Re: [Asterisk-Users] E3 card
Eric Wieling aka ManxPower wrote: Kevin P. Fleming wrote: Tamas J wrote: Does anybody know what will be the around price for the announced E3 card from Digium? When is it planned to be ready? Pricing and release date have not been announced at this time. The product has also been removed from Digium's web site, as far as I can tell. Perhaps they decided the market is not that big, and reduced the priority of the card. If so, they are not the only ones to do this. Regards, Steve ___ 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] Does PCI Developer Kit work with kernel 2.6
I just got it working with a 2.6 kernel without any problems, although I'm having a little bit of trouble setting the callerid on it Keith On Jul 1, 2005, at 11:00 PM, Michael Jia wrote: Hi, In digium website. http://store.yahoo.com/asteriskpbx/newitastdmde.html It is said Dev Kit PCI card works with 2.4 kernel. I am wondering if it is also working with 2.6 kernel? Anyone knows? Thanks Michael ___ 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 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 spandsp
Theodore Cekan wrote:
I got this working on Fedora Core 3 i386, but it does not work on
CentOS x86_64. Anything I should know about x86_64?
I have reports from people who say spandsp works OK on x86_64. I had to
do a few fixed for them to make it build, which are in any recent copy
of spandsp. I just acquired a new X2 machine, so I will be doing more
extensive testing of my software built for x86_64 over the next few weeks.
Regards,
Steve
Thanks.
Ted
Theodore Cekan wrote:
Hello,
I am trying to setup spandsp for the first time. I have spandsp
0.0.2pre18 and * v1.0.8 patched. For testing I have this extension:
exten => 1234,1,rxfax(/tmp/testfax.tif)
* answers the call, but makes no fax tones. I get this logged:
-- Executing RxFAX("SIP/x.x.x.x-00668660", "/tmp/testfax.tif") in
new stack
then nothing, silence, no communication.
Any help is appreciated.
Ted
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[Asterisk-Users] Does PCI Developer Kit work with kernel 2.6
Hi, In digium website. http://store.yahoo.com/asteriskpbx/newitastdmde.html It is said Dev Kit PCI card works with 2.4 kernel. I am wondering if it is also working with 2.6 kernel? Anyone knows? Thanks Michael ___ 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] TDM11B Dev Kit PCI + Asterisk CVS Head
I just downloaded the latest asterisk and zaptel from cvs and I'm
having trouble setting the callerid on the fxs interface. From what
I've read you should be able to set it in the zapata.conf but I can't
get it to work. I had a cheap X100P clone but recently bought the Dev
Kit PCI . I've built a seperate server and have everything up and
running except everytime I go to check my voicemail I have to enter
my extension each time. Here is the exten for voicemail that I have
set up.
[vmail-main]
exten => 580,1,Wait,1
exten => 580,2,Answer
exten => 580,3,VoiceMailMain([EMAIL PROTECTED])
exten => 580,4,Wait,2
exten => 580,5,hangup
I've set up a test extension that will say cid
exten => 222,1,Wait,2
exten => 222,2,SayDigits(${CALLERIDNUM})
exten => 222,3,Hangup
It works with a sip soft phone I've tested it with.
Here is the zapata.conf
[channels]
busydetect=no
musiconhold=default
transfer=yes
cancallforward=yes
usecallerid=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
immediate=no
context=internal
signalling=fxo_ks
channel=1
context=pstn-in
callerid="Keith" <100>
signalling=fxs_ks
channel=4
callerid=asrecieved
Any Idea what I'm doing wrong?
Keith
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Re: [Asterisk-Users] asterisk showing more than once on ps
--- [EMAIL PROTECTED] wrote: > Actually this is on an enterprise system - I have > never seen more > than one. Just checked now and there were several > calls up and only > one ps. > > > On Jul 1, 2005, at 8:33 PM, Luki wrote: > > >> Do not know why, but have noticed redhat = 1, and > debian = many > >> > > Not quite. RedHat Enterprise also = many at times, > depending on number > > of concurrent calls; usually one when idle. Maybe > it has something to > > do with kernel 2.4 vs 2.6 and how threads show up > in ps. > > > > --Luki > > ___ > > 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 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 > What I've noticed (running the auto-start script on White-Box Enterprise Linux ... basically the same as RedHat Enterprise Linux of some vintage) is that when the auto-start executes safe_asterisk, I see multiple instances of asterisk under ps. However, if I use "asterisk -r" to connect, then stop and restart safe_asterisk manually, ps drops to showing exactly 1 instance of asterisk running. For what it's worth ... Maya Yahoo! Sports Rekindle the Rivalries. Sign up for Fantasy Football http://football.fantasysports.yahoo.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] Epia C3 Linux
Fedora Core what? Core 2 won't install, because they seem to have built the installer kernel for i686 by mistake. Core 4 won't install for some reason - the install begins, but fouls up in the middle. Core 3 installs without trouble. After installation you need to disable cpuspeed, or something goes wrong every day or two. Then things run OK. Regards from someone who has been find this out by trial and error in the past week, Steve Wiley Siler wrote: I just tried Fedora Core CD1 and it died on autopartitioning…. W *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Michael Stahl *Sent:* Friday, July 01, 2005 2:00 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* RE: [Asterisk-Users] Epia C3 Linux It installed directly from the FC3 dvd, no changes...no external drivers required *From:* Wiley Siler [mailto:[EMAIL PROTECTED] *Sent:* Friday, July 01, 2005 2:42 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* RE: [Asterisk-Users] Epia C3 Linux Did it require any special work or did you just download the ISO for FC3 and install? Thanks, Wiley *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Michael Stahl *Sent:* Friday, July 01, 2005 11:19 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* RE: [Asterisk-Users] Epia C3 Linux I have Fedora Core 3 running great on an Epia mobo *From:* Wiley Siler [mailto:[EMAIL PROTECTED] *Sent:* Friday, July 01, 2005 12:54 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [Asterisk-Users] Epia C3 Linux Anyone know a good distro for an Epia Mobo with the C3 chip? I have been trying to get Debian and Gentoo installed (new to me) and so far having little luck. Does anyone know a good install for this processor/mobo combo? Thanks Wiley ___ 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] AMP/A@H (asterisk at home) custom incoming routing
A few things to followup on my earlier post: 1.) Definitely put the [tdm-in] context in the file "/etc/asterisk/ extensions_custom.conf". That way your changes will not get overwritten. 2.) I am still unable to make call waiting on the incoming ZAP line work, b/c I have not thought up a good way to make this happen. hasn't anyone done this before? 3.) When setting up the ZAP trunk, I found it usefull to put "w" in the dial prefix field to force the system to wait for the dial tone. If I didn't do that, I could not call out on the ZAP Channel. Tom On Jun 29, 2005, at 12:46 AM, Tom Rymes wrote: Folks, First off, this is messy, and I hope someone will be kind enough to help me clean this up (the part added to extensions_additional.conf). You've been warned! For those of your using AMP or [EMAIL PROTECTED], there has been a lot of talk about how to route incoming calls to different places based on which trunk is ringing. The standard answer is that you can only do this by using DIDs, which is all fine and good, unless you are using a plain old copper line that doesn't support DID. Anyhow, I have figured out how to make a call that comes in on a specific ZAP channel ring at a specific extension (not that it was brain surgery). I'm not certain if it would be better to use the file extensions_custom.conf instead of extensions_additional.conf, does anyone know? [snip] ___ 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] passing through MWI info from SBC
Chris Gamble wrote: Does this just sound worse than it is? "With SBC you are out of luck, since Asterisk doesn't detect dialtone (it dials blind, sometimes too quickly for the CO to catch the first digit, resulting in wrong numbers )) or stutter dialtone either, and reportedly has had any indication of the DC status of a POTS line removed due to problems." IF one is working only in the VOIP/SIP/IAX world, then it matters not, but IF one has to construct a working interface with analog lines, either PSTN or PBX, then it is probably even worse than it sounds Someone along the way didn't bother to do their homework, or learn much from even the simple modem interfaces of the last 15-20 years. There is no excuse for not listening for dialtone, resulting in blind dialing , forcing waits( w) to be inserted into the dial string, an imprecise cure at best. Many need to migrate from POTS in stages, for many reasons, so to sluff off this interface certainly doesn't help acceptance for Asterisk. As to FSK MW indication, I would like to learn more. Is this an expansion of the Bell 202 modem CallerID communication, or something entirely different. That would solve one problem for some, where it is provided by the telco. John Novack ___ 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 showing more than once on ps
Actually this is on an enterprise system - I have never seen more than one. Just checked now and there were several calls up and only one ps. On Jul 1, 2005, at 8:33 PM, Luki wrote: Do not know why, but have noticed redhat = 1, and debian = many Not quite. RedHat Enterprise also = many at times, depending on number of concurrent calls; usually one when idle. Maybe it has something to do with kernel 2.4 vs 2.6 and how threads show up in ps. --Luki ___ 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 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 showing more than once on ps
> Do not know why, but have noticed redhat = 1, and debian = many Not quite. RedHat Enterprise also = many at times, depending on number of concurrent calls; usually one when idle. Maybe it has something to do with kernel 2.4 vs 2.6 and how threads show up in ps. --Luki ___ 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] Computer to use {Scanned}
I'm using a cheap PCChip Mboard and an AMD 2800 Simpron 1Gb of RAM in a cheap 2U case. I have two X100P and 5 Broadvoice lines. I have about 20 IP phones and softphones. Works great. 73's David KE6UPI On Fri, 2005-07-01 at 09:36 -0700, Robert Goodyear wrote: > On Jul 1, 2005, at 4:03 AM, Eric Wieling aka ManxPower wrote: > > > Robert Goodyear wrote: > > > >> I'm sure you really only want to know about the absence of problems. > >> From watching this list for 6 months it seems the SuperMicro products > >> are most lauded and have exhibited no hardware conflicts. Various > >> votes on Dell products, so you're probably best to stay away, even > >> though I've got five installs with TE110Ps in them that have never > >> missed a beat -- Dimension boxes, not PowerEdge. > > > > The SuperMicro Xeon board we tried failed miserably with both the > > T100P and TE110P. It had the ServerWorks IDE Chipset, which I suspect > > was the problem. > > > > -- > > Eric Wieling * BTEL Consulting * 504-210-3699 x2120 > > Bummer! I thought I'd heard all good things about them... sorta like > VoIP providers; as soon as everyone agrees things are OK, something > goes awry! > > -Rob. > > ___ > 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 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 showing more than once on ps
Do not know why, but have noticed redhat = 1, and debian = many On Jul 1, 2005, at 7:18 PM, Mark Charlton wrote: On 7/2/05, Anton Krall <[EMAIL PROTECTED]> wrote: Guys. Anybody know why sometimes on some servers Asterisk shows more than once while doing a ps? [EMAIL PROTECTED] akrall]# ps -ax|grep asterisk 20555 ?S 0:00 /bin/sh /usr/sbin/safe_asterisk 20557 ?S 0:00 asterisk -vvvg -c 20558 ?S 0:00 asterisk -vvvg -c 20560 ?S 0:00 asterisk -vvvg -c 20561 ?S 0:00 asterisk -vvvg -c 20562 ?S 0:00 asterisk -vvvg -c 20564 ?S 0:00 asterisk -vvvg -c 20565 ?S 0:00 asterisk -vvvg -c 20566 ?S 0:02 asterisk -vvvg -c 20567 ?S 0:00 asterisk -vvvg -c 21958 ?S 0:00 asterisk -vvvg -c 30470 pts/2S 0:00 grep asterisk Im running asterisk via the start/stop script on init.d On my other servers, it only shows once... I would guess its the use of the -c flag to run it From: http://www.voip-info.org/tiki-index.php?page=Asterisk+options # asterisk -vvc:Start Asterisk PBX, don't go background have you been monitoring it from the command line. I use -r to get the background version. Just a guess. Mark ___ 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 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] MOH - request to schdule in the past
I have googled this to death, and all I get are reference to the MoH
needing a Zaptel timing source, and then people saying no they don't
any more.
-- Set Response Timeout to 2
-- Executing BackGround("SIP/211-57ba", "my-greeting") in new stack
-- Playing 'my-greeting' (language 'en')
== CDR updated on SIP/211-57ba
-- Executing MusicOnHold("SIP/211-57ba", "") in new stack
-- Started music on hold, class 'default', on SIP/211-57ba
-- Stopped music on hold on SIP/211-57ba
== Spawn extension (webtree, 7, 1) exited non-zero on 'SIP/211-57ba'
Jul 1 20:19:32 NOTICE[1376]: res_musiconhold.c:309 monmp3thread:
Request to schedule in the past?!?!
It seems to work for a little while after a reboot then dies a death.
The extension just gets dead air, and the error message is on the CLI,
it always comes up after a call, and seemingly randomly inbetween.
Any suggestions where to look?
extensions.conf has
exten => _[123],3,Dial(SIP/200&SIP/202&SIP/210&SIP/211&SIP/212|40|m)
and
exten => 7,1,musiconhold()
both do the same thing.
musiconhold.conf has
[classes]
default => mp3:/var/lib/asterisk/mohmp3,-z
;loud => mp3:/var/lib/asterisk/mohmp3
;random => quietmp3:/var/lib/asterisk/mohmp3,-z
Thanks
Mark
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Re: [Asterisk-Users] asterisk showing more than once on ps
On 7/2/05, Anton Krall <[EMAIL PROTECTED]> wrote: > Guys. > > Anybody know why sometimes on some servers Asterisk shows more than once > while doing a ps? > > [EMAIL PROTECTED] akrall]# ps -ax|grep asterisk > 20555 ?S 0:00 /bin/sh /usr/sbin/safe_asterisk > 20557 ?S 0:00 asterisk -vvvg -c > 20558 ?S 0:00 asterisk -vvvg -c > 20560 ?S 0:00 asterisk -vvvg -c > 20561 ?S 0:00 asterisk -vvvg -c > 20562 ?S 0:00 asterisk -vvvg -c > 20564 ?S 0:00 asterisk -vvvg -c > 20565 ?S 0:00 asterisk -vvvg -c > 20566 ?S 0:02 asterisk -vvvg -c > 20567 ?S 0:00 asterisk -vvvg -c > 21958 ?S 0:00 asterisk -vvvg -c > 30470 pts/2S 0:00 grep asterisk > > Im running asterisk via the start/stop script on init.d > > On my other servers, it only shows once... > > I would guess its the use of the -c flag to run it From: http://www.voip-info.org/tiki-index.php?page=Asterisk+options # asterisk -vvc:Start Asterisk PBX, don't go background have you been monitoring it from the command line. I use -r to get the background version. Just a guess. Mark ___ 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] Got SIP response 481 "Invalid CSeq Number"backfrom X.X.X.X
I had the same problem , don't remember how I resolved it. What is your dial-peer and sip settings on AS5300? I will compare it to mine for any difference. - Original Message - From: "Brian C. Fertig" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Friday, July 01, 2005 3:26 PM Subject: RE: [Asterisk-Users] Got SIP response 481 "Invalid CSeq Number"backfrom X.X.X.X I had the same problem and I believe it was the payload size of the codec. What code are you using? ..o---o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Federico Alves Sent: Friday, July 01, 2005 4:20 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Got SIP response 481 "Invalid CSeq Number" backfrom X.X.X.X I get this error message when sending calls to a Cisco Gateway AS 5300, one call out of 10. Is there any configuration hack either on Asterisk or the Cisco that would this problem go away?? Federico Alves ___ 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 This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ 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 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 showing more than once on ps
Guys. Anybody know why sometimes on some servers Asterisk shows more than once while doing a ps? [EMAIL PROTECTED] akrall]# ps -ax|grep asterisk 20555 ?S 0:00 /bin/sh /usr/sbin/safe_asterisk 20557 ?S 0:00 asterisk -vvvg -c 20558 ?S 0:00 asterisk -vvvg -c 20560 ?S 0:00 asterisk -vvvg -c 20561 ?S 0:00 asterisk -vvvg -c 20562 ?S 0:00 asterisk -vvvg -c 20564 ?S 0:00 asterisk -vvvg -c 20565 ?S 0:00 asterisk -vvvg -c 20566 ?S 0:02 asterisk -vvvg -c 20567 ?S 0:00 asterisk -vvvg -c 21958 ?S 0:00 asterisk -vvvg -c 30470 pts/2S 0:00 grep asterisk Im running asterisk via the start/stop script on init.d On my other servers, it only shows once... ___ 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] Epia C3 Linux
Which fedora? Core 4 has some issue with the boot images on some hardware. Best to try text install rather than graphical. Core 1 and 2 had various issues with several motherboards (asus, via..) and needed a custom boot disk to install. Fedora, as all redhat distros are compiled for 386, which has benefits and disadvantages. Should do what you need it to do provided you can run the installer. Wiley Siler wrote: I just tried Fedora Core CD1 and it died on autopartitioning…. W *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Michael Stahl *Sent:* Friday, July 01, 2005 2:00 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* RE: [Asterisk-Users] Epia C3 Linux It installed directly from the FC3 dvd, no changes...no external drivers required *From:* Wiley Siler [mailto:[EMAIL PROTECTED] *Sent:* Friday, July 01, 2005 2:42 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* RE: [Asterisk-Users] Epia C3 Linux Did it require any special work or did you just download the ISO for FC3 and install? Thanks, Wiley *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Michael Stahl *Sent:* Friday, July 01, 2005 11:19 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* RE: [Asterisk-Users] Epia C3 Linux I have Fedora Core 3 running great on an Epia mobo *From:* Wiley Siler [mailto:[EMAIL PROTECTED] *Sent:* Friday, July 01, 2005 12:54 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [Asterisk-Users] Epia C3 Linux Anyone know a good distro for an Epia Mobo with the C3 chip? I have been trying to get Debian and Gentoo installed (new to me) and so far having little luck. Does anyone know a good install for this processor/mobo combo? Thanks Wiley ___ 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 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] Visual ring notification
Nope not in Puerto Rico, years ago when I was trying to get a "collins" surname domain it was there and two other totally unrelated countries (I think Bulgaria and France or something like that) never been their signed up over the web, send them a check once a year. > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Kristian Kielhofner > Sent: Friday, 1 July 2005 6:35 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Visual ring notification > > Dean Collins wrote: > > Lol, hardly just a USA thing. > > While some other countries certainly have their equivalents, OSHA > rules > certainly do not apply to Anguilla, which is where Chris resides... > > And, if you are in Puerto Rico, well, I'll let you tell me how OSHA > applies to a US territory (or self-governing commonwealth/whatever)... > > -- > Kristian Kielhofner > ___ > 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 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] H323 , OH323 and Sound Card
Hi all asterisk h323 user Please help me to know is there any sound quality effect without sound card . I am using onboard 32 bit sound card which comes with Asus NCCH-DL mother board. and getting delay from asterisk to any h323 device. but incomming is fine by OH323. 6.5 ver. is there any bit problem or onboard sound card problem or somewhere elase. please let me know. here i got another result , when stop that onboard card and install a old sound card 4 bit , creative and sound problem fixed. now really i am surprise. is there any issue on sound card then please suggest me what model and bit card is perfect for OH323 and my asterisk . Thanks. Bashir ___ 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] Provider Survey
I think he was looking for customer experiences. Michael D Schelin wrote: Call Mike at ShellTel 626-276-9009 List Receiver wrote: Having used Broadvoice for a while with marginal service, I want to move on to another provider. So my question to the List is who is good? I know now one service is perfect but somebody out there has to be decent. Who have you guys had the best luck with? -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 ___ 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] Visual ring notification
Dean Collins wrote: Lol, hardly just a USA thing. While some other countries certainly have their equivalents, OSHA rules certainly do not apply to Anguilla, which is where Chris resides... And, if you are in Puerto Rico, well, I'll let you tell me how OSHA applies to a US territory (or self-governing commonwealth/whatever)... -- Kristian Kielhofner ___ 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] Provider Survey
Call Mike at ShellTel 626-276-9009 List Receiver wrote: Having used Broadvoice for a while with marginal service, I want to move on to another provider. So my question to the List is who is good? I know now one service is perfect but somebody out there has to be decent. Who have you guys had the best luck with? ___ 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 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] Provider Survey
Having used Broadvoice for a while with marginal service, I want to move on to another provider. So my question to the List is who is good? I know now one service is perfect but somebody out there has to be decent. Who have you guys had the best luck with? smime.p7s Description: S/MIME cryptographic signature ___ 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] passing through MWI info from SBC
Is there alternative access to voicemail? Like web access? If there was then you can write a program to check the site every 5 or 10 minutes maybe every 30, and parse out the message listing and then send MWI to the phones if there is a VM waiting. Or You could get one of those cheep 20 dollar devices and hook it up to the serial port on the asterisk box and then you can use sty to monitor the port for changes. Then send the message that way. I'd have to research it a little more but it very possible. Just a thought :)On 7/1/05, Chris Gamble <[EMAIL PROTECTED]> wrote: Does this just sound worse than it is?>> "With SBC you are out of luck, since Asterisk doesn't detect dialtone (>> it dials blind, sometimes too quickly for the CO to catch the first>> digit, resulting in wrong numbers )) or stutter dialtone either, and >> reportedly has had any indication of the DC status of a POTS line>> removed due to problems."-Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]]On Behalf Of John NovackSent: Friday, July 01, 2005 10:42 AM To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] passing through MWI info from SBCMike Myers wrote:>Hi.. I am about to replace my aging Nortel Venture >system with an Asterisk system and 6 Polycom IP 501>phones, and a couple sipura 841's for less used areas.>>We have 3 phone lines here. One is SBC, one Vonage,>and one Voipjet... One hangup is that I can't figure >out how to pass through a voicemail waiting indication>from SBC. This is important because my wife and her>family all exchange voicemails with each other on the>SBC voicemail system. They can leave messages for >each other without having the phones ring, etc... We>have a 2 yr old at home, and her sister has some small>kids too, so that's how they manage to send voicemails>when they are unsure if the kids are sleeping, etc... >Anyway, preserving this capability of using the SBC VM>and being notified when a message is waiting is>critical for good WAF.>>The vonage line and voipjet line can be intergrated>into the Asterisk VM. My Nortel venture phones light >the MWI if any line has VM on it, and the display>tells you which lines have VM waiting. I would love>to be able to duplicate this function on the Polycom's>and hopefully the Sipura's as well. >>I've looked for answers on this, but haven't found>one, hence the post. My apologies if I have missed>something.>>Thanks much,>Mike>>You haven't missed much. With SBC you are out of luck, since Asterisk doesn't detect dialtone (it dials blind, sometimes too quickly for the CO to catch the firstdigit, resulting in wrong numbers )) or stutter dialtone either, and reportedly has had any indication of the DC status of a POTS lineremoved due to problems.Only choice would to port the number to a VOIP provider and provide theVM in Asterisk.Similar problem with Vonage VM. John Novack___Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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] Epia C3 Linux
I just tried Fedora Core CD1 and it died on autopartitioning…. W From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Stahl Sent: Friday, July 01, 2005 2:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Epia C3 Linux It installed directly from the FC3 dvd, no changes...no external drivers required From: Wiley Siler [mailto:[EMAIL PROTECTED] Sent: Friday, July 01, 2005 2:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Epia C3 Linux Did it require any special work or did you just download the ISO for FC3 and install? Thanks, Wiley From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Stahl Sent: Friday, July 01, 2005 11:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Epia C3 Linux I have Fedora Core 3 running great on an Epia mobo From: Wiley Siler [mailto:[EMAIL PROTECTED] Sent: Friday, July 01, 2005 12:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Epia C3 Linux Anyone know a good distro for an Epia Mobo with the C3 chip? I have been trying to get Debian and Gentoo installed (new to me) and so far having little luck. Does anyone know a good install for this processor/mobo combo? Thanks Wiley ___ 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] spandsp fax out fails
On Friday 01 Jul 2005 00:17, David Romero wrote: > Are you sharing the IRQ of the zap card whit other device? > some times when the zap card share IRQ whit other device spansdp fail. Turned out that while a fax may be a tiff file, it does not mean that a tiff file is a fax. The size of the generated tiff file was wrong. Interestingly though, when I try to fax out the PRI to one of our own DDI's, that to say it come back in on the PRI, the fax software just sits there looking stupid! > On 6/30/05, Bob Goddard <[EMAIL PROTECTED]> wrote: > > I've a stock RH9 system with spandsp 0.18. Faxing out over a PRI to a > > USRobotics modem on a stock Suse9.3 system with hylafax fails with the > > following errors in the hylafax logs: > > > > Jun 30 19:28:53.23: [ 608]: RECV/CQ: Bad 1D pixel count, row 0, got 595, > > expected 1728 > > Jun 30 19:28:53.23: [ 608]: RECV/CQ: Bad 1D pixel count, row 1, got 595, > > expected 1728 > > Jun 30 19:28:53.23: [ 608]: RECV/CQ: Bad 1D pixel count, row 2, got 595, > > expected 1728 > > etc... > > > > I'm using the following call file as a test: > > > > Channel: Zap/g1/XX > > MaxRetries: 0 > > WaitTime: 20 > > Application: txfax > > Data: /root/t.tif|caller > > > > > > The * console does not give any useful info even when the verbose > > setting is on max. > > > > The tiff file in question does not seem to be a problem is it is a > > 2 page file which is viewable just fine with xv. > > > > Does anyone have any clue as to what is wrong? It fails even if I > > set it up such that it dials out then back in on itself. ___ 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] How to Configure a H323 Phone (newbie here)
I would also be interested. I've tried several times unsuccessfully to set up H323 with Asterisk. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adeel -31Sent: 01 July 2005 23:32To: asterisk-users@lists.digium.comSubject: [Asterisk-Users] How to Configure a H323 Phone (newbie here) i read that asterisk supports iax,sip and h323 protocols i've used sip & iax softphones ... now i've a hardphone... an IP phone (Netphone) that supports h323 . i've compiled pwlib ,oh323 and asterisk -oh323 successfully ... but i m unable to place calls to/by my phone... i m confused whether to use h323.conf or oh323.conf and how ? i think it's different from iax.conf & sip.conf can anyone send me his working oh323.conf . or give some link that can be helpful in configuration Adeel Yahoo! SportsRekindle the Rivalries. Sign up for Fantasy Football ___ 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] Linux Firewall Question
Excellent..stuff, I have installed both shorewall and webmin and begin playing with conf. Thanks Obaid. - Original Message - From: "Anand S. Katti" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Friday, July 01, 2005 12:44 AM Subject: RE: [Asterisk-Users] Linux Firewall Question > > > > > Hi, > > > I am running Asterisks on Public IP with Fedora Core 3. > > > > > > What is the recommendation for making Linux secure on the > > > Public IP since I am new to Linux. Which Firewall should I > > > use? I am not intending to use Linux as router. > > > > > > Can any one provide some configuration documentation. > > > > I use shorewall, and I have found it powerful, and fairly easy to use. > > > > http://www.shorewall.net/ > > Shorewall is an excellant iptables based firewall. You can create zones > and policy's to govern them. If you are also looking for easy GUI to > configure it, then best way to do it is using WEBMIN http://www.webmin.com > > Its a free web based interface for system management. It includes Module > for shorewall confuguration. > > -ask > > ___ > 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 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] Problem with DTFM and complex international setup
I don't find this option in the Makefile. I find RADIO_RELAX which is something to do with radios and DTMF. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mohit Muthanna Sent: 01 July 2005 23:24 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Problem with DTFM and complex international setup Try compiling Asterisk with RELAX_DTMF (See Makefile). Mohit. On 7/1/05, Rob Scott <[EMAIL PROTECTED]> wrote: > We have some guys working in the US who can't always dial back to our > company in Europe easily (lots of clients require authorization to > make international calls), so I set up the following: > >- ipkall.com number links to a FWD number >- office Asterisk box registers with FWD > > Then I programmed Asterisk to accept office extension number using > DTFM tones. > This works OK. > > Then I programmed Asterisk so that it is possible, using a PIN code, > to dial out from Asterisk onto the local PSTN. > > This also works occasionally. > Looking at the message from the Asterisk box it is clear that > sometimes numbers are missed or repeated in the dial string. This I > suspect is because Asterisk is listening to the DTMF tones but the > signal is dropped; sometimes the drop means that a whole digit is > dropped and sometimes is means that a digit is repeated. > > Does anyone know how I can fix this to make it more reliable > (out-of-band DTMF?) or a better way to achieve a reliable setup? > ___ > 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 > -- Mohit Muthanna [mohit (at) muthanna (uhuh) com] "There are 10 types of people. Those who understand binary, and those who don't." ___ 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 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] Errors Question
Today we have been having some problems with the dchannel of out T1's. I was wondering if there is a way for asterisk to send out an email or page whenan error occurs. Not I know errors happen quite offen for many reasons, but I would like an email sent when there is a TI problem, or specefic other problems. I am using the safe_asterisk to run asterisk and it sends an email when asterisk crashes, which is not very often. Thanks, Kyle ___ 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] Visual ring notification
You can get this... http://www.radioshack.com/product.asp?catalog_name=CTLG&category_name=&product_id=43-178 and if you want to get crazy you can disassemble it and change the strobe to be one of these http://homesecuritystore.com/ezStore123/DTProductZoom.asp?productID=571 they both use 12 volts and you can mount the strob anywhere.On 7/1/05, Rich Adamson <[EMAIL PROTECTED] > wrote:> >> > - cheap ata with telco-style industrial horn >> Just CYA regarding OSHA regulations on permissible noise levels.Headlines Read: three fingers missing after the Baker's handsslipped into the tomato slicer when the phone rang... ;)___ Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list 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] Visual ring notification
You can check out a bunch of products for deaf people http://www.hearmore.com/store/prodView.asp?idproduct=2639&idstore=1&idCategory=105 THat is one in particular that looks like what you might want. Sorry for posting twice :)On 7/1/05, andrew matthews <[EMAIL PROTECTED]> wrote: You can get this... http://www.radioshack.com/product.asp?catalog_name=CTLG&category_name=&product_id=43-178 and if you want to get crazy you can disassemble it and change the strobe to be one of these http://homesecuritystore.com/ezStore123/DTProductZoom.asp?productID=571 they both use 12 volts and you can mount the strob anywhere.On 7/1/05, Rich Adamson < [EMAIL PROTECTED] > wrote:> >> > - cheap ata with telco-style industrial horn >> Just CYA regarding OSHA regulations on permissible noise levels.Headlines Read: three fingers missing after the Baker's handsslipped into the tomato slicer when the phone rang... ;)___ Asterisk-Users mailing listAsterisk-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 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 Configure a H323 Phone (newbie here)
i read that asterisk supports iax,sip and h323 protocols i've used sip & iax softphones ... now i've a hardphone... an IP phone (Netphone) that supports h323 . i've compiled pwlib ,oh323 and asterisk -oh323 successfully ... but i m unable to place calls to/by my phone... i m confused whether to use h323.conf or oh323.conf and how ? i think it's different from iax.conf & sip.conf can anyone send me his working oh323.conf . or give some link that can be helpful in configuration Adeel Yahoo! Sports Rekindle the Rivalries. Sign up for Fantasy Football ___ 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] Visual ring notification
Lol, hardly just a USA thing. > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Kristian Kielhofner > Sent: Friday, 1 July 2005 5:22 PM > To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial > Discussion > Subject: Re: [Asterisk-Users] Visual ring notification > > Chris Mason (Lists) wrote: > > > >> Just CYA regarding OSHA regulations on permissible noise levels. > >> > >> > >> > > OSHA who? > > Don't worry about it Chris, it's another US thing: > > "Occupational Safety & Health Administration" > > http://www.osha.gov/ > > > -- > Kristian Kielhofner > ___ > 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 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] Got SIP response 481 "Invalid CSeq Number" backfrom
as far as I know there isn't. I use 80 bytes for G711U that may or may not fix your issue. You can also do a ethereal trace to find out what the actual error is. ..o---o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Federico Alves Sent: Friday, July 01, 2005 5:15 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Got SIP response 481 "Invalid CSeq Number" backfrom I am using G711. In the Cisco, how many bytes should I use for the payload? Is there any way to configure the payload in Asterisk? Thanks in advance. Federico Alves ___ 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 This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ 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] Problem with DTFM and complex international setup
Try compiling Asterisk with RELAX_DTMF (See Makefile). Mohit. On 7/1/05, Rob Scott <[EMAIL PROTECTED]> wrote: > We have some guys working in the US who can't always dial back to our > company in Europe easily (lots of clients require authorization to make > international calls), so I set up the following: > >- ipkall.com number links to a FWD number >- office Asterisk box registers with FWD > > Then I programmed Asterisk to accept office extension number using DTFM > tones. > This works OK. > > Then I programmed Asterisk so that it is possible, using a PIN code, to > dial out from Asterisk onto the local PSTN. > > This also works occasionally. > Looking at the message from the Asterisk box it is clear that sometimes > numbers are missed or repeated in the dial string. This I suspect is > because Asterisk is listening to the DTMF tones but the signal is > dropped; sometimes the drop means that a whole digit is dropped and > sometimes is means that a digit is repeated. > > Does anyone know how I can fix this to make it more reliable > (out-of-band DTMF?) or a better way to achieve a reliable setup? > ___ > 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 > -- Mohit Muthanna [mohit (at) muthanna (uhuh) com] "There are 10 types of people. Those who understand binary, and those who don't." ___ 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] Visual ring notification
Chris Mason (Lists) wrote: Just CYA regarding OSHA regulations on permissible noise levels. OSHA who? Don't worry about it Chris, it's another US thing: "Occupational Safety & Health Administration" http://www.osha.gov/ -- Kristian Kielhofner ___ 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] Got SIP response 481 "Invalid CSeq Number" back from
I am using G711. In the Cisco, how many bytes should I use for the payload? Is there any way to configure the payload in Asterisk? Thanks in advance. Federico Alves ___ 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] Visual ring notification
Just CYA regarding OSHA regulations on permissible noise levels. OSHA who? -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ 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] E3 card
Kevin P. Fleming wrote: Tamas J wrote: Does anybody know what will be the around price for the announced E3 card from Digium? When is it planned to be ready? Pricing and release date have not been announced at this time. The product has also been removed from Digium's web site, as far as I can tell. -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 ___ 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] Epia C3 Linux
It installed directly from the FC3 dvd, no changes...no external drivers required From: Wiley Siler [mailto:[EMAIL PROTECTED] Sent: Friday, July 01, 2005 2:42 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] Epia C3 Linux Did it require any special work or did you just download the ISO for FC3 and install? Thanks, Wiley From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael StahlSent: Friday, July 01, 2005 11:19 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] Epia C3 Linux I have Fedora Core 3 running great on an Epia mobo From: Wiley Siler [mailto:[EMAIL PROTECTED] Sent: Friday, July 01, 2005 12:54 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Epia C3 Linux Anyone know a good distro for an Epia Mobo with the C3 chip? I have been trying to get Debian and Gentoo installed (new to me) and so far having little luck. Does anyone know a good install for this processor/mobo combo? Thanks Wiley ___ 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] Voicemail => SMS
A little off topic, but I'm on orange, what's the domain and what is the format
e.g. 07973 or +447973...
From: [EMAIL PROTECTED] on behalf of Wilson Pickett
Sent: Fri 01/07/2005 6:56
To: Mark Charlton; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Voicemail => SMS
> I have been trying for a while to find a way to get an SMS send when I
> receive a voicemail into my asterisk system. I don't want to send an
> SMS if the caller doesn't leave a message. I have voicemail.conf set
> up to email and delete.
I use a backward solution to this problem, but it works. Orange, my
cell provider offers free SMS alerts for email sent to
[EMAIL PROTECTED] I send my vmail messages to my regular email
server which keeps them for online email retrieval. A procmail recipe
on the server then makes up an email without the vmail attachment to
my orange address with the callerid in the subject. Orange sends an
SMS that tells me I have a vmail message from ${CALLERID}. Although it
seems like a silly solution it does _exactly_ what you asked about.
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Re: [Asterisk-Users] Call Transfer Problem
Adam Robins wrote: The double-star now works great. If I press it while on a call, I go into transfer mode. The problem is that the # still works as well! Shouldn't the atzfer specification turn off the #? Blind transfers are on '#' by default, so you may need to move them to another sequence as well. ___ 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] Got SIP response 481 "Invalid CSeq Number" backfrom X.X.X.X
I had the same problem and I believe it was the payload size of the codec. What code are you using? ..o---o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Federico Alves Sent: Friday, July 01, 2005 4:20 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Got SIP response 481 "Invalid CSeq Number" backfrom X.X.X.X I get this error message when sending calls to a Cisco Gateway AS 5300, one call out of 10. Is there any configuration hack either on Asterisk or the Cisco that would this problem go away?? Federico Alves ___ 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 This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ 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] Got SIP response 481 "Invalid CSeq Number" back from X.X.X.X
I get this error message when sending calls to a Cisco Gateway AS 5300, one call out of 10. Is there any configuration hack either on Asterisk or the Cisco that would this problem go away?? Federico Alves ___ 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] how to send voicemail notifcation every 15 minutes until message is checked
Thanks - a cronjob for the user was going to be my last resort. Was not sure if there was a setting like "repeatnotify=15" to repeat the notice every 15 minutes. Thanks for your feedback though! On 7/1/05, Michiel van Baak <[EMAIL PROTECTED]> wrote: > On 13:33, Fri 01 Jul 05, Pedro wrote: > > I have searched quite a few places and have not seen this discussed. > > Basically I was wondering how would you go about having an option for > > a user to be notified every 15 minutes until their new voicemail > > message is checked. Since the notification e-mails we send get sent > > to cell phones or actual pagers (via e-mail), there are times when a > > person is out of range and misses a page or just simply is too busy to > > check voicemail and then forgets. They want to be reminded 15 minutes > > later until that new message is checked. > > > > Current version of asterisk that we are running is CVS-v1-0-11/12/04 > > (which has been running rock-solid I might add). Any thoughts are > > appreciated. > > Hi, > > You can check the new mail count with the manager interface > or by looking at the spool dir. > If you put this in cron every 15 minutes, you're done. > > Michiel > ___ > 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 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] Visual ring notification
> > > > - cheap ata with telco-style industrial horn > > Just CYA regarding OSHA regulations on permissible noise levels. Headlines Read: three fingers missing after the Baker's hands slipped into the tomato slicer when the phone rang... ;) ___ 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 spandsp
I got this working on Fedora Core 3 i386, but it does not work on CentOS
x86_64. Anything I should know about x86_64?
Thanks.
Ted
Theodore Cekan wrote:
Hello,
I am trying to setup spandsp for the first time. I have spandsp
0.0.2pre18 and * v1.0.8 patched. For testing I have this extension:
exten => 1234,1,rxfax(/tmp/testfax.tif)
* answers the call, but makes no fax tones. I get this logged:
-- Executing RxFAX("SIP/x.x.x.x-00668660", "/tmp/testfax.tif") in
new stack
then nothing, silence, no communication.
Any help is appreciated.
Ted
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RE: [Asterisk-Users] Visual ring notification
> -Original Message- > From: Rich Adamson [mailto:[EMAIL PROTECTED] > Sent: Friday, July 01, 2005 2:39 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Visual ring notification > > > - cheap ata with telco-style industrial horn Just CYA regarding OSHA regulations on permissible noise levels. ___ 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] Voicemail => SMS
On 01/07/05, Mark Charlton <[EMAIL PROTECTED]> wrote: > On 7/1/05, Peter Bowyer <[EMAIL PROTECTED]> wrote: > > On 01/07/05, Mark Charlton <[EMAIL PROTECTED]> wrote: > > > > > > > I have been fighting with the Bayham Systems FastSMS AGI script, and I > > > re-wrote it as a stand alone Perl script. I am now calling it with > > > the EXTERNNOTIFY option in the voicemail.conf file. It gets passed > > > the context, extension and number of messages which I build into a > > > text, and since they all go to the same location its no problem. I'm > > > planning on using the extension info to open the mailbox, and read the > > > text file for the latest message to pull out the caller for the text. > > > I might also have an extension map in a text file so I can look up who > > > to notify about a VM. > > > > I also hacked Bayham Systems' script. I need to control the MWI on GSM > > phones, which is turned on and off by custom SMS messages (which > > Bayham helpfully provide macros for). And I needed to avoid sending > > repeat notifications when a second or subsequent new message was left. > > > > I ended up keeping a flag in a db file which stores per mobile number > > what state the MWI is in for that phone. If the flag says the MWI is > > already set when a new message comes in, then the script is a no-op, > > otherwise it sends the 'MWI on' SMS and flips the flag. Ditto in > > reverse. > > > > Works fine and dandy here. > > > > > > Peter > > How do you monior when the VM is listened to to turn off the MWI on the > mobile? > Mark I have a 'h' extension in the context where voicemailmain gets called, which calls the MWI script. -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] ___ 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] passing through MWI info from SBC
Does this just sound worse than it is? >> "With SBC you are out of luck, since Asterisk doesn't detect dialtone ( >> it dials blind, sometimes too quickly for the CO to catch the first >> digit, resulting in wrong numbers )) or stutter dialtone either, and >> reportedly has had any indication of the DC status of a POTS line >> removed due to problems." -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Novack Sent: Friday, July 01, 2005 10:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] passing through MWI info from SBC Mike Myers wrote: >Hi.. I am about to replace my aging Nortel Venture >system with an Asterisk system and 6 Polycom IP 501 >phones, and a couple sipura 841's for less used areas. > >We have 3 phone lines here. One is SBC, one Vonage, >and one Voipjet... One hangup is that I can't figure >out how to pass through a voicemail waiting indication >from SBC. This is important because my wife and her >family all exchange voicemails with each other on the >SBC voicemail system. They can leave messages for >each other without having the phones ring, etc... We >have a 2 yr old at home, and her sister has some small >kids too, so that's how they manage to send voicemails >when they are unsure if the kids are sleeping, etc... >Anyway, preserving this capability of using the SBC VM >and being notified when a message is waiting is >critical for good WAF. > >The vonage line and voipjet line can be intergrated >into the Asterisk VM. My Nortel venture phones light >the MWI if any line has VM on it, and the display >tells you which lines have VM waiting. I would love >to be able to duplicate this function on the Polycom's >and hopefully the Sipura's as well. > >I've looked for answers on this, but haven't found >one, hence the post. My apologies if I have missed >something. > >Thanks much, >Mike > > You haven't missed much. With SBC you are out of luck, since Asterisk doesn't detect dialtone ( it dials blind, sometimes too quickly for the CO to catch the first digit, resulting in wrong numbers )) or stutter dialtone either, and reportedly has had any indication of the DC status of a POTS line removed due to problems. Only choice would to port the number to a VOIP provider and provide the VM in Asterisk. Similar problem with Vonage VM. John Novack ___ 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 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] SIPGetHeader application in asterisk-1.0.9
Kamran Ahmad wrote: hello i want to use SIPGetHeader application in asterisk-1.0.9. Jul 2 00:04:33 WARNING[19575]: pbx.c:1293 pbx_extension_helper: No application 'SIPGetHeader' for extension (default, 2000, 1) That application is not, and never will be in 1.0.x as it's a new feature and new features to not go into 1.0.x. -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 ___ 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] Visual ring notification
> I have put a pbx into a resort with Polyycom phones, everythign works > great, except the kitchen staff cannot hear the phone ring. I know many > legacy systems employ a big red flashing light, any ideas on doing > something similar? Several choices, some better then others ;) - radio shack use to sell kits that sensed audio noices and tripped a relay. Install inside the phone and sense the ringer audio. Suspect similar kits sold by others. - cheap ata with inexpensive rj11 radio shack ringer adapter, and sip/11&sip/22 - cheap ata with telco-style industrial horn - modify sip phone electronics to add a relay closure (and lights) based on ringer signal ___ 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] SIPGetHeader application in asterisk-1.0.9
hello i want to use SIPGetHeader application in asterisk-1.0.9. Jul 2 00:04:33 WARNING[19575]: pbx.c:1293 pbx_extension_helper: No application 'SIPGetHeader' for extension (default, 2000, 1) Any one using this __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] Voicemail => SMS
On 7/1/05, Peter Bowyer <[EMAIL PROTECTED]> wrote: > On 01/07/05, Mark Charlton <[EMAIL PROTECTED]> wrote: > > > > > I have been fighting with the Bayham Systems FastSMS AGI script, and I > > re-wrote it as a stand alone Perl script. I am now calling it with > > the EXTERNNOTIFY option in the voicemail.conf file. It gets passed > > the context, extension and number of messages which I build into a > > text, and since they all go to the same location its no problem. I'm > > planning on using the extension info to open the mailbox, and read the > > text file for the latest message to pull out the caller for the text. > > I might also have an extension map in a text file so I can look up who > > to notify about a VM. > > I also hacked Bayham Systems' script. I need to control the MWI on GSM > phones, which is turned on and off by custom SMS messages (which > Bayham helpfully provide macros for). And I needed to avoid sending > repeat notifications when a second or subsequent new message was left. > > I ended up keeping a flag in a db file which stores per mobile number > what state the MWI is in for that phone. If the flag says the MWI is > already set when a new message comes in, then the script is a no-op, > otherwise it sends the 'MWI on' SMS and flips the flag. Ditto in > reverse. > > Works fine and dandy here. > > > Peter How do you monior when the VM is listened to to turn off the MWI on the mobile? Mark ___ 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: passing through MWI info from SBC
Mike Myers wrote: Thanks, but my wife never pays attention to the stutter tone, just the MWI. I know I can light the Polycom's MWI under Asterisk's control, but if Asterisk can't hear the FSK tone then it's not useful... Does anyone know if the FSK CAN be detected by Asterisk? Asterisk listens for the CallerID before going off hook, so it shouldn't be too far off the mark. Anyone know what is sent to indicate MW? JN Also, she will definitely want to use the new phones, since she makes heavy use of intercom functions in our current venture phones. I was raised on a farm and learned it was Ok to raise voice gain to the level that the signal would be receieved. But my wife was raised in the suburbs with a tight vocal AGC was in effect, and when the AGC threshold was exceeded in our house, the channel would go silent for long periods of time after a momementary burst of noise that overloaded my frontend. :-) The intercom solved the problem for our marriage. Thanks, Mike ___ 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] Saydigits
Hi... has anyone written or seen a variation of Saydigits that behaves like Background (listening and responding to DTMF)? If there's such a beast, I'd sure like to know... if not, how hard would it be to implement? Jesus ___ 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't build cdr_addon_mysql.
Hmm.. I'm having this problem today: /usr/local/mysql/lib/mysql/libmysqlclient.a /usr/local/mysql/lib/mysql/libmysqlclient.la /usr/local/mysql/lib/mysql/libmysqlclient.so /usr/local/mysql/lib/mysql/libmysqlclient.so.14 /usr/local/mysql/lib/mysql/libmysqlclient.so.14.0.0 cc -shared -Xlinker -x -o cdr_addon_mysql.so cdr_addon_mysql.o -lmysqlclient -lz-L/usr/local/mysql/lib /usr/lib/gcc-lib/i486-slackware-linux/3.3.4/../../../../i486-slackware-linux/bin/ld: cannot find -lmysqlclient collect2: ld returned 1 exit status make: *** [cdr_addon_mysql.so] Error 1 I've tried many many things to get it going but have failed (incl. reinstalling zlib)... any ideas? Regards, Sahil Gupta VoiceValley On Fri, 1 Jul 2005, Brian West wrote: You could have just done "ln -s asterisk-1.0.9 asterisk" and it would have fixed that. It should by default do -I../asterisk /b --- Anakin: “You’re either with me, or you’re my enemy.” Obi-Wan: “Only a Sith could be an absolutist.” On Jun 30, 2005, at 1:13 AM, Chris Mason (Lists) wrote: Marcel van Kaam, Fonetica wrote: I had the same problem with installing addons. I checked out in the file cdr_addons_mysql.c what the location of the asterisk.h must be and changed the cdr_addons_mysql.c to the location of the asterisk.h file. After this it worked. Also to be sure do: locate asterisk.h to check or you have the file on your system. Marcel Yes, that worked. For the record, it had to be #include "../asterisk-1.0.9/asterisk.h" -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ 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 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 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: passing through MWI info from SBC
Robert Webb wrote: >Or, could you use something like the zap_barge option >tied >into a routine that monitors for the FSK and then when >it >is received, it then runs the routine that is already in >place to set the MWI for the FXS ports. >Crude, I know, but the only way I can think of to pass >it. >Robert Robert, this is interesting. From what I can tell zap_barge allows ASterisk to connect to an ongoing call, but I don't think this will work since FSK tones are sent when no call is present. Also, how would I write a routine that detects the FSK tones? Is there a generic DSP function that could be used for this? Thanks, Mike Yahoo! Sports Rekindle the Rivalries. Sign up for Fantasy Football http://football.fantasysports.yahoo.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: passing through MWI info from SBC
Chris A. Icide wrote: >Here is what I would do. Install a TDM04 card with a >couple fxos. >Connect the analog phones that your wife will be using >to the tdm card. >In zapata.conf, set those phones to immediate=yes, and >when you get an >event on the fxo port, connect it to the fxs port with >the stutter >tone. This way, when she picks up the phone, it will >immediately >connect her to the sbc provided dial tone, and she can >hear the stutter >or lack thereof. When a call comes inbout however, >you can still route >it as you want. >Not a perfect solution, since the phones she will be >using are forced to >use SBC, but the best solution I can think of. >-Chris Thanks, but my wife never pays attention to the stutter tone, just the MWI. I know I can light the Polycom's MWI under Asterisk's control, but if Asterisk can't hear the FSK tone then it's not useful... Also, she will definitely want to use the new phones, since she makes heavy use of intercom functions in our current venture phones. I was raised on a farm and learned it was Ok to raise voice gain to the level that the signal would be receieved. But my wife was raised in the suburbs with a tight vocal AGC was in effect, and when the AGC threshold was exceeded in our house, the channel would go silent for long periods of time after a momementary burst of noise that overloaded my frontend. :-) The intercom solved the problem for our marriage. Thanks, Mike Yahoo! Sports Rekindle the Rivalries. Sign up for Fantasy Football http://football.fantasysports.yahoo.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] new chan_sccp release
http://chan-sccp.berlios.de/ (you can find a mailing list here and a bug tracker) 20050701 ftp://ftp.berlios.de/pub/chan-sccp/chan_sccp.20050701.tar.gz I did rework the original chan_sccp. Asterisk cisco phones channel driver - fixed all the issues, added localized softkeys and display messages (the phone load XML localized strings from the tftp server). - complete rewrite of the call flow. - cleaned the console. Now you can use sccp debug to set the debug level from 0 to 10 (verbose) - added new directives (incominglimit, tos and rtptos, digittimeout, firstdigittimeout and more. I need to modify the config file) - added support for callwaiting - no need to edit the makefile. It does discover the asterisk version. You just need to extract and make install. - new locking system - more and more :-) I will work on it to add native transfer/blind transfer and call forward capabilities Please test it. Sergio Chersovani ___ 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] Visual ring notification
I need one I can build into a kitchen hood and will be seen at 20'. Think fire alarm. -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ 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] pattern matching based on callerid, not working
I'm not sure it's the source of your problem, but I'm sure it could wind
up being the source of others:
I think that should be:
exten => _9./3003,1,Set(CALLERID(number)=281443)
exten => _9./3004,1,Set(CALLERID(number)=281444) ; these should
exten => _9./3005,1,Set(CALLERID(number)=281445) ; all be priority
exten => _9./3006,1,Set(CALLERID(number)=281446) ; 1, not n
exten => _9.,n,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,wt)
On Fri, 01 Jul 2005 13:15:08 -0500, "Matthew Boehm"
<[EMAIL PROTECTED]> said:
> according to the wiki, I should be able to do this:
>
> exten => _9./3003,1,Set(CALLERID(number)=281443)
> exten => _9./3004,n,Set(CALLERID(number)=281444)
> exten => _9./3005,n,Set(CALLERID(number)=281445)
> exten => _9./3006,n,Set(CALLERID(number)=281446)
> exten => _9.,n,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,wt)
>
> and have the correct calleridnum's set for each extension based on their
> current calleridnum.
>
> Basically, priority 1 will execute only if callerid is currently 3003.
> pri2 will only execute if callerid is 3004, etc..
>
> however, attempts to do this all fail with auto-fallthru BUSY.
>
> Im using most recent CVS-HEAD.
>
> Any ideas?
>
> -Matthew
>
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I'm not sure it's the source of your problem, but I'm sure it could wind
up being the source of others:
I think that should be:
exten => _9./3003,1,Set(CALLERID(number)=281443)
exten => _9./3004,1,Set(CALLERID(number)=281444) ; these should
exten => _9./3005,1,Set(CALLERID(number)=281445) ; all be priority
exten => _9./3006,1,Set(CALLERID(number)=281446) ; 1, not n
exten => _9.,n,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,wt)
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RE: [Asterisk-Users] Re: passing through MWI info from SBC
I'm using $5 surplussed AT&T Message Waiting Lights, which I modified to show different colors. It's a "works-for-me" solution, of course. > -Original Message- > From: Mike Myers [mailto:[EMAIL PROTECTED] > Subject: [Asterisk-Users] Re: passing through MWI info from SBC > > find a way to communicate the message waiting signal > in Asterisk. It's a show stopper in my house. > > I am surprised that noone has run into this issue > before. I just *had* to respond to that, as I don't think it's quite so surprising. I'll probably get bashed for making sweeping generalizations, but I doubt that telco voice-mail (and related MWI) are very commonplace in a business environment. So, that probably knocks out half the asterisk installations out there. Of the other half, I'd propose that a majority chose asterisk partly because of its excellent voice-mail features. I personally find telco (and cellular) voice-mail to be a PITA, and configured my * server to never show a busy signal. If all "lines" are busy, it'll go to my *-based voicemail. Even cellular calls are forwarded to *, rather than T-Mobile's voicemail system. All that to say that in my estimation, you may be one of only a handful of * users who actually are interested in detecting MWI on CO lines. With DIDs costing virtually nothing (compared to land lines) these days, consider getting an extra one for "silent" calls. You could even set up voicemail boxes for your relatives on your * system and allow them to check remotely. Or give ATAs to those of your relatives who are broadband-savvy... ___ 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] Epia C3 Linux
Did it require any special work or did you just download the ISO for FC3 and install? Thanks, Wiley From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Stahl Sent: Friday, July 01, 2005 11:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Epia C3 Linux I have Fedora Core 3 running great on an Epia mobo From: Wiley Siler [mailto:[EMAIL PROTECTED] Sent: Friday, July 01, 2005 12:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Epia C3 Linux Anyone know a good distro for an Epia Mobo with the C3 chip? I have been trying to get Debian and Gentoo installed (new to me) and so far having little luck. Does anyone know a good install for this processor/mobo combo? Thanks Wiley ___ 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: passing through MWI info from SBC
On Fri, 01 Jul 2005 11:10:27 -0700 "Chris A. Icide" <[EMAIL PROTECTED]> wrote: John Novack wrote: Mike Myers wrote: Wow, this is a serious problem for me. I don't need to actually check the voicemail itself from Asterisk, just to be able to tell that there is voicemail waiting. Are you saying there is no way in Asterisk to do this? Is that true for using Digium hardware as well as FXO ports on a SIP ATA? Vonage VM doesn't matter to me, since I'll turn it off and use Asterisk for that functionality, but determining SBC's VM status is very important. My whole wife's family (multiple households) uses it. In the past, if one family tried to switch to a non SBC provider, they always returned in less than a week because of lack of VM interoperation. So my wife will put the kibosh on the whole Asterisk project unless I can light the MWI light when SBC VM is waiting. Since the cheapest analog phones can do this, I don't think she's going to understand that these $200 Polycom phones can't... :-( Is there no way around this? Thanks, Mike Here is what I would do. Install a TDM04 card with a couple fxos. Connect the analog phones that your wife will be using to the tdm card. In zapata.conf, set those phones to immediate=yes, and when you get an event on the fxo port, connect it to the fxs port with the stutter tone. This way, when she picks up the phone, it will immediately connect her to the sbc provided dial tone, and she can hear the stutter or lack thereof. When a call comes inbout however, you can still route it as you want. Not a perfect solution, since the phones she will be using are forced to use SBC, but the best solution I can think of. -Chris Or, could you use something like the zap_barge option tied into a routine that monitors for the FSK and then when it is received, it then runs the routine that is already in place to set the MWI for the FXS ports. Crude, I know, but the only way I can think of to pass it. Robert ___ 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: [Asterisk-ss7] Asterisk - ss7
I thought everyone should know this. Jorge, After reading your page in the http://voip-info.org/tiki-index.php?page=Asterisk+SS7 please advise Your U.S. customers that SS7 is not done the same way as in the rest of the world and the requirements are different. The U.S carrier's require 2 redundant links. I know this first hand because we run an SS7 network. CARDOSO Jorge Miguel wrote: http://voip-info.org/tiki-index.php?page=Asterisk+SS7 ___ Asterisk-SS7 mailing list Asterisk-SS7@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-ss7 ___ 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] Epia C3 Linux
I have Fedora Core 3 running great on an Epia mobo From: Wiley Siler [mailto:[EMAIL PROTECTED] Sent: Friday, July 01, 2005 12:54 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Epia C3 Linux Anyone know a good distro for an Epia Mobo with the C3 chip? I have been trying to get Debian and Gentoo installed (new to me) and so far having little luck. Does anyone know a good install for this processor/mobo combo? Thanks Wiley ___ 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] pattern matching based on callerid, not working
according to the wiki, I should be able to do this:
exten => _9./3003,1,Set(CALLERID(number)=281443)
exten => _9./3004,n,Set(CALLERID(number)=281444)
exten => _9./3005,n,Set(CALLERID(number)=281445)
exten => _9./3006,n,Set(CALLERID(number)=281446)
exten => _9.,n,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,wt)
and have the correct calleridnum's set for each extension based on their
current calleridnum.
Basically, priority 1 will execute only if callerid is currently 3003.
pri2 will only execute if callerid is 3004, etc..
however, attempts to do this all fail with auto-fallthru BUSY.
Im using most recent CVS-HEAD.
Any ideas?
-Matthew
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Re: [Asterisk-Users] Re: passing through MWI info from SBC
John Novack wrote: Mike Myers wrote: Wow, this is a serious problem for me. I don't need to actually check the voicemail itself from Asterisk, just to be able to tell that there is voicemail waiting. Are you saying there is no way in Asterisk to do this? Is that true for using Digium hardware as well as FXO ports on a SIP ATA? Vonage VM doesn't matter to me, since I'll turn it off and use Asterisk for that functionality, but determining SBC's VM status is very important. My whole wife's family (multiple households) uses it. In the past, if one family tried to switch to a non SBC provider, they always returned in less than a week because of lack of VM interoperation. So my wife will put the kibosh on the whole Asterisk project unless I can light the MWI light when SBC VM is waiting. Since the cheapest analog phones can do this, I don't think she's going to understand that these $200 Polycom phones can't... :-( Is there no way around this? Thanks, Mike Here is what I would do. Install a TDM04 card with a couple fxos. Connect the analog phones that your wife will be using to the tdm card. In zapata.conf, set those phones to immediate=yes, and when you get an event on the fxo port, connect it to the fxs port with the stutter tone. This way, when she picks up the phone, it will immediately connect her to the sbc provided dial tone, and she can hear the stutter or lack thereof. When a call comes inbout however, you can still route it as you want. Not a perfect solution, since the phones she will be using are forced to use SBC, but the best solution I can think of. -Chris ___ 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: passing through MWI info from SBC
John, et al, --- John Novack <[EMAIL PROTECTED]> wrote: ... > > AFAIK, there is no way around this with a POTS line. > SBC's only indication of MW is stutter dial tone, > correct? > No, I believe they send tones that activate the MWI light on the phones. I am pretty sure about this, but you can clear VM on SBC via a cellphone, and the MWI on the phones will go out, even though none of the phones go off hook, and therefore shouldn't be able to detect a stutter tone. SBC does give you a stutter tone, but that's not how my wife knows there is a message waiting. If you look at this website: http://www.sandman.com/messwait.html , I think it must be the case that SBC is using FSK message waiting. For that matter, my vonage ATA does the same thing - if I clear VM via the website, the light goes off even if I don't go offhook. > Since Asterisk doesn't detect ANY sort of dialtone, > either with the X100 > or TDM400, it seems you are out of luck. > AFAIK, the ATA's don't detect stutter dial tone > either, though some may > listen for dialtone before dialing. > > Perhaps some others can offer a solution. Well, I don't think I need it to detect stutter tone, but decoding the FSK MWI signal would seem a pretty common thing to want to do, and it should be pretty common in phone chipsets. > Curious, since a cheap $20 box sitting on your line > can give you a > visual indication. > >What are you doing now with the Nortel? Does it know? Yes. The Nortel Venture phones are analog phones that use a data carrier on top of line to talk to each other for things liek intercom, directory functions, etc... They support 3 lines. If they detect a message waiting on any of the 3 lines, the MWI is lit, and a text message on the phone indicates which line or lines have voicemail waiting. Very straightforward. >BTW - the Sipura 841 is an OK inexpensive phone, but >speakerphone and >display are unusable. >A somewhat better buy is the Grandstream. At least you >can read the >display. Speakerphone suffers from echo, and it has no >built in echo >canceller. >The more expensive brands are , well, too expensive >and seem difficult >to configure. Good feedback to know, I'll do some more digging on reviews, though the question may be moot if I can't find a way to communicate the message waiting signal in Asterisk. It's a show stopper in my house. I am surprised that noone has run into this issue before. Thanks, Mike __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] how to send voicemail notifcation every 15 minutes until message is checked
On 13:33, Fri 01 Jul 05, Pedro wrote: > I have searched quite a few places and have not seen this discussed. > Basically I was wondering how would you go about having an option for > a user to be notified every 15 minutes until their new voicemail > message is checked. Since the notification e-mails we send get sent > to cell phones or actual pagers (via e-mail), there are times when a > person is out of range and misses a page or just simply is too busy to > check voicemail and then forgets. They want to be reminded 15 minutes > later until that new message is checked. > > Current version of asterisk that we are running is CVS-v1-0-11/12/04 > (which has been running rock-solid I might add). Any thoughts are > appreciated. Hi, You can check the new mail count with the manager interface or by looking at the spool dir. If you put this in cron every 15 minutes, you're done. Michiel ___ 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 newbie and phones which don't want tocomunicate
hi do u have the sip phones extensions in the extension.conf and are they in the right context (sip-incoming)??? are the sip phone registering to asterisk?? try stop asterisk and reconnect as asterisk -vvvc to check see them registering... From: [EMAIL PROTECTED] on behalf of Sistemista WebSolvingJaa Sent: Fri 7/1/2005 6:43 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] asterisk newbie and phones which don't want tocomunicate Hi list, i'm an asterisk newbie and i've to setup a net with an asterisk server and several ip phones linked on the net. i hope my questions are IT ans if you have some link for solving those problems please mail me. i've wrote the sip.conf in this way: [2011] type=friend username=2011 secret=1945 regexten=1234 host=dynamic ;host=192.168.100.242 permit=192.168.100.0/24 context=sip-incoming canreinvite=yes dtmfmode=rfc2833 nat=1 [2012] type=friend username=2012 secret=1945 regexten=1234 host=dynamic ;host=192.168.100.221 permit=192.168.100.0/24 context=sip-incoming canreinvite=yes dtmfmode=rfc2833 nat=1 and the extension.conf if quitelly the same as the original. the phones softwares are setted up correctly, but from a phone i can't call another phone on the net. can somebody suggest me a possible solution? thanks a lot ___ 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 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] make error for zaptel
maybe zaptel verion incompatability try other newer or stable older versions not sure thats just a hint From: [EMAIL PROTECTED] on behalf of Zoltan Szecsei Sent: Fri 7/1/2005 2:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] make error for zaptel Hi, I'm running SuSE 9.3 fully updated using YOU and I *have* re-booted the box (in a hope to sort out the uname -r issue mentioned below). I'm using the Asterisk Doc Proj vol 1 to guide me through the initial setup. I have no special HW and intend to use asterisk on an internal network just to get some experience. I have downloaded what I think I need and placed it in /usr/src (see listing below). I run make clean ; make linux26 (what about the usual make with no parameters?) and I get a crash. Note that uname -r returns a *different* version of what the linux is linked to (thanks to YOU??) I have tried make clean ; make (no params) and it still crashes. Can anyone offer me some suggestions? - or do I go first to the SuSE list to sort out the uname -r & usr/src/linux issue? TIA, Zoltan. gl0:/usr/src # ls -la total 499 drwxr-xr-x 17 root root680 2005-06-30 14:46 . drwxr-xr-x 13 root root368 2005-06-30 09:21 .. drwxr-xr-x 3 root root320 2005-05-19 06:53 astcc -rw-r--r-- 1 root root 130943 2005-06-25 19:09 astcc.tar drwxr-xr-x 22 root root 2336 2005-06-23 04:16 asterisk-1.0.8 drwxr-xr-x 7 root root432 2005-06-23 04:21 asterisk-addons-1.0.8 drwxr-xr-x 3 root root184 2005-06-23 04:23 asterisk-sounds-1.0.8 drwxr-xr-x 2 root root440 2005-05-23 06:47 btp -rw-r--r-- 1 root root 32975 2005-06-25 19:08 btp.tar drwxr-xr-x 23 root root776 2005-06-12 17:24 dicts drwxr-xr-x 5 root root416 2005-04-01 18:50 gastman -rw-r--r-- 1 root root 332857 2005-06-25 19:08 gastman.tar drwxr-xr-x 25 root root736 2005-06-12 17:29 kernel-modules drwxr-xr-x 2 root root520 2005-06-23 04:11 libpri-1.0.8 lrwxrwxrwx 1 root root 19 2005-06-25 22:01 linux -> linux-2.6.11.4-21.7 drwxr-xr-x 3 root root 72 2005-06-25 22:01 linux-2.6.11.4-20a drwxr-xr-x 3 root root 72 2005-03-24 02:03 linux-2.6.11.4-20a-obj drwxr-xr-x 19 root root720 2005-06-25 22:01 linux-2.6.11.4-21.7 drwxr-xr-x 3 root root 72 2005-06-02 16:54 linux-2.6.11.4-21.7-obj lrwxrwxrwx 1 root root 23 2005-06-25 22:01 linux-obj -> linux-2.6.11.4-21.7-obj drwxr-xr-x 7 root root168 2005-06-12 17:43 packages drwxr-xr-x 2 root root 2720 2005-07-01 12:43 zaptel-1.0.8 gl0:/usr/src # uname -r 2.6.11.4-20a-smp gl0:/usr/src # cd zaptel-1.0.8/ gl0:/usr/src/zaptel-1.0.8 # make clean rm -f torisatool makefw tor2fw.h rm -f zttool rm -f *.o ztcfg tzdriver sethdlc sethdlc-new rm -f zonedata.lo tonezone.lo libtonezone.so.1.0 *.lo rm -f *.ko *.mod.c .*o.cmd rm -f gendigits tones.h rm -f libtonezone* rm -f tor2ee rm -f core gl0:/usr/src/zaptel-1.0.8 # make linux26 cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -c -o gendigits.o gendigits.c cc -o gendigits gendigits.o -lm ./gendigits cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\"/etc/zaptel.conf\"makefw.c -o makefw ./makefw tormenta2.rbt tor2fw > tor2fw.h Loaded 69900 bytes from file cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -c -o ztcfg.o ztcfg.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -DBUILDING_TONEZONE -o zonedata.lo zonedata.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -DBUILDING_TONEZONE -o tonezone.lo tonezone.c ar rcs libtonezone.a zonedata.lo tonezone.lo cc -o ztcfg ztcfg.o -lm -L. libtonezone.a cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -c -o torisatool.o torisatool.c cc -o torisatool torisatool.o cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -c -o ztmonitor.o ztmonitor.c cc -o ztmonitor ztmonitor.o cc -c ztspeed.c cc -o ztspeed ztspeed.o make -C /lib/modules/`uname -r`/build SUBDIRS=/usr/src/zaptel-1.0.8 modules make[1]: Entering directory `/usr/src/linux-2.6.11.4-20a-obj/i386/smp' make[1]: *** No rule to make target `modules'. Stop. make[1]: Leaving directory `/usr/src/linux-2.6.11.4-20a-obj/i386/smp' make: *** [linux26] Error 2 gl0:/usr/src/zaptel-1.0.8 # ___ 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 mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update o
Re: [Asterisk-Users] Can't build cdr_addon_mysql.
You could have just done "ln -s asterisk-1.0.9 asterisk" and it would have fixed that. It should by default do -I../asterisk /b --- Anakin: “You’re either with me, or you’re my enemy.” Obi-Wan: “Only a Sith could be an absolutist.” On Jun 30, 2005, at 1:13 AM, Chris Mason (Lists) wrote: Marcel van Kaam, Fonetica wrote: I had the same problem with installing addons. I checked out in the file cdr_addons_mysql.c what the location of the asterisk.h must be and changed the cdr_addons_mysql.c to the location of the asterisk.h file. After this it worked. Also to be sure do: locate asterisk.h to check or you have the file on your system. Marcel Yes, that worked. For the record, it had to be #include "../asterisk-1.0.9/asterisk.h" -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ 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 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] how does pattern routes works
Pattern-matching extensions must be prefaced with an underscore thus: _1NXXNXX Enjoy! On Fri, 1 Jul 2005 10:32:46 -0700 (PDT), "wassim darwish" <[EMAIL PROTECTED]> said: > i tried to write to usa destination 1* it worked well > but when i tried to specify the number of digits i > wrote > 1NXXNXX but it did'nt work.can anybody help me > please > please. > > > > > Yahoo! Sports > Rekindle the Rivalries. Sign up for Fantasy Football > http://football.fantasysports.yahoo.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 Pattern-matching extensions must be prefaced with an underscore thus: _1NXXNXX Enjoy! ___ 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] Unable to forward frame/voice
Hi, We've exhausted our internal capabilities as well as Sangoma tech support and were hoping someone with some expertise could help us with a pointer. Briefly, our issue is as follows. Periodically (several times an hour), we get either of the following error messages in our asterisk messages log. These correspond with dropped outbound calls on a one-to-one basis when the second error happens. The first error sometimes causes a dropped call and sometimes does not: Jun 30 16:40:27 WARNING[5395] app_dial.c: Unable to forward frame Jun 30 16:45:07 WARNING[5455] app_dial.c: Unable to forward voice Our hardware is as follows: Compaq DL380 Dual PIII 1Ghz, 1.2 GB RAM, Onboard SmartArray for SCSI RAID Sangoma A102U dual-port T1 card Digi Datafire T1 fax/modem board Our software is as follows: Linux 2.4.30 Asterisk, Zaptel and Libpri from CVS HEAD as of 6/28/05 Sangoma wanpipe 2.3.3-beta11 (latest as of this post) Patton electronic's latest drivers and firmware for our Digi Datafire board (still no 2.6 Linux support, which is why we're on 2.4) Hylafax 4.2.1 driving the Digi Datafire The path (for the problem calls) looks like this: Digi Datafire -> Sangoma Port B -> Sangoma Port A -> Telco Basically, sending a fax over a PRI with asterisk doing TDM bridging in the middle. We have confirmed the following (based on similar posts to this list related to the same problem with Digium boards as well as Sangoma tech support assistance): 1. Sangoma Port A takes clocking from the telco 2. Sangoma Port B retransmits A's clocking and acts as master 3. Sangoma tech support says our configs are correct 4. Zaptel.conf is set up with Sangoma Port A as the primary clock source, and Port B to not be used as a clock source 5. LBO, switch options, etc. are correct for the environment (since 98% of outbound calls are fine, this seems fairly obvious) 6. ISDN Transfer Capability gets properly set to 3K1AUDIO for calls 7. No IRQ sharing on the system 8. IDE DMA mode is irrelevant, since there are no IDE disks in the system (other than the CDROM) We have tried the following: 1. Asterisk, libpri and zaptel versions from 6/1/2005, 6/15/2005 and 6/28/2005 - no change in behavior 2. Wanpipe drivers 2.3.3-beta8 and 2.3.3-beta11 - no change in behavior 3. Wanpipe configured both with and without the D-Channel hardware HDLC - no change in behavior 4. Firmware versions 24 (shipped version) and 25 (latest version) on the Sangoma card - no change in behavior 5. callprogress and busydetect both 'yes' and 'no' in zapata.conf (currently 'no') - no change in behavior 6. Added SetTransferCapability(3K1AUDIO) to our dialplan, just to be sure - no change in behavior General environment: 1. We run TDM only, no VoIP protocols are in use. SIP, IAX2, MGCP are all noload in modules.conf. 2. This problem occurs with as few as one simultaneous channel active and as many as 15 simultaneous channels active with equal frequency (i.e.: not load related). The load on the box is negligible in any case, plenty of RAM is free, etc. 3. Restarting asterisk does seem to cause the problem not to re-present itself for 30 minutes to 2 hours. When asterisk is restarted, the Sangoma and Zaptel kernel modules are also unloaded and reloaded. Again, any pointers or help would be greatly appreciated. Thanks, Christian ___ 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 does pattern routes works
i tried to write to usa destination 1* it worked well but when i tried to specify the number of digits i wrote 1NXXNXX but it did'nt work.can anybody help me please please. Yahoo! Sports Rekindle the Rivalries. Sign up for Fantasy Football http://football.fantasysports.yahoo.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 HELP HELP NEEDED WITH ASTCC
Ade Agbero wrote: Finally, We have lift off, a shaky one though. I deleted my Astcc.gi and replaced it with Darren's copy posted on his website and I have finally been able to get something recorded as BILLCOST. I got it working too here with Darren's astcc.agi. And billing as expected so finally It's working. It would be nice if someone could update the cvs with Darren's astcc.agi, because the current one doesn't work, even patched.. it gets worse. Thanks for your attention Darren! -- Juan Luis Moyano [EMAIL PROTECTED] ___ 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 send voicemail notifcation every 15 minutes until message is checked
I have searched quite a few places and have not seen this discussed. Basically I was wondering how would you go about having an option for a user to be notified every 15 minutes until their new voicemail message is checked. Since the notification e-mails we send get sent to cell phones or actual pagers (via e-mail), there are times when a person is out of range and misses a page or just simply is too busy to check voicemail and then forgets. They want to be reminded 15 minutes later until that new message is checked. Current version of asterisk that we are running is CVS-v1-0-11/12/04 (which has been running rock-solid I might add). Any thoughts are appreciated. ___ 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] Voicemail => SMS
On 01/07/05, Mark Charlton <[EMAIL PROTECTED]> wrote: > > > I have been fighting with the Bayham Systems FastSMS AGI script, and I > re-wrote it as a stand alone Perl script. I am now calling it with > the EXTERNNOTIFY option in the voicemail.conf file. It gets passed > the context, extension and number of messages which I build into a > text, and since they all go to the same location its no problem. I'm > planning on using the extension info to open the mailbox, and read the > text file for the latest message to pull out the caller for the text. > I might also have an extension map in a text file so I can look up who > to notify about a VM. I also hacked Bayham Systems' script. I need to control the MWI on GSM phones, which is turned on and off by custom SMS messages (which Bayham helpfully provide macros for). And I needed to avoid sending repeat notifications when a second or subsequent new message was left. I ended up keeping a flag in a db file which stores per mobile number what state the MWI is in for that phone. If the flag says the MWI is already set when a new message comes in, then the script is a no-op, otherwise it sends the 'MWI on' SMS and flips the flag. Ditto in reverse. Works fine and dandy here. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] ___ 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] Got this error after my installation when i doztcfg -vv
hello u can see the readme.udev in the zaptel directory that's normally answers ur question From: [EMAIL PROTECTED] on behalf of Ian Bert Tusil Sent: Fri 7/1/2005 9:37 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Got this error after my installation when i doztcfg -vv how can i solve the error on the last part? need help. thnx... Zaptel Configuration == SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Slaves: 02) Channel 03: Individual Clear channel (Default) (Slaves: 03) Channel 04: Individual Clear channel (Default) (Slaves: 04) Channel 05: Individual Clear channel (Default) (Slaves: 05) Channel 06: Individual Clear channel (Default) (Slaves: 06) Channel 07: Individual Clear channel (Default) (Slaves: 07) Channel 08: Individual Clear channel (Default) (Slaves: 08) Channel 09: Individual Clear channel (Default) (Slaves: 09) Channel 10: Individual Clear channel (Default) (Slaves: 10) Channel 11: Individual Clear channel (Default) (Slaves: 11) Channel 12: Individual Clear channel (Default) (Slaves: 12) Channel 13: Individual Clear channel (Default) (Slaves: 13) Channel 14: Individual Clear channel (Default) (Slaves: 14) Channel 15: Individual Clear channel (Default) (Slaves: 15) Channel 16: Individual Clear channel (Default) (Slaves: 16) Channel 17: Individual Clear channel (Default) (Slaves: 17) Channel 18: Individual Clear channel (Default) (Slaves: 18) Channel 19: Individual Clear channel (Default) (Slaves: 19) Channel 20: Individual Clear channel (Default) (Slaves: 20) Channel 21: Individual Clear channel (Default) (Slaves: 21) Channel 22: Individual Clear channel (Default) (Slaves: 22) Channel 23: Individual Clear channel (Default) (Slaves: 23) Channel 24: D-channel (Default) (Slaves: 24) 24 channels configured. Notice: Configuration file is /etc/zaptel.conf line 145: Unable to open master device '/dev/zap/ctl' ___ 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 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] Epia C3 Linux
Wiley Siler wrote: >Oliver, > >Thanks for the response! Do you know where I can find an example of how >to do this? I have never had to install a custom kernel before. > > For Gentoo there is a superb dokumentation on http://www.gentoo.org/doc/en/index.xml to do this. Regards, Oliver ___ 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] Epia C3 Linux
Oliver, Thanks for the response! Do you know where I can find an example of how to do this? I have never had to install a custom kernel before. Thanks! Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Oliver Rath Sent: Friday, July 01, 2005 10:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Epia C3 Linux Wiley Siler wrote: > Anyone know a good distro for an Epia Mobo with the C3 chip? > > > > I have been trying to get Debian and Gentoo installed (new to me) and > so far having little luck. > > > > Does anyone know a good install for this processor/mobo combo? > > > You have to compile without mmx and sse, best 586compatible, because linux is recognizing C3 as PIII, what is definitly wrong. Hth, Oliver ___ 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 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] Visual ring notification
I had the same problem in a welding lab. I did this on a 3Com NBX, but I'm sure that the same idea would apply to Asterisk: I went to RadioShack, and bought one of their "visual ringers", for the hearing impaired (basically flashes a white strobe light, and sounds a really loud ringer), and attached it to an ATA adapter. Then I created a flat call group, and reassigned its number to what the single phone's extension used to be. That way, whenever anyone would dial that extension, the strobe would activate and the louder ringer would sound, in addition to the telephone. Andrew M Stemen [EMAIL PROTECTED] http://www.andrewmstemen.com Chris Mason (Lists) wrote: I have put a pbx into a resort with Polyycom phones, everythign works great, except the kitchen staff cannot hear the phone ring. I know many legacy systems employ a big red flashing light, any ideas on doing something similar? ___ 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 HELP HELP NEEDED WITH ASTCC
Finally, We have lift off, a shaky one though. I deleted my Astcc.gi and replaced it with Darren's copy posted on his website and I have finally been able to get something recorded as BILLCOST. I say "something" because the amount I expected and amount entered are different. Please see below, I expect any call less than 60secs to generate a 10 charge (see caller id 1234), but the amount being generated doesn't follow the charges set in Routes (see caller id 2, 3, 4): ANY IDEAS WHY??? Caller*ID Called Number Trunk Disposition Billable Seconds Billed Cost <1234> 19313256895 DANSAM ANSWER 14 10 <2> 19313256895 DANSAM ANSWER 4 1 <3> 19313256895 DANSAM ANSWER 48 8 <4> 19313256895 DANSAM ANSWER 21 4 Pattern Comment Trunks Connect Fee Inc. Seconds Cost per additional minute 44.* DANSAM 0 0 10 1.* DANSAM 0 0 10 Darren Wiebe <[EMAIL PROTECTED]> wrote: Could you please post the output from the asterisk console when astcc.agi crashes? I really would like to get this resolved.Darren Wiebe[EMAIL PROTECTED]Juan Luis Moyano wrote:>Ade Agbero wrote:> >>>I tried using your working astcc.agi file instead of mine, but that>>failed to work too. >>>Having the same issues here.. it seems astcc.agi is crashing.>> >___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users How much free photo storage do you get? Store your holiday snaps for FREE with Yahoo! Photos. Get Yahoo! Photos___ 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] Epia C3 Linux
Wiley Siler wrote: > Anyone know a good distro for an Epia Mobo with the C3 chip? > > > > I have been trying to get Debian and Gentoo installed (new to me) and > so far having little luck. > > > > Does anyone know a good install for this processor/mobo combo? > > > You have to compile without mmx and sse, best 586compatible, because linux is recognizing C3 as PIII, what is definitly wrong. Hth, Oliver ___ 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: passing through MWI info from SBC
Mike Myers wrote: John Novack wrote: Mike Myers wrote: Hi.. I am about to replace my aging Nortel Venture system with an Asterisk system and 6 Polycom IP 501 phones, and a couple sipura 841's for less used areas. We have 3 phone lines here. One is SBC, one Vonage, and one Voipjet... One hangup is that I can't figure out how to pass through a voicemail waiting indication from SBC. This is important because my wife and her family all exchange voicemails with each other on the SBC voicemail system. They can leave messages for each other without having the phones ring, etc... We have a 2 yr old at home, and her sister has some small kids too, so that's how they manage to send voicemails when they are unsure if the kids are sleeping, etc... Anyway, preserving this capability of using the SBC VM and being notified when a message is waiting is critical for good WAF. The vonage line and voipjet line can be intergrated into the Asterisk VM. My Nortel venture phones light the MWI if any line has VM on it, and the display tells you which lines have VM waiting. I would love to be able to duplicate this function on the Polycom's and hopefully the Sipura's as well. I've looked for answers on this, but haven't found one, hence the post. My apologies if I have missed something. Thanks much, Mike You haven't missed much. With SBC you are out of luck, since Asterisk doesn't detect dialtone ( it dials blind, sometimes too quickly for the CO to catch the first digit, resulting in wrong numbers )) or stutter dialtone either, and reportedly has had any indication of the DC status of a POTS line removed due to problems. Only choice would to port the number to a VOIP provider and provide the VM in Asterisk. Similar problem with Vonage VM. John Novack Wow, this is a serious problem for me. I don't need to actually check the voicemail itself from Asterisk, just to be able to tell that there is voicemail waiting. Are you saying there is no way in Asterisk to do this? Is that true for using Digium hardware as well as FXO ports on a SIP ATA? Vonage VM doesn't matter to me, since I'll turn it off and use Asterisk for that functionality, but determining SBC's VM status is very important. My whole wife's family (multiple households) uses it. In the past, if one family tried to switch to a non SBC provider, they always returned in less than a week because of lack of VM interoperation. So my wife will put the kibosh on the whole Asterisk project unless I can light the MWI light when SBC VM is waiting. Since the cheapest analog phones can do this, I don't think she's going to understand that these $200 Polycom phones can't... :-( Is there no way around this? Thanks, Mike AFAIK, there is no way around this with a POTS line. SBC's only indication of MW is stutter dial tone, correct? Since Asterisk doesn't detect ANY sort of dialtone, either with the X100 or TDM400, it seems you are out of luck. AFAIK, the ATA's don't detect stutter dial tone either, though some may listen for dialtone before dialing. Perhaps some others can offer a solution. Curious, since a cheap $20 box sitting on your line can give you a visual indication. What are you doing now with the Nortel? Does it know? BTW - the Sipura 841 is an OK inexpensive phone, but speakerphone and display are unusable. A somewhat better buy is the Grandstream. At least you can read the display. Speakerphone suffers from echo, and it has no built in echo canceller. The more expensive brands are , well, too expensive and seem difficult to configure. JMO John Novack ___ 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] Epia C3 Linux
Anyone know a good distro for an Epia Mobo with the C3 chip? I have been trying to get Debian and Gentoo installed (new to me) and so far having little luck. Does anyone know a good install for this processor/mobo combo? Thanks Wiley ___ 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: [asterisk] VocTel service provider
I haven't heard much feedback yet - anyone here using VocTel? The connection problem turned out to be my firewall, but I'm curious if others experience any voice choppiness or high latency. Some posters have related the problem to specific VOIP providers, some seem to be ISP related (local network latency). Any feedback? OCG ___ 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] Computer to use
On Jul 1, 2005, at 4:03 AM, Eric Wieling aka ManxPower wrote: Robert Goodyear wrote: I'm sure you really only want to know about the absence of problems. From watching this list for 6 months it seems the SuperMicro products are most lauded and have exhibited no hardware conflicts. Various votes on Dell products, so you're probably best to stay away, even though I've got five installs with TE110Ps in them that have never missed a beat -- Dimension boxes, not PowerEdge. The SuperMicro Xeon board we tried failed miserably with both the T100P and TE110P. It had the ServerWorks IDE Chipset, which I suspect was the problem. -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 Bummer! I thought I'd heard all good things about them... sorta like VoIP providers; as soon as everyone agrees things are OK, something goes awry! -Rob. ___ 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] Sometimes yes - sometimes no (dialplan)
On Jul 1, 2005, at 1:47 AM, Ronald_Wiplinger wrote: I am confused about one of my installed server The dial plan seems to be ok, but sometimes NOTHING happens if I try to dial an extension (from X-Lite), next time it is done. X-Lite does not have a tone, nothing and does also have no time out. It seems it is not connected to the server. However, a sip show users / sip show peers shows that the phone is connected. SIP clients generate their own dialtone, so if you've got no tone, that sounds suspicious of a problem with the client itself. I assume you've debugged the problem by registering a hard SIP client on that server? -- Robert Goodyear Brand Up LLC http://www.brand-up.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] Visual ring notification
Analog relay in the same ring group with a bell? Cheers, Dean > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Chris Mason (Lists) > Sent: Friday, 1 July 2005 12:09 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Visual ring notification > > I have put a pbx into a resort with Polyycom phones, everythign works > great, except the kitchen staff cannot hear the phone ring. I know many > legacy systems employ a big red flashing light, any ideas on doing > something similar? > > -- > Chris Mason > NetConcepts > (264) 497-5670 Fax: (264) 497-8463 > Int: (305) 704-7249 Fax: (815)301-9759 > Cell: 264-235-5670 > Yahoo IM: [EMAIL PROTECTED] > > ___ > 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 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] Visual ring notification
I have put a pbx into a resort with Polyycom phones, everythign works great, except the kitchen staff cannot hear the phone ring. I know many legacy systems employ a big red flashing light, any ideas on doing something similar? FYI, the Uniden UIP200 has a big red flashing light. Regards, -- Jason Becker Director & CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ 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] Quality of provider: VocTel
On Wednesday 29 June 2005 23:36, Michael Stahl wrote: > How have you found the quality (Choppy / smooth audio)? > Any problems registering? (I have been unable to register for hours) I use them for some of my termination, they seem to work just fine (no quality/registration issues). Actually once I did have a problem where I couldn't seem to get any calls out but power-cycling *MY* firewall fixed it, so it certainly wasn't an issue on their end. -A. ___ 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:passing through MWI info from SBC
John Novack wrote: >Mike Myers wrote: >>Hi.. I am about to replace my aging Nortel Venture >>system with an Asterisk system and 6 Polycom IP 501 >>phones, and a couple sipura 841's for less used areas. >> >>We have 3 phone lines here. One is SBC, one Vonage, >>and one Voipjet... One hangup is that I can't figure >>out how to pass through a voicemail waiting indication >>from SBC. This is important because my wife and her >>family all exchange voicemails with each other on the >>SBC voicemail system. They can leave messages for >>each other without having the phones ring, etc... We >>have a 2 yr old at home, and her sister has some small >>kids too, so that's how they manage to send voicemails >>when they are unsure if the kids are sleeping, etc... >>Anyway, preserving this capability of using the SBC VM >>and being notified when a message is waiting is >>critical for good WAF. >> >>The vonage line and voipjet line can be intergrated >>into the Asterisk VM. My Nortel venture phones light >>the MWI if any line has VM on it, and the display >>tells you which lines have VM waiting. I would love >>to be able to duplicate this function on the Polycom's >>and hopefully the Sipura's as well. >> >>I've looked for answers on this, but haven't found >>one, hence the post. My apologies if I have missed >>something. >> >>Thanks much, >>Mike > > >You haven't missed much. >With SBC you are out of luck, since Asterisk doesn't >detect dialtone ( >it dials blind, sometimes too quickly for the CO to >catch the first >digit, resulting in wrong numbers )) or stutter >dialtone either, and >reportedly has had any indication of the DC status of a POTS line >removed due to problems. >Only choice would to port the number to a VOIP >provider and provide the >VM in Asterisk. >Similar problem with Vonage VM. >John Novack Wow, this is a serious problem for me. I don't need to actually check the voicemail itself from Asterisk, just to be able to tell that there is voicemail waiting. Are you saying there is no way in Asterisk to do this? Is that true for using Digium hardware as well as FXO ports on a SIP ATA? Vonage VM doesn't matter to me, since I'll turn it off and use Asterisk for that functionality, but determining SBC's VM status is very important. My whole wife's family (multiple households) uses it. In the past, if one family tried to switch to a non SBC provider, they always returned in less than a week because of lack of VM interoperation. So my wife will put the kibosh on the whole Asterisk project unless I can light the MWI light when SBC VM is waiting. Since the cheapest analog phones can do this, I don't think she's going to understand that these $200 Polycom phones can't... :-( Is there no way around this? Thanks, Mike Yahoo! Sports Rekindle the Rivalries. Sign up for Fantasy Football http://football.fantasysports.yahoo.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] Linux Firewall Question
You should be able to do a good job with IPTABLES which is included in FC3. You can limit source & destp IP and protocol, etc. Type "man iptables | more" for more details... OCG -Original Message- From: Terry H. Gilsenan [mailto:[EMAIL PROTECTED] Sent: Thursday, June 30, 2005 8:11 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Linux Firewall Question > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of OMS > Sent: Friday, 1 July 2005 9:56 AM > To: Asterisk-Users@lists.digium.com > Subject: [Asterisk-Users] Linux Firewall Question > > Hi, > I am running Asterisks on Public IP with Fedora Core 3. > > What is the recommendation for making Linux secure on the Public IP > since I am new to Linux. Which Firewall should I use? I am not > intending to use Linux as router. > > Can any one provide some configuration documentation. I use shorewall, and I have found it powerful, and fairly easy to use. http://www.shorewall.net/ T ___ 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] Visual ring notification
How good is your electrical engineering? ..o---o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Mason (Lists) Sent: Friday, July 01, 2005 12:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Visual ring notification I have put a pbx into a resort with Polyycom phones, everythign works great, except the kitchen staff cannot hear the phone ring. I know many legacy systems employ a big red flashing light, any ideas on doing something similar? -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ 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 This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ 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] Voicemail => SMS
On 7/1/05, Wilson Pickett <[EMAIL PROTECTED]> wrote:
> > I have been trying for a while to find a way to get an SMS send when I
> > receive a voicemail into my asterisk system. I don't want to send an
> > SMS if the caller doesn't leave a message. I have voicemail.conf set
> > up to email and delete.
>
> I use a backward solution to this problem, but it works. Orange, my
> cell provider offers free SMS alerts for email sent to
> [EMAIL PROTECTED] I send my vmail messages to my regular email
> server which keeps them for online email retrieval. A procmail recipe
> on the server then makes up an email without the vmail attachment to
> my orange address with the callerid in the subject. Orange sends an
> SMS that tells me I have a vmail message from ${CALLERID}. Although it
> seems like a silly solution it does _exactly_ what you asked about.
>
I have been fighting with the Bayham Systems FastSMS AGI script, and I
re-wrote it as a stand alone Perl script. I am now calling it with
the EXTERNNOTIFY option in the voicemail.conf file. It gets passed
the context, extension and number of messages which I build into a
text, and since they all go to the same location its no problem. I'm
planning on using the extension info to open the mailbox, and read the
text file for the latest message to pull out the caller for the text.
I might also have an extension map in a text file so I can look up who
to notify about a VM.
This works after a fashion, and crucially is only triggered when
someone actually leaves a valid voice mail message. It is limited in
the fact I can't pass any other system details than extn and context.
Plus the voicemail count is wrong since the attach=yes|delete=yes has
already deleted the message when it counts them.
But it works.
Thanks for all the help and advice.
Mark
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