Jason Konik
2004-Mar-19 11:17 UTC
[Asterisk-Users] Re: I would like to UNsubscribe from this list thanks
On Fri, 19 Mar 2004 13:06:16 -0600, asterisk-users-request wrote> Send Asterisk-Users mailing list submissions to > asterisk-users@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@lists.digium.com > > You can reach the person managing the list at > asterisk-users-admin@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@linuxautrement.com> > To: Asterisk List <asterisk-users@lists.digium.com> > Date: Fri, 19 Mar 2004 17:47:55 +0100 > Reply-To: asterisk-users@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@linuxautrement.com> > > --__--__-- > > Message: 2 > Date: Fri, 19 Mar 2004 09:21:01 -0800 (PST) > From: Kurt Pasewaldt <kurtwp@yahoo.com> > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Asterisk Voice Mail Integration with Cisco > CME Reply-To: asterisk-users@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@charter.net> > To: asterisk-users@lists.digium.com > Subject: RE: [Asterisk-Users] firefly softphone > Reply-To: asterisk-users@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@omniis.com> > To: <asterisk-users@lists.digium.com> > Subject: [Asterisk-Users] firefly softphone > Reply-To: asterisk-users@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@mail.jeffandtilghman.com> > To: asterisk-users@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@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@loon.org.uk> > To: <asterisk-users@lists.digium.com> > Subject: RE: [Asterisk-Users] Speaking of ring tones... > Date: Fri, 19 Mar 2004 17:28:21 -0000 > Reply-To: asterisk-users@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@bkglobal.com> > To: asterisk-users@lists.digium.com > Date: Fri, 19 Mar 2004 11:34:37 -0600 > Subject: [Asterisk-Users] DID with X100P? > Reply-To: asterisk-users@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@puwaba.com> > To: <asterisk-users@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@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@yahoo.com> > Subject: RE: [Asterisk-Users] Can i do voice chat without using > the hardware To: asterisk-users@lists.digium.com Reply-To: asterisk- > users@lists.digium.com > > --- David J Carter <david.carter@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@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@yahoo.com > Cell: 310-990-7550 > Office: 310-336-5189 Christopher.J.Albertson@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@routers.com> > To: Asterisk-a-users-list <asterisk-users@lists.digium.com> > Subject: [Asterisk-Users] g729 suggestions? > Reply-To: asterisk-users@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 ofdefinitive> 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@tiger.com.pl> > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] Identifying a call with manager interface > Reply-To: asterisk-users@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@gnosys.biz> > To: asterisk-users@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@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@palaver.net> > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] Important: The Asterisk Mailing list > (new subject) Reply-To: asterisk-users@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@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > > End of Asterisk-Users Digest