Re: [Asterisk-Users] SoftFAX/spandsp cvs access
That would be great. Steve, please also consider using sourceforge.net to host the project. Alex. - Original Message - From: Jeb Campbell [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, April 01, 2004 12:01 PM Subject: [Asterisk-Users] SoftFAX/spandsp cvs access Steve, first thanks for the great work (especially the bugfixes). As development on SoftFAX/spandsp is especially fast and from the source it appears that you are using version control, it would be very nice for us users and testers to have read access to a repository. My cvs/subversion is internal and I understand not opening up systems, but maybe digium would host this if you cannot as this is a HUGE feature for asterisk. Thanks, Jeb Campbell [EMAIL PROTECTED] Cell: 865-385-1437 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MOH: Copyright issues?
AFAIK, in US the copyright expires 25 years after the original copyright holder (author, recording artist, but not sure about an assignee) dies, or after ~70 years from the date of creation (in cases where a corporation holds a copyright for sure), but do not hold your breath, as the companies like Disney constantly lobby to extend this period, otherwise you would certainly see Mickey Mouse cartoons in public domain by now. As far as royalties are concerned, I suppose MOH in US for some company could be considered on par with a bar, which translates to pennies per played song, as long as no more than ~100 people are listening to it at once. But please do not take this a as sound law advice, as I am no lawyer ;-). Cheers! Alex. - Original Message - From: Brian Capouch [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, March 19, 2004 3:14 AM Subject: [Asterisk-Users] MOH: Copyright issues? After reading a (hopefully) joke web news article today that said the RIAA was thinking about asking automobile owners to pay extra royalties when there's more than one passenger in the car, I began to worry about putting the classic 1974 Pointer Sisters' tune, Little Pony in my mohmp3 directory. I know I can always explicitly search out royalty free music, but I wonder if my 50+ year old recordings of The Sons of the Pioneers, or the CD of Clara Rockmore playing a Theremin I bought at the Exploratorium, would wind up with me in the slammer (or the poorhouse!!) if I put them on my system? Does anyone know a way of knowing where a given recording of a song stands? Thx. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] g729 suggestions?
Rich, It is rather hard to tell how many VA g729 licenses you might require at any given time, it all depends on the exact circumstances. Not that you did not describe yours well, but you left out some of the specifics. In general, VA g729 codec licenses are counted like so: 1. A license is required only when the codec functionality is *actually* in use, i.e. when asterisk needs to transcode g729 into anything else, or the other way around; 2. During a transcoded g729 call both directions (encoding and decoding) count as 1 license (but there could also be bugs crawling around in VA code regarding this matter); So, to answer your questions directly: 1. You need 2 licenses to use both C7960s at the same time, assuming they are not calling each other at that moment; 2. Accessing Voicemail counts as any other regular call; 3. A license is certainly needed for an Internet C7960 g729 - x100p pstn call? 4. A license is required for any other internal asterisk function (C7960 to IVR, C7960 to MOH, Voicemail, etc)? Also: 1. If you could record all your IVR prompts in g729, you would not need a VA license to play them to a g729 phone. 2. If you record Voicemails in g729 (not exactly possible with * at this time) and all your phones used g729, you would not need a license; but you would need a license to access the Voicemail from PSTN; recoding voicemails in g729 + other formats would need a license because asterisk would need to transcode incoming g729 stream; 3. If your g729 C7960 calls someone, and the called party places the callee on hold with MOH, a total of *2* licenses will be necessary -- 1 for the call and 1 for MOH; I hope that help! Alex. - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: Asterisk-a-users-list [EMAIL PROTECTED] Sent: Friday, March 19, 2004 11:11 AM Subject: [Asterisk-Users] g729 suggestions? 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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SoftFAX/spndsp
Steve, Many thanks for your work, first of all. We have also been testing spandsp lib with rxfax * app here via RTP/ULAW (on LAN, direct from Sipura SPA2000 and GS HT-286 to *). It seems to be getting better with each new release of spandsp ;-). We do not get any more fast carrier training problems with various PC fax modems and a couple HP all-in-one fax machines, but the received faxes (tiff files) still come out garbled. It works much better with fax modems, faxing from win2000 using its 'native' fax driver. The line errors usually begin somewhere around scanline 1700 (in fine resolution; 200 dpi ?) and after that everything is garbled. With HP fax machines, the line errors start immediately and everything is garbled. It also takes an awful long time to transmit a single page (retransmits?). The only thing that is readable in faxes received from HP machines is the header with date, TSI name/number and page # (is that generated by spandsp or transmitted in the image by the fax?). In both of above cases, the negotiated protocol is V.29 at 9600bps. I can send the logs if you want, they tend to get too big for inclusion in this email with all those line errors. I can also send the received .tiffs. Now, the question I had was if it is practically possible to fax over RTP/ULAW with asterisk. Do you think some changes to asterisk RTP stack might be necessary to accomplish this? It seems very promising as it is right now. Transmitting a fax over POTS with the same fax modems into asterisk with an X100P board works like a charm! But we could not try the same with the fax machines -- don't have a spare line right now. Cheers! Alex. - Original Message - From: Steve Underwood [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, March 19, 2004 2:02 PM Subject: [Asterisk-Users] SoftFAX/spndsp Hi, I have investigated some more fax machines that did not work with spandsp, and made it more tolerant. ftp://ftp.opencall.org/pub/spandsp/spandsp-0.0.1c.tar.gz is the result. From what I have seen in today's investigations, I think this one will work with considerably more quirky fax machines and bad phone lines. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codec negotation with re-invites..
Sounds to me that your asterisk first negotiates g729 with your phone, then negotiates ulaw with the gateway (since it *is* the preferred codec in your config), and on a re-invite the logic breaks up either in the phone or in the gateway (or perhaps in the asterisk itself, I am not absolutely clear on the details of re-invites). Try changing the order of codec preference for the gateway and see if that fixes your g729 phone and breaks the ulaw phone at the same time. Alex. - Original Message - From: Billy Huddleston [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, March 12, 2004 2:02 PM Subject: [Asterisk-Users] Codec negotation with re-invites.. I'm about over this.. okay,, here is what I got.. [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = inbound ; Default for incoming calls tos=lowdelay tos=184 disallow=all; Disallow all codecs allow=ulaw [gateway] type=friend host=1.1.6.9 canreinvite=yes qualify=yes dtmfmode=rfc2833 context=default disallow=all allow=ulaw allow=g729 [sipphoneg729] type=friend secret=password nat=yes host=dynamic canreinvite=yes qualify=200 context=longdistance-g729 dtmfmode=rfc2833 mailbox=2199 disallow=all allow=g729 [sipphoneulaw] type=friend secret=password nat=yes host=dynamic canreinvite=yes qualify=200 context=longdistance dtmfmode=rfc2833 mailbox=2199 disallow=all allow=ulaw okay, when I place a call from sipphoneulaw to the outside world via gateway, everything works fine.. If I place a call from sipphoneg729, it doesn't work.. One leg to the gateway will be ulaw, the leg to the phone will be g729, and, I have 1 way audio.. The sip phone can hear anything from the gateway, but, the gateway can't hear the phone. I've even went as far as to setup a seperate context for the g729 phone and do this.. ,SetVar,SIP_CODEC=g729 which, says it sets it to g729, but it's still a ulaw call.. Guys, this is a real problem... We're going be doing mixed configs.. and if a gateway says it can do both, and phone says it can only do one... then we should be using the compatable codec... PLEASE help.. This is going to cause problems in our rollout. Thanks, Billy +--+ | Billy Huddleston Senior Systems Administrator | | Net-Express http://www.nxs.net | | 114 Sherway Rd. Voice: 865-691-2011 | | Knoxville, TN 37922 Fax: 865-691-9894 | | [EMAIL PROTECTED]| +--+ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Codecs [G.729]
Have a look at the following document. I know it is not *exactly* what you are asking, but it gives you an idea on how the actual bandwidth consumption changes depending on the actual network medium you are utilizing. http://www.cisco.com/warp/public/788/pkt-voice-general/bwidth_consume.html Have fun. Alex. - Original Message - From: Unavailable ID To: [EMAIL PROTECTED] Sent: Monday, March 08, 2004 7:29 PM Subject: [Asterisk-Users] Asterisk Codecs [G.729] Hello all, I'm looking for advice for codec that works best for asterisk. Anyone has real testing with all codecs, specially with G.729. I have tested with single callon few codecs that come with asterisk by using IPTraf and the rate as of below: ulaw64 Kbps, sample-based Also known as alaw/ulaw166kbits/secalaw64 Kbps, sample-based Also known as alaw/ulaw167kbits/secgsm13 Kbps (full rate), 20ms frame size66kbits/secspeex2.15 to 44.2 Kbpsn/aiLBC15Kbps,20ms frame size: 13.3 Kbps, 30ms frame size57.6kbits/secG.7298 Kbps, 10ms frame sizelicense Have anyone test it with G.729? Please let me know. Thanks.
Re: [Asterisk-Users] exit
You must have started asterisk with asterisk -c so you cannot bail out of CLI with exit -- you are in console mode. Instead, start it without -c so it respawns another service process and exits to shell, after that you can run asterisk -r and bail out with exit all you please ;-). - Original Message - From: Greg Kedrovsky [EMAIL PROTECTED] To: asterisk-user [EMAIL PROTECTED] Sent: Thursday, February 26, 2004 10:41 PM Subject: [Asterisk-Users] exit Talk about a stoopid question... How do I exit the CLI of Asterisk. Typing exit (per the pdf manual and my google results) brings up a message saying QUIT and EXIT are no longer available, that STOP NOW is used to shutdow the pbx. I do not want to shutdown the pbx. I just wanna get outta the CLI and back to my Linux command line. Gosh... I feel like a 1st grader that can't get my fly open to take a pee. -Greg -- Mutt 1.4.1i on Slackware 9.1 Linux Curridabat, San Jose, Costa Rica http://www.greg-and-sue.com/screenshot.jpg Yahoo Instant Messenger ID: gregkedro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
