC completely on the box, so not /proc/interrupts looks like This: CPU0 CPU1 0: 18394300 0 XT-PIC timer 1: 2 0 XT-PIC keyboard 2: 0 0 XT-PIC cascade 8: 1 0 XT-PIC rtc 10: 183925433 0 XT-PIC wct4xxp 11: 920477 0 XT-PIC eth0 14: 654191 0 XT-PIC ide0 15:136 0 XT-PIC ide1 NMI: 0 0 LOC: 18393988 18394010 ERR: 0 MIS: 0 However, the output of lspci -vb looks like this: 00:00.0 Host bridge: Intel Corp. E7501 Memory Controller Hub (rev 01) Flags: bus master, fast devsel, latency 0 Capabilities: [40] #09 [1105] 00:00.1 Class ff00: Intel Corp. E7000 Series Host RASUM Controller (rev 01) Subsystem: Intel Corp.: Unknown device 3425 Flags: fast devsel 00:02.0 PCI bridge: Intel Corp. E7000 Series Hub Interface B PCI-to-PCI Bridge (rev 01) (prog-if 00 [Normal decode]) Flags: bus master, 66Mhz, fast devsel, latency 32 Bus: primary=00, secondary=01, subordinate=03, sec-latency=0 Memory behind bridge: fc10-fc2f 00:02.1 Class ff00: Intel Corp. E7000 Series Hub Interface B RASUM Controller (rev 01) Subsystem: Intel Corp.: Unknown device 3425 Flags: fast devsel 00:1d.0 USB Controller: Intel Corp. 82801CA/CAM USB (Hub #1) (rev 02) (prog-if 00 [UHCI]) Subsystem: Intel Corp.: Unknown device 3425 Flags: bus master, medium devsel, latency 0, IRQ 10 I/O ports at 00:1d.1 USB Controller: Intel Corp. 82801CA/CAM USB (Hub #2) (rev 02) (prog-if 00 [UHCI]) Subsystem: Intel Corp.: Unknown device 3425 Flags: bus master, medium devsel, latency 0, IRQ 5 I/O ports at 6c20 00:1d.2 USB Controller: Intel Corp. 82801CA/CAM USB (Hub #3) (rev 02) (prog-if 00 [UHCI]) Subsystem: Intel Corp.: Unknown device 3425 Flags: bus master, medium devsel, latency 0, IRQ 10 I/O ports at 6c40 00:1e.0 PCI bridge: Intel Corp. 82801BA/CA/DB/EB PCI Bridge (rev 42) (prog-if 00 [Normal decode]) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=04, subordinate=04, sec-latency=32 I/O behind bridge: 7000-7fff Memory behind bridge: fc30-fdff 00:1f.0 ISA bridge: Intel Corp. 82801CA LPC Interface Controller (rev 02) Flags: bus master, medium devsel, latency 0 00:1f.1 IDE interface: Intel Corp. 82801CA Ultra ATA Storage Controller (rev 02) (prog-if 8a [Master SecP PriP]) Subsystem: Intel Corp.: Unknown device 3425 Flags: bus master, medium devsel, latency 0 I/O ports at I/O ports at I/O ports at I/O ports at I/O ports at 6c60 Memory at 4000 (32-bit, non-prefetchable) 00:1f.3 SMBus: Intel Corp. 82801CA/CAM SMBus Controller (rev 02) Subsystem: Intel Corp.: Unknown device 3425 Flags: medium devsel I/O ports at 1100 01:1c.0 PIC: Intel Corp. 82870P2 P64H2 I/OxAPIC (rev 04) (prog-if 20 [IO(X)-APIC]) Subsystem: Intel Corp.: Unknown device 3425 Flags: bus master, 66Mhz, fast devsel, latency 0 Memory at fc10 (32-bit, non-prefetchable) Capabilities: [50] PCI-X non-bridge device. 01:1d.0 PCI bridge: Intel Corp. 82870P2 P64H2 Hub PCI Bridge (rev 04) (prog-if 00 [Normal decode]) Flags: bus master, 66Mhz, fast devsel, latency 40 Bus: primary=01, secondary=02, subordinate=02, sec-latency=64 Capabilities: [50] PCI-X bridge device. 01:1e.0 PIC: Intel Corp. 82870P2 P64H2 I/OxAPIC (rev 04) (prog-if 20 [IO(X)-APIC]) Subsystem: Intel Corp.: Unknown device 3425 Flags: bus master, 66Mhz, fast devsel, latency 0 Memory at fc101000 (32-bit, non-prefetchable) Capabilities: [50] PCI-X non-bridge device. 01:1f.0 PCI bridge: Intel Corp. 82870P2 P64H2 Hub PCI Bridge (rev 04) (prog-if 00 [Normal decode]) Flags: bus master, 66Mhz, fast devsel, latency 40 Bus: primary=01, secondary=03, subordinate=03, sec-latency=48 Memory behind bridge: fc20-fc2f Capabilities: [50] PCI-X bridge device. 03:03.0 Communication controller: Xilinx Corporation: Unknown device 0314 (rev 01) Flags: bus master, medium devsel, latency 32, IRQ 10 Memory at fc20 (32-bit, non-prefetchable) 04:03.0 VGA compatible controller: ATI Technologies Inc Rage XL (rev 27) (prog-if 00 [VGA]) Subsystem: Intel Corp.: Unknown device 3425 Flags: bus master, stepping, medium devsel, latency 66, IRQ 11 Memory at fd00 (32-bit, non-prefetchable) I/O ports at 7000 Memory at fc34 (32-bit, non-prefetchable) Capabilities: [5c] Power Management version 2
Re: [Asterisk-Users] Mission-Critical Deployments
- Original Message - From: "John Goerzen" <[EMAIL PROTECTED]> To: Sent: Friday, November 18, 2005 2:37 AM Subject: [Asterisk-Users] Mission-Critical Deployments > I work for a company that is nearing the end-of-life on its existing > Nortel Meridian switch and is considering Asterisk. We have > approximately 200 existing extensions, and probably 150 out of those 200 > are using basic analog phones and would stay that way. The rest would > have VOIP phones at the desk. > > We're seriously considering switching to Asterisk. I've done quite a > bit of tinkering with Asterisk for my home, but I'm not certain about a > few aspects of how we might deploy Asterisk in the enterprise. > > Here are my questions: > > 1. Where could I look for some resources on server sizing? Is it >any problem to support this number of users with a single server? A decent dual xeon should be fine for that...or 2 or 3 smaller servers... (depends on the funtionality you need) > 2. What do we need to do for our data network to make VOIP reliable? >QoS, basic traffic prioritization on the switch, vlan, ??? If it's doable, a serapate data network for VOIP. A friends install moved to that after running VOIP on their main network, and it made a huge difference. YMMV. > 3. What's the best way to integrate these 150 analog extensions? >I've seen interface boxes that usually come in 24-port sizes. Some >have an Ethernet/SIP interface to hook up to Asterisk, and others >have a T1 interface. What sounds best and is the most reliable? Here I am going to disagree with you. Buy cheap IP phones. The hardware, setup and lack of functionality of analog extensions makes them a second choice for me. > 4. What is a good company to contract with for emergency support? >Digium? Find a local consultant. There are quite a few around... > 5. What are people doing to make VOIP phones resiliant in the face of >power outages? I have found the device called a UPS to be a useful in this regard, when hooked up to POE. > Is there anybody here that would be willing to serve as a reference > check for Asterisk should we pursue that path? I could. But I live in melbourne, australia. Which is not the same as austria. > > Thanks, > > -- John All just my 2 cents. PaulH ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] HFC ISDN card and mISDN driver
Hamish Whittal wrote: I am fighting with my ISDN HFC card to get the necessary compiled and working. I'd say, go ahead and try the install-misdn script from beronet (www.beronet.com/downloads), it might solve your problems. Then you'll use mISDN (and not zaptel) to use your card with *. Cheers, Kristof. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoIP Gateway Providers
Hi, Can anyone recommend a good reputable VoIP gateway service provider that I can use with my Asterisk server in wa.us? All I can seem to find is VoIP service directly to the desk. I'd prefer a provider that can provide DID-type services, because that is my big selling point to the company. Thanks, Jeff Ramsey MIS Administrator Tubafor Mill, Inc. [EMAIL PROTECTED] 360.269.1650 PGP.sig Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Sound Choppy
HI Brian, Thanks for the reply. I have checked that and it does not look IRQ is being shared. I have diabled, Serial, Parellel, USB, Floppy ! just to keep resources clean. lspci -vb 00:00.0 Host bridge: Intel Corporation 915G/P/GV/GL/PL/910GL Processor to I/O Controller (rev 04) Subsystem: Hewlett-Packard Company: Unknown device 300a Flags: bus master, fast devsel, latency 0 Capabilities: [e0] Vendor Specific Information 00:02.0 VGA compatible controller: Intel Corporation 82915G/GV/910GL Express Chipset Family Graphics Controller (rev 04) (prog-if 00 [VGA]) Subsystem: Hewlett-Packard Company: Unknown device 300a Flags: bus master, fast devsel, latency 0, IRQ 10 Memory at cfd0 (32-bit, non-prefetchable) I/O ports at 30c0 Memory at e000 (32-bit, prefetchable) Memory at cfd8 (32-bit, non-prefetchable) Capabilities: [d0] Power Management version 2 00:1c.0 PCI bridge: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6 Family) PCI Express Port 1 (rev 03) (prog-if 00 [Normal decode]) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=20, subordinate=20, sec-latency=0 Capabilities: [40] Express Root Port (Slot+) IRQ 0 Capabilities: [80] Message Signalled Interrupts: 64bit- Queue=0/0 Enable- Capabilities: [90] #0d [] Capabilities: [a0] Power Management version 2 Capabilities: [100] Virtual Channel Capabilities: [180] Unknown (5) 00:1c.1 PCI bridge: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6 Family) PCI Express Port 2 (rev 03) (prog-if 00 [Normal decode]) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=40, subordinate=40, sec-latency=0 Memory behind bridge: f020-f04f Capabilities: [40] Express Root Port (Slot+) IRQ 0 Capabilities: [80] Message Signalled Interrupts: 64bit- Queue=0/0 Enable- Capabilities: [90] #0d [] Capabilities: [a0] Power Management version 2 Capabilities: [100] Virtual Channel Capabilities: [180] Unknown (5) 00:1e.0 PCI bridge: Intel Corporation 82801 PCI Bridge (rev d3) (prog-if 01 [Subtractive decode]) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=05, subordinate=05, sec-latency=32 I/O behind bridge: 1000-1fff Memory behind bridge: f050-f07f Capabilities: [50] #0d [] 00:1f.0 ISA bridge: Intel Corporation 82801FB/FR (ICH6/ICH6R) LPC Interface Bridge (rev 03) Flags: bus master, medium devsel, latency 0 00:1f.1 IDE interface: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6 Family) IDE Controller (rev 03) (prog-if 8a [Master SecP PriP]) Subsystem: Hewlett-Packard Company: Unknown device 300a Flags: bus master, medium devsel, latency 0, IRQ 5 I/O ports at 30c8 I/O ports at 30e8 I/O ports at 30d0 I/O ports at 30ec I/O ports at 30a0 00:1f.2 IDE interface: Intel Corporation 82801FB/FW (ICH6/ICH6W) SATA Controller (rev 03) (prog-if 8f [Master SecP SecO PriP PriO]) Subsystem: Hewlett-Packard Company: Unknown device 300a Flags: bus master, 66Mhz, medium devsel, latency 0, IRQ 5 I/O ports at 30d8 I/O ports at 30f0 I/O ports at 30e0 I/O ports at 30f4 I/O ports at 30b0 Capabilities: [70] Power Management version 2 05:09.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b119:0001 Flags: bus master, medium devsel, latency 32, IRQ 11 I/O ports at 1000 Memory at f050 (32-bit, non-prefetchable) Capabilities: [40] Power Management version 2 40:00.0 Ethernet controller: Broadcom Corporation NetXtreme BCM5751 Gigabit Ethernet PCI Express (rev 01) Subsystem: Hewlett-Packard Company: Unknown device 3005 Flags: bus master, fast devsel, latency 0, IRQ 5 Memory at f040 (64-bit, non-prefetchable) Capabilities: [48] Power Management version 2 Capabilities: [50] Vital Product Data Capabilities: [58] Message Signalled Interrupts: 64bit+ Queue=0/3 Enable- Capabilities: [d0] Express Endpoint IRQ 0 Capabilities: [100] Advanced Error Reporting Capabilities: [13c] Virtual Channel -Original message- From: "Brian M. Arlinghaus" [EMAIL PROTECTED] Date: Fri, 18 Nov 2005 00:04:17 +0300 To: "Asterisk Users Mailing List - Non-Commercial Discussion"asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Sound Choppy > Is your Digium card sharing an IRQ with anything else? > > Use the lspci -vb command to show which device is using which IRQ. If, > for example, a network card is using the same one as the Digium card, you > will quite possibly get choppy sound and echo. > > You can usually change IRQs in the BIOS setup. In some
Re: [Asterisk-Users] Bristuff / Junghanns / Customer Service
Frederic Steinfels wrote: Nethertheless I have found SEVERAL errors that lead to complete hangups and core dumps. I was running gdb for KPJ, writing extensive bug reports and some of those bugs were fixed. Last January I told KPJ that I can still not use my Simens Gigagaset cordless phones and sent him some bug Hi, I've been using some quadBRI's, never had to much problems with them. (I have some echo issues, but I hope these'll be resolved on the next driver update, when we can use new echo can's..) So, I can't confirm the bug(s) you're experiencing, but I must say that I didn't have any hangups or dumps ever.. The driver seems to be stable, but maybe we are using it in a very different situation.. Maybe it's a possibility to try the chan_misdn driver (www.beronet.com has the instructions), it could help you.. they produce approx the same cards but address it through mISDN instead of zaptel.. Cheers! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Can anyone explain reason for this echo
Eric Bishop <[EMAIL PROTECTED]> wrote: >If I call our Asterisk box via Disa and then place a call to one of the >problem analogue numbers (native Zap bridge) I don't get any echo. So the >echo seems to occur only when using a SIP handset and making a call to an >analogue number. The echo is probably always there. You only notice it with the SIP phone because of the additional latency that this introduces. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HFC ISDN card and mISDN driver
Hi Folks, I am fighting with my ISDN HFC card to get the necessary compiled and working. I am running Ubuntu Breezy, kernel 2.6.14 (self compiled). Based upon posts on the local VoIP site (www.voipinfo.co.za) and www.voip-info.org, it seems that I should get Jolly's mISDN driver for these cards. Got it. Also got the ones from cvs at isdn4linux. Using Jolly's, I am not able to re-compile the kernel. It stops with the following error: make[1]: Entering directory `/usr/src/linux-2.6.14' CHK include/linux/version.h CC [M] drivers/isdn/hardware/mISDN/avm_fritz.o drivers/isdn/hardware/mISDN/avm_fritz.c: In function ‘fritzpci_probe’: drivers/isdn/hardware/mISDN/avm_fritz.c:1332: error: ‘struct pci_dev’ has no member named ‘slot_name’ make[5]: *** [drivers/isdn/hardware/mISDN/avm_fritz.o] Error 1 make[4]: *** [drivers/isdn/hardware/mISDN] Error 2 make[3]: *** [drivers/isdn/hardware] Error 2 make[2]: *** [drivers/isdn] Error 2 make[1]: *** [drivers] Error 2 make[1]: Leaving directory `/usr/src/linux-2.6.14' make: *** [stamp-build] Error 2 I will try the cvs stuff tomorrow AM. Then there's the question of whether to install the zaphfc stuff (bristuff). I noted that to get this installed one needs to install libpri - why? Surely we are not doing a pri install, but a bri install. I am mightly confused at the mo. Perhaps someone who has done this can just summarise what all these bits are and what I should install. Thanks in advance, Hamish ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] CVS v1-2-0 make problems?
On 11/17/05, Kristian Kielhofner <[EMAIL PROTECTED]> wrote: > Asterisk-users, > >Has anyone else had problems with the v1-2-0 CVS rev? Here's the deal: > >LATE last night I checkout out 1.2.0 with CVS: > > rm -rf asterisk zaptel libpri > cvs co -r "v1-2-0" zaptel > cvs co -r "v1-2-0" libpri > cvs co -r "v1-2-0" asterisk > >zaptel and libpri build fine. Asterisk, however, seems to get stuck in > a infinite loop while (guessing) determining version. The loop occurs > when using cmp to check version.h and version.h.tmp. It goes on > forever, forever, and forever. > >However, using the 1.2.0 tarballs work perfectly, for libpri, zaptel, > and asterisk. > >Yes, this is for AstLinux and it is using my cross-build environment. > (Which has worked very well for tracking CVS HEAD at build.astlinux.org, > and as mention before can build using the 1.2.0 tarballs). > >I'd have more time to dig deep into it, but I am just trying to get a > 1.2 build of AstLinux done. I somewhat foolishly promised one by > tomorrow :). > >Anyone else experiencing this? Are my CVS commands wrong? What's up? > > Thanks in advance, and a HUGE thank you to everyone at Digium for > getting 1.2 out! > Kristian, No. You're not the only one to be having that problem, and you're correct that it is only a problem with checked out via CVS versions of Asterisk. I believe the consensus/solution was that with regard to the release, you can avoid the problem by using the tarball and going forward the dev branch is going to be using SVN which avoids the problem all together. Thank you for AstLinux! It rocks! :) BJ -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] SIP/H.323 HardPhones
Use SIP. PaulH - Original Message - From: <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Friday, November 18, 2005 7:53 AM Subject: [Asterisk-Users] SIP/H.323 HardPhones > hi, > > Do anyone know a low-cost, simple SIP or H.323 hardphone that works well > with Asterisk? > > Jan > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Sound Choppy
Is your Digium card sharing an IRQ with anything else? Use the lspci -vb command to show which device is using which IRQ. If, for example, a network card is using the same one as the Digium card, you will quite possibly get choppy sound and echo. You can usually change IRQs in the BIOS setup. In some cases, you may have to disable built-in hardware. I, for example, have a Dell PowerEdge 2850 with two built-in Intel NICs. I have had to disable both NICs and the USB controller and install another NIC in a PCI slot so that my two Digium cards are not sharing any IRQs. Regards, Brian - Original Message - From: "Abdock" <[EMAIL PROTECTED]> To: Sent: Thursday, November 17, 2005 2:46 PM Subject: [Asterisk-Users] Sound Choppy Hello, I have a calling server, dialing though the telco lines using Digium card, teh call gets connected but if one side speaks little loud or if they speak simultaneous then the voice starts to break. Using g729 codec - IAX trunk - international gateway. Anyone can hep ? Thanks. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CVS v1-2-0 make problems?
Asterisk-users, Has anyone else had problems with the v1-2-0 CVS rev? Here's the deal: LATE last night I checkout out 1.2.0 with CVS: rm -rf asterisk zaptel libpri cvs co -r "v1-2-0" zaptel cvs co -r "v1-2-0" libpri cvs co -r "v1-2-0" asterisk zaptel and libpri build fine. Asterisk, however, seems to get stuck in a infinite loop while (guessing) determining version. The loop occurs when using cmp to check version.h and version.h.tmp. It goes on forever, forever, and forever. However, using the 1.2.0 tarballs work perfectly, for libpri, zaptel, and asterisk. Yes, this is for AstLinux and it is using my cross-build environment. (Which has worked very well for tracking CVS HEAD at build.astlinux.org, and as mention before can build using the 1.2.0 tarballs). I'd have more time to dig deep into it, but I am just trying to get a 1.2 build of AstLinux done. I somewhat foolishly promised one by tomorrow :). Anyone else experiencing this? Are my CVS commands wrong? What's up? Thanks in advance, and a HUGE thank you to everyone at Digium for getting 1.2 out! -- Kristian Kielhofner ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP/H.323 HardPhones
hi, Do anyone know a low-cost, simple SIP or H.323 hardphone that works well with Asterisk? Jan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] newbie questions
[EMAIL PROTECTED] wrote: hi, My second question: for a branch office of about 20 people, which E1 card do you advise? Would the TE210P be a good choice? (number of concurrent calls would be max 10 for now) Why? An E1 has 30 lines, so you would be perfect with a TE110P. Connecting an E1 to a company PABX is however an expensiveoption, so you might want to compare the prices with 10 analogue lines or maybe 5 BRI lines. I would not let the price of hardware decide this because you will need to pay a fixed cost per month for PSTN lines, so check these prices first. Asterisk is scalable in the sence that you can add more later if you have an available PCI slot. Even though they don't appear to be shipping yet, don't forget the TDM2400P's from Digium. Up to 24 FXO or FXS ports per full length PCI card. -- Christopher L. Wade, CCNA, CCDA, CQS-CIPCES, CQS-CWLSS ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] suggestions for hard phones?
Yes the SPA-941 has STUN support from the SIP tab in the admin interface. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Sizemore Sent: Thursday, November 17, 2005 12:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] suggestions for hard phones? Does the SPA-941 support stun? Kerry Garrison wrote: > My two favorite phones (in order) are: > > Linksys SPA-941 > http://voipspeak.net/index.php?option=com_content&task=view&id=41 > > Grandstream GXP-2000 > http://voipspeak.net/index.php?option=com_content&task=view&id=25 > > The problem is the change of credentials, that is an interesting > issue. With either phone, you can have multiple accounts assigned to > it and the user can set the DO-Not-Disturb for their line when they > come and go. That is probably the easiest way to accomplish it. The > second would be a single extension for the actual phone and then a > call queue for each person. When each person comes into work, they log > into their own call queue. The first approach is easier to implement. > > Kerry Garrison > http://voipspeak.net > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of John > Fraser > Sent: Thursday, November 17, 2005 3:44 AM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] suggestions for hard phones? > > Hi all, > > I am looking for SIP hard phones to use in a call center. > The feature that I need the most is quick change of logon credentials > as we run 3 shifts. each agent will have their own extension number and password. > any suggestions would be greatly appreciated. > > thank you > John Fraser > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > No virus found in this incoming message. > Checked by AVG Free Edition. > Version: 7.1.362 / Virus Database: 267.13.3/173 - Release Date: > 11/16/2005 > > > -- > No virus found in this outgoing message. > Checked by AVG Free Edition. > Version: 7.1.362 / Virus Database: 267.13.3/173 - Release Date: > 11/16/2005 > > ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.13.3/173 - Release Date: 11/16/2005 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.13.3/173 - Release Date: 11/16/2005 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] newbie questions
hi, My second question: for a branch office of about 20 people, which E1 card do you advise? Would the TE210P be a good choice? (number of concurrent calls would be max 10 for now) Why? An E1 has 30 lines, so you would be perfect with a TE110P. Connecting an E1 to a company PABX is however an expensiveoption, so you might want to compare the prices with 10 analogue lines or maybe 5 BRI lines. I would not let the price of hardware decide this because you will need to pay a fixed cost per month for PSTN lines, so check these prices first. Asterisk is scalable in the sence that you can add more later if you have an available PCI slot. Jan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Can anyone explain reason for this echo
Hi, Not uncommon what you are seeing. Try playing with your echo can. type Mark2 etc. on your analogue zaps also try tweaking your tx & rx gain using ztmonitor to view your incoming levels. Analogue is def. a lot more prone to echo than ISDN so get tweeking ; ) Cheers, MICHAEL TOOP Tel > 011 602 9309 Fax > 011 656 1342 Mobile > 083 364 2370 Web > www.bizcall.co.za Eric Bishop wrote: Our configuration is as follows: SIP phones -> TE410P -> PSTN When a SIP handset makes a call to other ISDN numbers - no problem. When a SIP handset make a call to analogue numbers - echo. I know for certain that the problem is at our end. Why? If I call our Asterisk box via Disa and then place a call to one of the problem analogue numbers (native Zap bridge) I don't get any echo. So the echo seems to occur only when using a SIP handset and making a call to an analogue number. Can anyone provide a logical explanation for this? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] newbie questions
Hi all I am new to this whole field, being it PSTN or voIP. I am currently reading the "Switching to VoIP" and "Asterisk: The Future of Telephony", so hopefully, I will be less clueless soon :) My first question: if I buy a Wildcard TDM400P, with one X100M and three S100M modules, I would be able to have 1 telephone number given out by my company to come in to my asterisk server, and I could plug in 3 analog phones onto that card, am I correct? Hence, do we have a 1-to-1 relationship here for either modules? My second question: for a branch office of about 20 people, which E1 card do you advise? Would the TE210P be a good choice? (number of concurrent calls would be max 10 for now) Why? Thank you all. Cheers fred signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk 1.2 Change in: agi_channel
I am testing out Asterisk 1.2 released today and got the following problem. In my AGI script when running Asterisk 1.2 I get the following AGI variable: agi_channel: IAX2/80.229.221.228:4569-2 In all earlier versions of Asterisk including Beta 1.2 I got the following: agi_channel: IAX2/[EMAIL PROTECTED]:4569-3 in the new CSV logs we have the same. "","70200","70103","default","""Are"" <70200>","IAX2/80.229.221.228:4569-2" 70104 is my IAX user. It looks like there is no clear way to extract the IAX user executing the call anymore. I have not been able to find this change documented anywhere. Is it by design or a bug?-- Are Casillahttp://astartelecom.com - Independent VOIP Telecoms Broker. Asterisk Consultants http://astbill.com - Open Source Billing, Routing and Management software for Asterisk and VOIP AstBill DEMO: http://demo.astbill.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] suggestions for hard phones?
Does the SPA-941 support stun? Kerry Garrison wrote: My two favorite phones (in order) are: Linksys SPA-941 http://voipspeak.net/index.php?option=com_content&task=view&id=41 Grandstream GXP-2000 http://voipspeak.net/index.php?option=com_content&task=view&id=25 The problem is the change of credentials, that is an interesting issue. With either phone, you can have multiple accounts assigned to it and the user can set the DO-Not-Disturb for their line when they come and go. That is probably the easiest way to accomplish it. The second would be a single extension for the actual phone and then a call queue for each person. When each person comes into work, they log into their own call queue. The first approach is easier to implement. Kerry Garrison http://voipspeak.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Fraser Sent: Thursday, November 17, 2005 3:44 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] suggestions for hard phones? Hi all, I am looking for SIP hard phones to use in a call center. The feature that I need the most is quick change of logon credentials as we run 3 shifts. each agent will have their own extension number and password. any suggestions would be greatly appreciated. thank you John Fraser ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.13.3/173 - Release Date: 11/16/2005 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.13.3/173 - Release Date: 11/16/2005 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Speech recognition or TTS with Asterisk?
John Brookes wrote: > Paul, > Can you say more about how I could get started on this? > I have been looking at Cepstral for TTS, but any will do. > Can this be implemented in Java? > JB > I provided the link for phpagi. Install it and install festival. Set up an extension going to the weather.php demo. If you are running debian 3.1 it should be workable. If you are running another distro there may be differences. If you want this running real soon contact me offlist about paid services. Otherwise you will be reading and learning. Free help via the list happens while I am taking coffee/donut breaks so patience is a needed virtue. > > - Original Message - From: "Paul" <[EMAIL PROTECTED]> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Thursday, November 17, 2005 11:29 AM > Subject: Re: [Asterisk-Users] Speech recognition or TTS with Asterisk? > > >> John Brookes wrote: >> >>> Hello, >>> I am interested in TTS with Asterisk. >>> Anyone implemented this port? >>> Thanks in adbvance, >>> John B >> >> >> Yes. I did it when I installed the phpagi stuff including the weather >> demo. It worked so I went ahead and strted playing around with it and >> was able to change things. >> >> http://phpagi.sourceforge.net/ >> >> I did all this using debian stable (aka sarge) linux. Only thing >> non-debian is files added to /usr/share/asterisk/agi-bin/ and you might >> have to >> >> Of course I had to edit extensions.conf >> >> exten => 17,1,agi(weather.php) >> exten => 18,1,agi(dtmf.php) >> exten => 19,1,agi(input.php) >> exten => 20,1,agi(my_ip.php) >> >> I haven't had time to put this on the server running 1.2 rc2 yet. >> >> ___ >> --Bandwidth and Colocation sponsored by Easynews.com -- >> >> Asterisk-Users mailing list >> Asterisk-Users@lists.digium.com >> http://lists.digium.com/mailman/listinfo/asterisk-users >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] calling to asterisk and listening to music (GSM) -->>Anyone, please?????
> Hi all! > > I'm trying to play some music from asterisk, and when I call to the PBX > from a GSM mobile phone, the more I speak while hearing the music, the > worst is the quality of the music I hear... My audio is at 8Khz, > 16bits/sample. > > I've tried different codecs for asterisk, but results are the same... > > If I call to the PBX from a conventional phone, I can speak while hearing > the music, with no quality loss... > > Any idea? Does it have anything with the mobile-GSM standard used for > coding? > > What to do? > > thanks, > > -e > ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sound Choppy
Hello, I have a calling server, dialing though the telco lines using Digium card, teh call gets connected but if one side speaks little loud or if they speak simultaneous then the voice starts to break. Using g729 codec - IAX trunk - international gateway. Anyone can hep ? Thanks. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 hobby box
Logan wrote: Hi everyone! Okay. I was reading on the voip-info.org about FXO and FXS. Is it possible just to get a card with FXO and FXS together? I know Digium sells them, but as I've said, I'm looking to spend too much. Thanks for everyone's input! Logan. FXO is easy, but FXS is more expensive. You'll likely need two cards (one for each). You can get $10 FXO cards on ebay, but something seems to be buggy as heck with those. I have problems with them nearly daily which requires reboots. You could try finding an Internet Phonejack (I think that's the FXS one). I bought one a while ago and it wasn't too expensive (compared to the Digium stuff). Not sure if the company exists any more. Phil ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Bristuff / Junghanns / Customer Service
Hi Frederic, Not to start some flame war here, but I've always known the Junghanns people to be quite cooperative, although it is a shame that they don't have two Klaus'es around there, since one is just simply too busy :) Florian ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip.conf settings for voip.net / broadvox?
Has anybody succeeded in making outbound/inbound SIP connections to voip.net (or broadvox, which voip.net is just a reseller of)? I can make calls fine through their ATA, but my control panel password doesn't seem to be my SIP credential. - a -- PGP/GPG: 5C9F F366 C9CF 2145 E770 B1B8 EFB1 462D A146 C380 Q: "Won't the pendulum swing back?" A: "It has never been a pendulum. Think tectonic plates instead." -- from patrick.net ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 64bit libs in /usr/lib
What is the proper way to use the Makefile for 1.2.0 so that my 64bit libs get installed into the proper place such as /usr/lib64 ? Right now they are being installed in /usr/lib and it is making packaging this software a pain. -- Jesse Keating GameHouse -- Systems Engineer ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Users Groups - Southern California?
Does anyone know of an Asterisk Users Group in the Orange County California area or is there enough interest in starting one up? Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.13.3/173 - Release Date: 11/16/2005 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Speech recognition or TTS with Asterisk?
Paul, Can you say more about how I could get started on this? I have been looking at Cepstral for TTS, but any will do. Can this be implemented in Java? JB - Original Message - From: "Paul" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Thursday, November 17, 2005 11:29 AM Subject: Re: [Asterisk-Users] Speech recognition or TTS with Asterisk? John Brookes wrote: Hello, I am interested in TTS with Asterisk. Anyone implemented this port? Thanks in adbvance, John B Yes. I did it when I installed the phpagi stuff including the weather demo. It worked so I went ahead and strted playing around with it and was able to change things. http://phpagi.sourceforge.net/ I did all this using debian stable (aka sarge) linux. Only thing non-debian is files added to /usr/share/asterisk/agi-bin/ and you might have to Of course I had to edit extensions.conf exten => 17,1,agi(weather.php) exten => 18,1,agi(dtmf.php) exten => 19,1,agi(input.php) exten => 20,1,agi(my_ip.php) I haven't had time to put this on the server running 1.2 rc2 yet. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
