[Asterisk-Users] Re: I would like to UNsubscribe from this list thanks
Jason Konik
jkonik at konik.ca
Fri Mar 19 11:17:46 MST 2004
On Fri, 19 Mar 2004 13:06:16 -0600, asterisk-users-request wrote
> Send Asterisk-Users mailing list submissions to
> asterisk-users at lists.digium.com
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://lists.digium.com/mailman/listinfo/asterisk-users
> or, via email, send a message with subject or body 'help' to
> asterisk-users-request at lists.digium.com
>
> You can reach the person managing the list at
> asterisk-users-admin at lists.digium.com
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of Asterisk-Users digest..."
>
> Today's Topics:
>
> 1. Re: firefly softphone (Dave Cotton)
> 2. Asterisk Voice Mail Integration with Cisco CME (Kurt Pasewaldt)
> 3. RE: firefly softphone (James and Melody Alspach)
>
> 4. Re: Important: The Asterisk Mailing list (new subject)
> (Tilghman Lesher)
> 5. RE: Speaking of ring tones... (Kevin Pearcey)
> 6. DID with X100P? (Victor Perez)
> 7. RE: Asterisk-Users digest, Vol 1 #3157 - 11 msgs (George Bean)
> 8. RE: Can i do voice chat without using the hardware (Chris
> Albertson)
> 9. g729 suggestions? (Rich Adamson)
> 10. Re: Identifying a call with manager interface (Maciek Kaminski)
> 11. Re: Problems with asterisk and gnophone on Gentoo box (Kevin)
> 12. Re: Important: The Asterisk Mailing list (new subject) (Brian
> Capouch)
>
> --__--__--
>
> Message: 1
> Subject: Re: [Asterisk-Users] firefly softphone
> From: Dave Cotton <dcotton at linuxautrement.com>
> To: Asterisk List <asterisk-users at lists.digium.com>
> Date: Fri, 19 Mar 2004 17:47:55 +0100
> Reply-To: asterisk-users at lists.digium.com
>
> On Fri, 2004-03-19 at 17:31, Nick Knight wrote:
> > Hello all,
> >
> >
> >
> > I have tried the firefly softphone on a couple of computers now – and
> > as soon as it registers with the Asterisk server (in fact tries to
> > register) but then crashes and tries to send crash report to MS.
>
> > Has any one had experience of this.
>
> IIRC it's because no codecs have been selected.
>
> --
> Dave Cotton <dcotton at linuxautrement.com>
>
> --__--__--
>
> Message: 2
> Date: Fri, 19 Mar 2004 09:21:01 -0800 (PST)
> From: Kurt Pasewaldt <kurtwp at yahoo.com>
> To: asterisk-users at lists.digium.com
> Subject: [Asterisk-Users] Asterisk Voice Mail Integration with Cisco
> CME Reply-To: asterisk-users at lists.digium.com
>
> Since the voice mail portion of CME is an additional
> charge, I was wondering has any body use Asterisk
> voice mail with Cisco CME.
>
> Kurt.
>
> __________________________________
> Do you Yahoo!?
> Yahoo! Mail - More reliable, more storage, less spam
> http://mail.yahoo.com
>
> --__--__--
>
> Message: 3
> Date: Fri, 19 Mar 2004 09:22:52 -0800
> From: James and Melody Alspach <alspachfam at charter.net>
> To: asterisk-users at lists.digium.com
> Subject: RE: [Asterisk-Users] firefly softphone
> Reply-To: asterisk-users at lists.digium.com
>
> Long time listener, first time caller ;-)
>
> I ran into this message after when I first installed it (although I
> let the install start and then sit in the background waiting for me
> to set it up, for about 6 hours while I worked on something else, so
> I am not sure if something could have happened there.). When I
> reinstalled, I got some other freaky messages about something thhat
> i do not remember. Finally, I went intot he reg. and removed
> anything that said firefly. Then I was able to reinstall without
> issue and the software seems to work great (now if I could just
> figure out what 'Max retries exceeded on call' means and how I can
> get it to actually register, I will be set :-) ) I have since
> found this link that tells you how to remove firefly for a clean reinstall.
> http://www.virbiage.com/firefly/help/remove.php
>
> Hope a noob could be of some help :-)
>
> James
>
> ~~~~~~~~~~~
>
> Message: 9
> Date: Fri, 19 Mar 2004 16:31:56 -0000
> From: "Nick Knight" <nick at omniis.com>
> To: <asterisk-users at lists.digium.com>
> Subject: [Asterisk-Users] firefly softphone
> Reply-To: asterisk-users at lists.digium.com
>
> Hello all,
>
> I have tried the firefly softphone on a couple of computers now -
> and as soon as it registers with the Asterisk server (in fact tries
> to register) but then crashes and tries to send crash report to MS.
> Has any one had experience of this.
>
> Nick
>
> --__--__--
>
> Message: 4
> From: Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] Important: The Asterisk Mailing list
> (new subject)
> Date: Fri, 19 Mar 2004 11:27:40 -0600 Reply-To: asterisk-
users at lists.digium.com
>
> On Friday 19 March 2004 02:16, Brian Capouch wrote:
> > Olle E. Johansson wrote:
> > > Do *not* send out personal replies on the list.
> >
> > Yes! Yes!! Yes!!!
> >
> > Let's change the way the list software works so people won't get
> > hammered by replying and rid this list of that pox once and for
> > all.
>
> No, no, no. Then everytime somebody hits "Reply All", the poster
> gets two messages: one from the poster, one from the listserv. And
> subsequent reply-all's add to this problem. Let's not make the
> situation worse.
>
> -Tilghman
>
> --__--__--
>
> Message: 5
> From: "Kevin Pearcey" <kevin at loon.org.uk>
> To: <asterisk-users at lists.digium.com>
> Subject: RE: [Asterisk-Users] Speaking of ring tones...
> Date: Fri, 19 Mar 2004 17:28:21 -0000
> Reply-To: asterisk-users at lists.digium.com
>
> Has anyone figured out how to change the volume of the ring tone on
> the snom 200?
>
> Its pretty loud in our office when a group of these things start to ring
> together.
>
> Kev
>
> > I kinda like it .. ;)
> > Nice & conservative.
> > OTOH, the new snom 200 I just got today has some reeeaaally
> > weird ring tones (and nothing really 'traditional'). Now,
> > maybe we should take a lesson from the cell-phone people, and
> > talk manufacturers into letting us download ringtone(s). Cheers,
> > WW
>
> --__--__--
>
> Message: 6
> From: Victor Perez <vperez at bkglobal.com>
> To: asterisk-users at lists.digium.com
> Date: Fri, 19 Mar 2004 11:34:37 -0600
> Subject: [Asterisk-Users] DID with X100P?
> Reply-To: asterisk-users at lists.digium.com
>
> Is there a way to use an X100P as a trunk with DID numbers and all?
>
> We just bought one of these and want to create some VoIP extensions connect=
> ed to our PBX as a trial. The PBX does not have capacity for any
> more T1 ca= rds so it is the only cheap way for this trial.
>
> If not, what kind of hardware would you recommend to setup some
> analog exte= nsions as DID trunks between a PBX and *?
>
> Thanks in advance,
> Victor Perez
>
> --__--__--
>
> Message: 7
> From: "George Bean" <gbean at puwaba.com>
> To: <asterisk-users at lists.digium.com>
> Date: Fri, 19 Mar 2004 12:36:04 -0500
> Subject: [Asterisk-Users] RE: Asterisk-Users digest, Vol 1 #3157 -
> 11 msgs Reply-To: asterisk-users at lists.digium.com
>
> Digi-Key has these fuses for $2.21 each. You can check them out at:
>
> http://www.digikey.com/scripts/DkSearch/dksus.dll?Detail?Ref=73728&Row=46959
> &Site=US
>
> Digi-Key is geared more toward small shipments (repairs, development
> and short run production) whereas Newark is more interested in large
> quantity sales. They can be a useful resource for many network &
> telecom projects.
>
> Regards,
> George Bean
> Puwaba Technologies
>
> On Fri, 19 Mar 2004, Jacques Leisy wrote:
>
> >>Sorry for a very stupid question, but I cannot find a supplier anywhere.
> >>
> >>Where can I buy the 3 Amps GMT fuses for the Adtran's PSU.
> >>
> >>Car fuse don't seems to fit. What is GTM the abbreviation of
>
> On Fri, 19 Mar 2004, Steve Creel wrote:
>
> >A good question (that I wish had been in the archives when I went
> >looking). You need a 3 amp GMT fuse.
>
> >Datasheet:
> >http://www.bussmann.com/library/bifs/5008.pdf
>
> >I bought a couple (and probably overpaid - $3.18 ea) at:
> >http://www.newark.com/NewarkWebCommerce/newark/en_US/support/catalog/produc
> >tDetail.jsp?id=02B3398
>
> >I think I saw them somewhere else for alot less, just don't remember
> >where.
>
> >Good luck,
> >Steve
>
> --__--__--
>
> Message: 8
> Date: Fri, 19 Mar 2004 09:38:57 -0800 (PST)
> From: Chris Albertson <chrisalbertson90278 at yahoo.com>
> Subject: RE: [Asterisk-Users] Can i do voice chat without using
> the hardware To: asterisk-users at lists.digium.com Reply-To: asterisk-
> users at lists.digium.com
>
> --- David J Carter <david.carter at codepipe.com> wrote:
> >
> >
> > My aim is that, i want to connect my PC (where i
> > installed the asterisk) to another PC in my network
> > for voice chating. For this purpose, what are the
> > steps to
> > be done? which are the files to be modified. I would
> > like to make use of the existing Hardware (sound card,
> > network card etc), i am not using any extra hardware.
> > Is X-Lite work in Linux? or any compatible s/w that
> > works under linux?
> >
> > Have a look at these sites: -
> >
> >
> > > http://www.codepipe.com/id25.htm
> > > http://www.jaredsmith.net/misc/hgta/
> > > http://www.wwworks-inc.com/asterisk/
> > > http://www.fnords.org/~eric/asterisk/
> > > http://bcwireless.net/moin.cgi/VoIPHowTo
> > > http://www.automated.it/guidetoasterisk.htm
> > > http://www.asterisk.org/index.php?menu=support
> > > http://www.voip-info.org/wiki-Asterisk+config+files
> > > http://www.voip-info.org/tiki-index.php?page=Asterisk
> > >
> > If you have the CLI> prompt then your almost there.
> >
> > If you have the audio set up in asterisk then you can use a
> > headset/microphone to call the other party.
> >
> > CLI>dial 1234
> >
> > when finished
> >
> > CLI>hangup
> >
> > Simple huh?
> >
> > Regards
> >
> >
> > Dave
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at 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
>
> =====
> Chris Albertson
> Home: 310-376-1029 chrisalbertson90278 at yahoo.com
> Cell: 310-990-7550
> Office: 310-336-5189 Christopher.J.Albertson at aero.org
> KG6OMK
>
> __________________________________
> Do you Yahoo!?
> Yahoo! Mail - More reliable, more storage, less spam
> http://mail.yahoo.com
>
> --__--__--
>
> Message: 9
> Date: Fri, 19 Mar 2004 10:11:38 -0600
> From: Rich Adamson <radamson at routers.com>
> To: Asterisk-a-users-list <asterisk-users at lists.digium.com>
> Subject: [Asterisk-Users] g729 suggestions?
> Reply-To: asterisk-users at lists.digium.com
>
> Running * stable from CVS-02/17/04 with multiple C7960's (sip behind
> nat on Internet), x100p's, multiple iax links across net, etc. About
> a dozen local sip hardphones including Snom 200 near *. IDE drives
> (no scsi).
>
> Thinking about moving the internet C7960's to g729, and seem to be coming
> up with lots of opinions in the archives, but not much in terms of
definitive
> answers. Also checked the wiki.
>
> If I only move "two" C7960's on the Internet to g729, is the correct
> calculation for number of licenses: 2 - C7960 sip channel licenses
> (assuming both will be in use at the same time to source a g729
> call, regardless whether the destination is a g711 7960,
> iax/gsm call, etc.) 1 - Voicemail (gsm disk format now) 3 - Total
> licenses
>
> Is a license needed for an "Internet C7960 g729" -> x100p pstn call?
>
> Is a license required for any other "internal" asterisk function
> (C7960 to IVR, C7960 to MOH, etc)?
>
> Rich
>
> --__--__--
>
> Message: 10
> Date: Fri, 19 Mar 2004 18:50:31 +0100
> From: Maciek Kaminski <maciejka at tiger.com.pl>
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] Identifying a call with manager interface
> Reply-To: asterisk-users at lists.digium.com
>
> Nicolas Bougues wrote:
>
> >Dear all,
> >
> >I'm trying to play with the manager interface.
> >
> >What I'd like to do is being able to originate a call and trace its
> >status through events.
> >
> >I use the "Originate" manager command. I then receive several events
> >telling me about the progress of the call, and then the "Response"
> >message.
> >
> >However, I didn't find a way to be sure that the first "Event" I
> >receive after the "Originate" really relates the call I'm making, and
> >not some other random call, since I believe that I may get events for
> >any channel, not just mine.
> >
> >Note that the Channel I'm using is IAX based, and looks like this :
> >IAX2[217.146.224.41:4569]/3 in the events messages. So I have no way
> >to know it's really mine.
> >
> >Event the final Response message doesn't state the "UniqueId" of the
> >call.
> >
> >Maybe I missed something obvious.
> >
> >Any idea ?
> >
> >
> Currrent manager originate behavior looks a little hacky. First it
> is blocking and may last for tenths of second what with fact that
> manager interface isn't concurrent(see http://www.voip-info.org/wiki-
> Asterisk+manager+experience) narrows range of originate
> applications. Secondly one can't get channel name that originate
> created. To straighten things out originate should be made
> asynchronous and make identyfing channel name via events possible.
>
> P.S.: There is a patch in mantis
> (http://bugs.digium.com/bug_view_page.php?bug_id=0000772) that makes
> originate asynchronous but it has not been approved yet.
>
> Maciek Kaminski
>
> --__--__--
>
> Message: 11
> From: Kevin <asterisk at gnosys.biz>
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] Problems with asterisk and gnophone on
> Gentoo box
> Date: Fri, 19 Mar 2004 12:53:04 -0500 Reply-To: asterisk-
users at lists.digium.com
>
> On Friday 19 March 2004 08:42, Alastair Maw wrote:
> > Mine:
> > > snd-pcm 60960 0 [snd-via82xx snd-pcm-oss]
> >
> > Yours:
> > > snd-pcm 65828 0 [snd-pcm-oss]
> >
> > Note that you don't actually have a sound driver loaded there! You
> > should have snd-nvaudio listed.
>
> Hi Alastair and John-
>
> With this message, I'm reporting at least partial success. This is
> shaping up to be a longish message, though, so I'll apologize for
> that in advance.
>
> Thanks again to you both for your replies.
>
> Alastair, I think that your post here has turned me on to the right
> answer, although I'm still working on verifying that.
>
> Your mention of OSS working under xmms and the absence of my nvaudio
> driver in the snd-pcm line prompted me to look hard at my nvaudio
> driver. It's from nVidia corp, and from studying the source code
> from the driver, I get the distinct impression that it's an OSS
> driver, not an ALSA driver (though it never comes out and says
> either way).
>
> The ALSA matrix recommends the intel8x0 driver for nForce chipsets,
> so I'm giving it a try. At first blush, I think it's working, but
> there are problems with KDE now:
>
> KDE is constantly complaining with a dialog box:
>
> Sound server fatal error:
> cpu overload, aborting
> [OK]
>
> This in spite of me disabling the sound system in kcontrol.
>
> artsd keeps starting and restarting for no apparent reason and/or
> whenever I use a KDE sound app like JuK---perhaps that's what it's
> supposed to do, but why it's overloading the cpu is still a mystery
> (and looking at top verifies that this is indeed true). I've read
> about disabling arts altogether in KDE and using alsa to take it's
> place somehow, but I'll have to research that elsewhere.
>
> xmms works with no apparent problems; with the libOSS.so and the
> libALSA.so plugins.
>
> Most importantly, asterisk does seem to be working now, although the
> Allison Smith Voice is extremely hiccupy.
>
> gnophone still seems iffy. Sometimes I start it and get:
> bash-2.05b# gnophone
> Loaded and activated '/usr/lib/gnophone/modules/audio-oss.so'
> Loaded and activated '/usr/lib/gnophone/modules/audio-phone.so'
> iax.c line 654 in iax_init: Started on port 5036
> Listening on port 5036
> Initialized phone core
> No audio devices found
> bash-2.05b#
>
> and then sometimes I start it and get an apparently successful
> startup
> (window pops up), but some complaints on the command line: bash-
> 2.05b# gnophone New input space: 0 of 40 64 byte fragments (0 bytes
> left) New output space: 40 of 40 64 byte fragments (2560 bytes left)
> Registering Unknown Audio device on /dev/dsp Loaded and activated
> '/usr/lib/gnophone/modules/audio-oss.so' Loaded and activated
> '/usr/lib/gnophone/modules/audio-phone.so' iax.c line 654 in
> iax_init: Started on port 5036 Listening on port 5036 Initialized
> phone core New input space: 0 of 40 64 byte fragments (0 bytes left)
> New output space: 40 of 40 64 byte fragments (2560 bytes left) No
> bytes to read Error reading voice data on Unknown Audio device on /dev/dsp
> Running GUI
>
> I'm still fiddling with my setup here and am hopeful about getting
> all things sound-related to work. However, if this doesn't work,
> then I'm inclined to think that the snd-intel8x0 driver does not
> fully support my hardware:
>
> 00:06.0 Multimedia audio controller: nVidia Corporation nForce2 AC97
> Audio Controler (MCP) (rev a1)
>
> Does anyone here have this hardware working with asterisk through
> ALSA-emulated OSS or plain OSS? If so, what driver are you using?
> snd-intel8x0? Any particular settings I need to tweak to fix the
> hiccupy Voice?
>
> John, what sort of hardware are you using with your snd-intel8x0 driver?
>
> In any case, at least my lsmod output looks better:
>
> bash-2.05b$ lsmod|grep snd
> snd-pcm-oss 39140 0
> snd-mixer-oss 13392 0 [snd-pcm-oss]
> snd-intel8x0 20296 0 (autoclean)
> snd-ac97-codec 48428 0 (autoclean) [snd-intel8x0]
> snd-mpu401-uart 3904 0 (autoclean) [snd-intel8x0]
> snd-rawmidi 14688 0 (autoclean) [snd-mpu401-uart]
> snd-pcm 65828 0 (autoclean) [snd-pcm-oss snd-intel8x0]
> gameport 1692 0 (autoclean) [snd-intel8x0]
> snd-page-alloc 6452 0 (autoclean) [snd-intel8x0 snd-pcm]
> snd-seq-oss 27456 0 (unused)
> snd-seq-midi-event 3840 0 [snd-seq-oss]
> snd-seq 40528 2 [snd-seq-oss snd-seq-midi-event]
> snd-timer 15556 0 [snd-pcm snd-seq]
> snd-seq-device 4176 0 [snd-rawmidi snd-seq-oss snd-seq]
> snd 33892 0 [snd-pcm-oss snd-mixer-oss snd-
> intel8x0 snd-ac97-codec snd-mpu401-uart snd-rawmidi snd-pcm snd-seq-
> oss snd-seq-midi-event snd-seq snd-timer snd-seq-device] soundcore
> 4196 6 [snd] bash-2.05b$
>
> >
> > If OSS is working under xmms, it looks to me like your kernel has OSS
> > support built in. You need to disable this, otherwise ALSA will get
> > terribly confused and won't work.
>
> If it is, then it's only as a module (which I don't think is loaded),
> not in the kernel executable itself. From my .config file I have:
>
> CONFIG_SOUND=m
> CONFIG_SOUND_ICH=m
> CONFIG_SOUND_OSS=m
>
> As I said above, I think that nvaudio driver module is an OSS
> driver. That's probably why xmms worked. Now that it's unloaded, I
> think my kernel is free of native OSS code.
>
> >
> > You can use either ALSA or OSS, not both. If you use ALSA you can
> > then put an OSS compatibility layer on top of it. But get ALSA
> > working first, then worry about the OSS layer.
>
> That makes sense. I seem to have both ALSA and the OSS layer
> working in some respects now, but not all.
>
> >
> > Check your dmesg output for ALSA failing to load due to this.
>
> Nothing there.
>
> >
> > Alternatively, get rid of ALSA entirely and just keep OSS (although
> > this isn't recommended - ALSA is much nicer).
>
> I agree. I'll keep working on ALSA unless I learn that the intel8x0
> driver doesn't fully support my hardware.
>
> Thanks again to you both for your help and patience.
>
> -Kevin
>
> --__--__--
>
> Message: 12
> Date: Fri, 19 Mar 2004 12:54:13 -0500
> From: Brian Capouch <brianc at palaver.net>
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] Important: The Asterisk Mailing list
> (new subject) Reply-To: asterisk-users at lists.digium.com
>
> Tilghman Lesher wrote:
> > On Friday 19 March 2004 02:16, Brian Capouch wrote:
> >
> >>Olle E. Johansson wrote:
> >>
> >>> Do *not* send out personal replies on the list.
> >>
> >>Yes! Yes!! Yes!!!
> >>
> >>Let's change the way the list software works so people won't get
> >>hammered by replying and rid this list of that pox once and for
> >>all.
> >
> >
> > No, no, no. Then everytime somebody hits "Reply All", the poster gets
> > two messages: one from the poster, one from the listserv. And
> > subsequent reply-all's add to this problem. Let's not make the
> > situation worse.
> >
>
> Big deal. One extra email is generated if the user is sloppy or forgetful.
>
> In the present case SEVEN THOUSAND NINE HUNDRED NINETY NINE needless
> emails are sent.
>
> It is incredible to me that people think this current behavior is
> superior--so often there are embarassing gaffes, and always needless
> traffic to the list. . .
>
> B.
>
> --__--__--
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
> End of Asterisk-Users Digest
More information about the asterisk-users
mailing list