Re: [asterisk-users] WARNING: chan_zap.c process_zap: Ignoring signalling
It actually CAN but because someone was lazy and didn't want to actually do the work to make it possible to do a full change during a reload. The biggest issue is ztcfg would have to be absorbed into chan_zap to make it 100% possible. In fact if Digium wanted to make Asterisk easier to configure/setup they would merge ztcfg into chan_zap and get rid of /etc/zaptel.conf and save a config step. /b On Oct 21, 2007, at 10:23 AM, Tzafrir Cohen wrote: On Sun, Oct 21, 2007 at 04:27:17PM +0200, Vincent wrote: ubuntu*CLI reload chan_zap.so -- Reloading module 'chan_zap.so' (Zapata Telephony) == Parsing '/etc/asterisk/zapata.conf': Found [Oct 21 16:22:37] WARNING[8240]: chan_zap.c:11120 process_zap: Ignoring signalling chan_zap cannot change signalling of a channel on reload. So that parameter is ignored on reload. False warning... -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] WARNING: chan_zap.c process_zap: Ignoring signalling
Thats a great step forward. Auto for PRI doesn't make sense... but two configs to describe the same thing makes no sense. /b On Oct 21, 2007, at 1:03 PM, Tzafrir Cohen wrote: On Sun, Oct 21, 2007 at 11:57:45AM -0500, Brian West wrote: It actually CAN but because someone was lazy and didn't want to actually do the work to make it possible to do a full change during a reload. The biggest issue is ztcfg would have to be absorbed into chan_zap to make it 100% possible. In fact if Digium wanted to make Asterisk easier to configure/setup they would merge ztcfg into chan_zap and get rid of /etc/zaptel.conf and save a config step. Look at the auto signalling in http://svn.digium.com/svn/asterisk/team/group/zapata_conf . This is rather nice for analog channels. Much less of a help, I'm afraid, for PRI. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Issues with making calls
Make sure chan_zap.so is loaded. /b On Oct 18, 2007, at 9:34 AM, Pablo Almido wrote: Hi List, I am from Peru, I have installed an asterisk server in my company with digium card E1 TE120P, I am having issues when i make calls, here the error from my server [Oct 18 09:13:50] WARNING[2377]: channel.c:3232 ast_request: No channel type registered for 'Zap' [Oct 18 09:13:50] WARNING[2377]: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'Zap' (cause 66 - Channel not implemented) == Everyone is busy/congested at this time (1:0/0/1) My sources are: libpri-1.4.1.tar.gz zaptel-1.4.5.1.tar.gz asterisk-1.4.11.tar.gz asterisk-addons-1.4.2.tar.gz asterisk-perl-0.10.tar.gz I have 1/2 E1 from my provider telephony, my configuration is [EMAIL PROTECTED] ~]# cat /etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 loadzone = us defaultzone=us #cat /etc/asterisk/zapata.conf [channels] context=default switchtype = euroisdn pridialplan = unknown signalling = pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 immediate=no amaflags=documentation musiconhold=default ;Configure Channels group=0 callgroup=0 pickupgroup=0 channel = 1-4 group=1 callgroup=1 pickupgroup=1 channel = 5-8 group=2 callgroup=2 pickupgroup=2 channel = 9-12 group=3 callgroup=3 pickupgroup=3 channel = 13-14 group=4 callgroup=4 pickupgroup=4 channel = 15 I have could make calls but, after of some minutes my server is hung, suggestions are welcome. Thanks for any help in advance. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Issues with making calls
Why would a config error stop the module from loading? That seems like a suboptimal behavior. /b On Oct 18, 2007, at 9:50 AM, Jared Smith wrote: That would seem to indicate that the chan_zap.so module isn't being loaded. What happens if you type module unload chan_zap.so and then module load chan_zap.so from the Asterisk CLI? I'll bet you'll find that either there's a problem in your zapata.conf file, or that chan_zap hasn't been compiled for some reason. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] centos 5 vs OpenSuse 10.3
I'm sorry I call bullshit on this one. CentOS has been 2.6 for some time. /b On Oct 18, 2007, at 11:22 AM, [EMAIL PROTECTED] wrote: Just 5 months ago CENTOS started to use Linux 2.6 one of the reasons I'd abandoned for SuSE a while back. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] segfault
You'll need to compile with debug symbols and have ulimited -c unlimited set. Then you can examine the core and find out what exactly caused the crash... Segfaults either are easy to find or very hard to find, depending on what is happening. It could also be bad ram. /b On Oct 16, 2007, at 9:13 PM, Rilawich Ango wrote: HI all, I got segfault in the system log that make asterisk crash. I still have no idea what cause this segfault. Is it a bug? Anyone has experience about it? phsip01 kernel: asterisk[3412]: segfault at 2aabd10f2b40 rip 0037e806ea75 rsp 41d3cc70 error 6 version: asterisk1.4.12.1 usage: in/out bound call, queue, ivr, attended call transfer ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Loud pop at the end of messages causing level problems
You should really never touch those. If you're having problems with the card call support because that is far from normal. /b On Oct 16, 2007, at 9:35 PM, Stephen Bosch wrote: Eric Deutsch wrote: Hi everyone, I’ve set up a little Asterisk system with a Digium TDM400P and everything works splendidly except for the messages callers leave. Every message that a caller leaves is very faint. I’ve already set volgain=6.0 in voicemail.conf, and that seems better, but to be at a good volume I estimate I may need to go up to 40.0. Is that reasonable? Before you tinker with the gain settings in voicemail.conf, I recommend you tweak the gain settings in zaptel.conf. -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mask Initial Processing with Ring Back Tone
Just dont answer it till the processing is done. No debate is needed for this. I do this millions of times per month. /b On Oct 11, 2007, at 2:56 PM, Eric \ManxPower\ Wieling [EMAIL PROTECTED] wrote: Victor wrote: I need to process a number of lines of code in the dialplan before answering a call. Can standard ring back tones be played to the caller while this is happening prior to answering the call. Which commands would facilitate this? I strongly doubt those lines are going to take up much time. You can use Playtones to play specific inband tones. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call recording on demand...
Look at features.conf /b On Oct 10, 2007, at 10:48 AM, Reggie Payne wrote: Hello All! I am new to the list. Does know how to record a call on demand? What I would like to do is setup something that during a call someone can hit a button a the call is recorded the after the call is over the recording is sent to their voicemail. Anyone? Thanks, Reg ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Click to Talk Web Applications with Asterisk
On Oct 10, 2007, at 11:12 AM, Ex Vito wrote: On 10/9/07, Senad Jordanovic [EMAIL PROTECTED] wrote: zoachien wrote: Google for mexuar. Zoa Or look at one that works with MS Windows, Linux or Apple http://www.bicomsystems.com/products/C/P/319/382/ FYI, Mexuar's solution -- Corraleta SDK -- *works* with win, linux and mac, from direct experience. What's not so clear from the OP is what is meant by click-to-call: a) Automated dialing solutions via PSTN ? b) Call via a web embedded soft-phone ? (this would be Mexuar) -- exvito I think what he wants is something that does third party call control (3pcc). WeSIP is one but you can't use it in a commercial application without paying for a license. FreeSWITCH can be controlled with 3pcc also and its free. That is what most if not all Click-to-Dial applications use. RFC3725 covers this. http://en.wikipedia.org/wiki/3pcc for more information. /b ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to order audio codecs...
if you have allow=g729,ulaw and you want to use g729 but the current channel is ulaw it will pick ulaw over g729 because it wants to escape doing any transcoding if possible. The best way to do this is setup different peers with different allow lines to force the outbound leg to the codec you wish. /b On Oct 10, 2007, at 11:02 AM, Marc LEURENT wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I have license for g729a audio codecs and I would like user to use them and when the limit of 10 is reached, I would like the others to use ulaw... Do youu know how to do it... I have put: allow=g729,ulaw disallow=all But ulaw is always chosen Have a nice day -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHDPePqjpLE0HiOBYRAvSWAJ9Z7gJMDuTw9EcL5of35SmF1slwIwCeM8n/ MfjqNU/3gkdLwKqo1tN5yV8= =3oU/ -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T-Mobile and WiFi Voip
its IMS /b On Oct 9, 2007, at 10:39 AM, Andres wrote: I had a friend yesterday showing me his new T-mobile blackberry with WiFi Voip.I could not believe it until I actually saw him making calls. There is no T-Mobile cell coverage at my house but he was able to simply access the WiFi router and make the call. It appears this VoIP offering is tightly integrated since you use the same phone number to make and receive calls over WiFi or Cell. Does anybody know if its SIP? I wanted to get some packet captures but he was in a hurry. -- Andres Technical Support http://www.telesip.net ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DS3 Interface
http://www.imagestream.com/PCI_921-CDS.html This card can do it. I have spoke with them about it and its very capable of doing what is needed for a DS3 in a standard linux box. /b On Oct 9, 2007, at 10:42 AM, Andrew Kohlsmith wrote: On Tuesday 09 October 2007 10:14:23 Matt wrote: Before you put any work into this... ask yourself... what exactly are you hoping to accomplish? There is no way one system can handle a DS3s worth of traffic... therefore, what good would this do? Whatever gave you the notion that a modern PC can't handle 672 simultaneous calls? -A. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DS3 Interface
I'm already doing that. /b On Oct 9, 2007, at 11:31 AM, Tim King wrote: I have started the open source project to get this going. I am working directly with the manufacture to form agreements and gain access to the hardware and source code for their drivers. The list price for the card is $4,995.00 USD. I will keep everyone posted and will have site for development and forums up soon. Thanks for the support Tim King CEO 7589 Cottonwood Drive Suite C Jenison, MI 49428 Phone 616.301.3290Fax: 616.667.1104 Website: http://www.compnetwork.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Baji Panchumarti Sent: Tuesday, October 09, 2007 12:07 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DS3 Interface On 10/9/07, Brian West wrote: http://www.imagestream.com/PCI_921-CDS.html [...] off-topic : I saw Imagestream at the Ohio Linuxfest a weekend ago. Also picked up a few literature bags by Digium :-) -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DS3 Interface
You apparently don't realize you're talking to. Thats ok, You keep working on it from your angle. We are evaluating when the time is right to implement this. We aren't doing this for Asterisk we are doing it for FreeSWITCH. /b On Oct 9, 2007, at 12:00 PM, Steve Totaro wrote: Competition is a good thing. Let's say you fail or your implementation is not as robust as the other project or visa versa. Just as long as the hardware vendor is different, it should be a good thing. If it the same hardware vendor, then maybe you two should work together. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DS3 Interface
Well we are plugging it in the OpenZAP abstraction layer we have already started on. This is usable by Asterisk also so asterisk would benefit from it. http://fisheye.freeswitch.org/browse/OpenZAP /b On Oct 9, 2007, at 12:31 PM, Steve Totaro wrote: BTW, this is the wrong list if it not for Asterisk. It has absolutely nothing to do with Asterisk. Please post to the appropriate FreeSwitch list. Thanks again, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DS3 Interface
And what was the purpose of this? /b On Oct 9, 2007, at 1:32 PM, Matt wrote: http://www.usdoj.gov/criminal/cybercrime/WestPlea.htm On 10/9/07, Brian West [EMAIL PROTECTED] wrote: You apparently don't realize you're talking to. Thats ok, You keep working on it from your angle. We are evaluating when the time is right to implement this. We aren't doing this for Asterisk we are doing it for FreeSWITCH. /b On Oct 9, 2007, at 12:00 PM, Steve Totaro wrote: Competition is a good thing. Let's say you fail or your implementation is not as robust as the other project or visa versa. Just as long as the hardware vendor is different, it should be a good thing. If it the same hardware vendor, then maybe you two should work together. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DS3 Interface
Matt, I talk very openly about this issue. It was very rude of you to bring this up. This plea was total bullshit. If you want to know the whole story feel free to call me and talk about it. 918-424-9378... anyone can call me and ask me questions about it. The plea was a deal worked out between the DOJ and my attorney which was good because I signed my plea on Sept. 4th 2001. If you try to fight the DOJ you will not win. That plea was the only way to settle the issue without a trial. All I did was click edit in frontpage and alert them of anonymous publishing priv. were on their servers and they called the FBI and three days later our office was raided. This I consider mudslinging by you and wasn't very gentle man like. /b On Oct 9, 2007, at 1:32 PM, Matt wrote: http://www.usdoj.gov/criminal/cybercrime/WestPlea.htm On 10/9/07, Brian West [EMAIL PROTECTED] wrote: You apparently don't realize you're talking to. Thats ok, You keep working on it from your angle. We are evaluating when the time is right to implement this. We aren't doing this for Asterisk we are doing it for FreeSWITCH. /b On Oct 9, 2007, at 12:00 PM, Steve Totaro wrote: Competition is a good thing. Let's say you fail or your implementation is not as robust as the other project or visa versa. Just as long as the hardware vendor is different, it should be a good thing. If it the same hardware vendor, then maybe you two should work together. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DS3 Interface
I'm number three on the dev team and not the soul person behind FreeSWITCH. Its very uncalled for. You are dragging our project thru the mud now also. Don't pass judgement on me. You sound quite childish and waste my time. Never judge a man till you walk a day in his shoes. /b On Oct 9, 2007, at 2:12 PM, Matt wrote: Perhaps it was uncalled for. However, if I were to consider using FreeSwitch I would want to know who was/is behind it. On 10/9/07, Brian West [EMAIL PROTECTED] wrote: And what was the purpose of this? /b ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DS3 Interface
Well hopefully people can read between the lines.. I have talked about this issue in public many times and don't try to hide it but the plea isn't how it went down. /b On Oct 9, 2007, at 1:50 PM, Steve Totaro wrote: Yes, I knew who I was talking to and now I know a little more about you Matt, that was totally uncalled for. Thanks, Steve Totaro Matt wrote: http://www.usdoj.gov/criminal/cybercrime/WestPlea.htm On 10/9/07, *Brian West* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: You apparently don't realize you're talking to. Thats ok, You keep working on it from your angle. We are evaluating when the time is right to implement this. We aren't doing this for Asterisk we are doing it for FreeSWITCH. /b On Oct 9, 2007, at 12:00 PM, Steve Totaro wrote: Competition is a good thing. Let's say you fail or your implementation is not as robust as the other project or visa versa. Just as long as the hardware vendor is different, it should be a good thing. If it the same hardware vendor, then maybe you two should work together. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api- digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - --- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DS3 Interface
I would recommend doing it on a 64bit platform for sure. Not sure Asterisk has very many linger issues on 64bit... I know I run it on 64bit without too much drama. /b On Oct 9, 2007, at 9:32 PM, Mr. James W. Laferriere wrote: Please , step back form the keyboard , take a deep breath . then maybe we can get on with the discussion of creating a driver under aterisk for a ds3 card . ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] libdundi?
Now the next question is why do no LGPL Dundi libs exist? /b ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Curiosity Max Calls
They why was it on the website? /b On Oct 8, 2007, at 11:59 AM, Tilghman Lesher wrote: On Sunday 07 October 2007 15:23, Steve Totaro wrote: How about the once announced Digium DS3 card (that I never saw come to market), that board must have some powerful onboard circuits or require a very powerful server SGI Numalink setup. I guess with dual procs and quad core systems, maybe thats not an issue anymore. No such board was ever announced. There were rumors of such a board, but nothing ever got past rumors. -- Tilghman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Curiosity Max Calls
The board never came to market 1. because the demand. 2. impossible to do with zaptel. /b On Oct 7, 2007, at 3:23 PM, Steve Totaro wrote: How about the once announced Digium DS3 card (that I never saw come to market), that board must have some powerful onboard circuits or require a very powerful server SGI Numalink setup. I guess with dual procs and quad core systems, maybe thats not an issue anymore. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good Book to learn SIP
Telling someone to read the RFC bah.. might as well give them a blanket and pillow because they will fall asleep. chan_sip is just ugly in every way. /b On Oct 7, 2007, at 9:26 PM, Baji Panchumarti wrote: http://www.faqs.org/rfcs/rfc3261.html as well as the source in asterisk (1.4.11 here) asterisk-1.4.11/channels/chan_sip.c ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G.722: ast_channel_make_compatible failure
I would like to point out that G.722 is a really awesome codec for wideband. Asterisk has some changes that will need to be made to support variable audio rates. We did this in FreeSWITCH from the start. I think Asterisk will be doing similar things to bridge an 8k to 16k channel via resample. FreeSWITCH can already do this so you could use FreeSWITCH in conjunction with Asterisk to solve this for now. FreeSWITCH can also do Wideband conferencing. In addition you can mix and match 8k and 16k conference participants. Just thought I would throw that out there as a way to bridge the gap. /b On Oct 5, 2007, at 1:13 AM, Ondrej Valousek wrote: Hi Kevin, Thanks for the answer - Hopefully this feature will be available some day. My opinion is, look for a transcoder only if the two (or more) parties does not offer any matching codec. Good to hear it is being worked on Best regards, Ondrej Kevin P. Fleming wrote: Ondrej Valousek wrote: My problem is, that the phone offering g722 could do alaw as well. I expected asterisk should just chose alaw for the communication - no transcoding is necessary then... That is not how Asterisk works, and is well known in the community as something that users would like to see changed, but has not yet been done. Asterisk negotiates the codecs (formats) for each call leg pretty much independently of the others, so if a G.722 endpoint initiates the first call leg, and the destination call leg cannot accept G.722, and there is no transcoder available, then the call will fail. If the non-G.722 endpoint initiates the first call leg then the call will likely go through, which is somewhat unfortunate :-) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G.722: ast_channel_make_compatible failure
You can hear and understand someone much better with g722... more emotion is transfered over the phone when using g722. G722 is free and in the clear. G722.1 and G722.2 are not. We have the G722 code in FreeSWITCH donated to us by Steve Underwood. What a great guy. /b On Oct 5, 2007, at 8:05 AM, Ondrej Valousek wrote: At this point I would like to know why you think it is awesome? I know the are some extensions/improvements to this codec but these are unfortunately not free so no use for asterisk. Thanks, Ondrej ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma vs Digium: (was Re: ping too)
Kevin, Thats good to know. I'll keep that in mind. Thanks, Brian PS: did you ever talk to mark about zaptel.h ? On Oct 5, 2007, at 8:12 AM, Kevin P. Fleming wrote: Those drivers would be there (as are the Xorcom XPP drivers) if they were properly submitted and met our coding guidelines. To date I have no knowledge of Sangoma ever submitting a driver for inclusion in Zaptel, and the last time anyone talked about a driver submission from Rhino the code was not in a state that met our minimum requirements for acceptance. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G.722: ast_channel_make_compatible failure
But its way too heavy on the CPU. /b On Oct 5, 2007, at 8:34 AM, Tzafrir Cohen wrote: But speex *Is* free. Including wideband. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma vs Digium: (was Re: ping too)
Sangoma has contributed to Asterisk in the past and they still do. They also have contributed to Yate, FreeSWITCH and various other software that is capable of using their hardware. This argument of Digium vs Sangoma is very emotional for some. I see it as competition is good and drives innovation. Digium can't take every bit of credit for Asterisk, you have to remember the community has a large part in making Asterisk as popular as it is. I know their is hostility directed at anyone that uses non-Digium hardware by some folks and their shouldn't be. Its an open market and an open platform. Rhino makes hardware that plugs into zaptel but yet I don't see their drivers in the zaptel repo... I don't see many of the third party hardware drivers in the zaptel repo. /b On Oct 5, 2007, at 7:51 AM, Steve Murphy wrote: Oh, Julian, I'd imagine what I'm about to say will fuel some flames! Here's a fairly powerful argument for all you asterisk users, as to why you should purchase a Digium product vs. a Sangoma: Because Digium uses a chunk of the purchase money to support Asterisk. And Sangoma DOES NOT. Digium employs several developers specifically to maintain and improve Asterisk. Sangoma DOES NOT. While they may maintain and improve their own versions of the various drivers, THEY DO NOT SHARE THEIR SOFTWARE. Matt F. told me last week we haven't seen ANYTHING from them for a LONG TIME, with respect to the zaptel drivers. If they have been contributing patches, they are disguising their association with Sangoma. Don't get me wrong. I AM a Digium employee! A software Developer to be specific, an Asterisk developer to be precise. So, I AM highly biased towards Digium! Digium has a harder job than Sangoma with respect to Asterisk. While Digium takes a chunk of its revenue, and uses it to maintain and improve Asterisk (not just the drivers), Sangoma doesn't, and it gives them a competitive edge. So, for all you folks who have bought Digium, I personally thank you! You have helped Asterisk, and you have personally helped ME. If you have long-range business or interest in Asterisk, you are indirectly contributing to its growth and improvement when you buy Digium products services. murf -- Steve Murphy Software Developer Digium ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G.722: ast_channel_make_compatible failure
On Oct 5, 2007, at 9:31 AM, Tzafrir Cohen wrote: How many hardware vendors support g722.1 ? g722.2 ? How pleasent are they to the CPU? How much does it cost them? I think polycom does and both are very heavy on CPU. Naturally I don't suggest to use speex/wb where there is enough bandwidth for g722 . But right now AFAIK g722 and speex are just about the only two free alternaitves. Non-free means loads of licensing issues (I swapped a network adapter, why can't I use g729 anymore?) You can use wideband speex with the googletalk client on FreeSWITCH to call into a conference also running on FreeSWITCH and conference in wideband. Its very heavy on CPU for speex you might get 10 channels on a beefy box at most. /b ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma vs Digium: (was Re: ping too)
I think Lee Howard nailed it. /b ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma vs Digium: (was Re: ping too)
The distinction doesn't matter because in the end they can do what ever they want with the code you disclaim to them. The whole thing is very political and pointless to hash over and over again. /b On Oct 5, 2007, at 2:52 PM, Tilghman Lesher wrote: When you contribute code to Asterisk, you retain ownership of your code. You are NOT disclaiming the contribution; you are LICENSING the contribution. This is an important legal distinction, and all too often, it gets muddled by people who either do not understand the distinction or have ulterior motives. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma vs Digium: (was Re: ping too)
I think the horse has been long dead! /b ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] About Megaco
On Oct 4, 2007, at 8:39 AM, Steve Totaro wrote: Try searching using MGCP which is what Megaco evolved into. http://www.voip-info.org/wiki-Asterisk+MGCP+channels Thanks, Steve Totaro Too bad the MGCP channel isn't the full implementation. /b ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] extensions.conf vs. AEL
In my opinion the dialplan isn't where that logic belongs. /b On Oct 3, 2007, at 12:32 AM, Yehavi Bourvine +972-8-9489444 [EMAIL PROTECTED] wrote: Hello, I see that most people are using the extensions.conf syntax (most of the examples and questions here use that syntax). recently I've translated all my dial plan to AEL syntax and I find it much easier, especially when you need IFs. Why most people don't use it? Am I missing something? Thanks! __Yehavi: ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] extensions.conf vs. AEL
You have various scripting languages things like that can go in! /b On Oct 3, 2007, at 4:12 AM, Garth van Sittert wrote: Where would you suggest all the logic goes Brian? Garth Garth van Sittert BSc (Physics Computer Science) - Main: 08600 BITCO Phone: +27 (0)11 875 6900 Fax: +27 (0)11 875 6901 Mobile: +27 (0)83 791 6662 Email: [EMAIL PROTECTED] MSN: [EMAIL PROTECTED] Web:www.bitco.co.za Brian West wrote: In my opinion the dialplan isn't where that logic belongs. /b On Oct 3, 2007, at 12:32 AM, Yehavi Bourvine +972-8-9489444 [EMAIL PROTECTED] wrote: Hello, I see that most people are using the extensions.conf syntax (most of the examples and questions here use that syntax). recently I've translated all my dial plan to AEL syntax and I find it much easier, especially when you need IFs. Why most people don't use it? Am I missing something? Thanks! __Yehavi: ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] extensions.conf vs. AEL
On Oct 3, 2007, at 9:39 AM, Jon Schøpzinsky wrote: Wouldnt that take a very large portion of datapower, to startup the parsers and such, instead of having the whole dialplan natively in Asterisk. We always try to do as much as possible in dialplan, so that we are not reliant on external scripts. Kind Regards Jon Leren Schøpzinsky Stepping thru the dialplan line by line is one of the most inefficient things in Asterisk... Every priority it checks and rechecks the dialplan and priorty at the very least 5 times per priority. I think this is one thing being addressed in 1.4 and later. Dialplan logic isn't a language in my opinion. /b ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] extensions.conf vs. AEL
Its just a different way to express the same thing in a more fluid way. /b On Oct 3, 2007, at 10:33 AM, Anthony Francis wrote: Doesn't this render having used AEL pointless? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] extensions.conf vs. AEL
I'm growing fond of XML. /b On Oct 3, 2007, at 10:39 AM, Steve Totaro wrote: To each his own. I like the flat files personally, they are more fluid to me. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] meetme conference using g729?
You still do not understand. It doesn't matter if the call coming in is g729 you must transcode it to signed linear, mix the frames and then code it back into g729 you end up with quality loss doing that. /b On Oct 2, 2007, at 4:42 PM, Mark Quitoriano wrote: anyway still if there's a hack for meetme to work with g729 codec this won't be an issue. So is there a hack or patch that i can use any codec for meetme? tnx ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] meetme conference using g729?
Thanks for making it clearer :) My mind is mush today! /b On Oct 2, 2007, at 5:39 PM, Tilghman Lesher wrote: Or, in other words, you cannot mix compressed data. You must first decompress the data for mixing, then recompress it for transmission. During both operations, there is a potential for signal degradation. -- Tilghman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7940G licensing with asterisk
Just buy the Linksys SPA962's they work better than the cisco phones in a NAT env. /b On Oct 1, 2007, at 6:13 PM, Andrew Joakimsen wrote: My understanding is: Smartnet: service contract basically allows you to download the newest sw release. Besides that you can buy phones without a license. Presumably as spares But you must buy a SIP license to technically be allowed to use that software that can be obtained from Smartnet. I know there was some changes a year or two back, but wasn't that just pricing? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] meetme conference using g729?
Ok Let me chime in on this one. If you can use ulaw/alaw because you'll end up with tandem encoding which will make the conference sound worse to some people. All audio coming in will get transcoded to signed linear and pushed down into zaptel then back up and out to the conference participants. You'll end up with the best audio quality if you limit the transcoding. /b On Oct 1, 2007, at 6:37 PM, Mark Quitoriano wrote: but is there a way to use g729 codec in meetme? On 10/2/07, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: In my experience, and theoretically by design, it doesn't matter what codec you are using when you call a meetme conference. Moj ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] . (period): Wildcard match; matches one or more characters
On Sep 28, 2007, at 4:52 PM, Mojo with Horan Company, LLC wrote: To use the wildcard characters, 'X', 'N', or '.', I had to also prefix my extension with '_', which enables pattern matching. Don't forget you also have Z which if I recall its 1-9, N is 2-9 and X is 0-9 /b ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoIP Provider for business
Good luck with that one. Most unlimited providers have limits. (even if they say unlimited) /b On Sep 19, 2007, at 12:32 AM, Jim Boykin wrote: Can someone suggests a good and resonable cost voip provider with business unlimited plan in USA and allows simultaneous outgoing calling. Thanks ~Jim ___ Sign up now for AstriCon 2007! September 25-28th. http:// www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] what is softswitch
Their really isn't many differences. A true softswitch will usually never speak to an end users device directly. /b On Sep 19, 2007, at 10:02 AM, satish patel wrote: Dear all what is softswitch what is difference between asterisk and softswitch ?? regards satish patel ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Freeswitch Vs Asterisk
Satish, It depends on your goals. FreeSWITCH is approaching an official release. Beta 1 is out now and various other tweaks in trunk. But its really up to you to evaluate your need and compare which fits your needs. I see them as complementary to each other so its really up to you. /b On Sep 19, 2007, at 10:56 AM, satish patel wrote: Dear all Which one would be best for large production enverment freeswitch or Asterisk and which on would be stable and fuctional ??? Regards Satish Patel Shape Yahoo! in your own image. Join our Network Research Panel today! ___ Sign up now for AstriCon 2007! September 25-28th. http:// www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] what is softswitch
Asterisk isn't a big iron switch. /b On Sep 19, 2007, at 11:08 AM, Tzafrir Cohen wrote: On Wed, Sep 19, 2007 at 11:15:25AM -0400, Alex Balashov wrote: Asterisk is a PBX. A softswitch is more or less a fully featured telephone switch, usually one that is extensively application-driven (more so than traditional big-iron switches) and multiprotocol. Hmmm, Still describes Asterisk. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] what is softswitch
With zaptel that will be impossible, asterisk can do GR303 not sure how well. /b On Sep 19, 2007, at 12:04 PM, Alex Balashov wrote: Perhaps I'll be a little more amicable when someone finds a way to bring at least five or six DS3s into Asterisk. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Died message
It will not after some types of crashes. /b On Sep 5, 2007, at 9:43 AM, Perssy Llamosas wrote: You are using safe_asterisk, it will restart automatically Asterisk after it crashes. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] remove unnecessary text (was: Re: Can asterisk give half-ring periodically for MWI?)
On Sep 5, 2007, at 7:42 PM, Philipp Kempgen wrote: I don't need messages to tell me *5* times about Astricon, who provides the bandwidth and how to unsubscribe. You sure about that unsubscribe part? People do seem to miss it :P /b ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cepstral's Allison is having trouble speaking clearly
Try setting the RTP packets to 0.020 instead of 0.030 which is the default on the SPA's /b On Sep 3, 2007, at 5:00 PM, Todd Reese wrote: Hi all, I have just install and licensed Cepstral's Allison08kHz on my Asterisk 1.4.11 system. I can call the Allison's extension from my Grandstream IP Phone and she's clear as a bell, but when a call to her extension traverses through one of the Linksys/Sipura 3102 or 2002, she's got the jitters bad. The SPA-202 has only an extension phone on it and the SPA-3102 is my FXO from my Vonage Motorola box. Any clues where to start looking to clear this up? TIA, Todd Reese ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FW: Account Registration Failed
Localnet is wrong... try localnet=192.168.1.0/24 /b On Sep 3, 2007, at 9:13 PM, neoh kumyee wrote: Hi, I am trying to run an Asterisk (1.4.11) server on Linux Suse. The server is behind NAT. I am testing with SIP client that developed from PJSIP running on Pocket PC Windows Mobile 5.0 . The client is also behind another NAT. STUN server is implement in SIP client. When a register Message send from client to server, Asterisk receive it and reply with 100 trying msg. However, there is no reply on 200 OK from server, as it course my SIP client registration failure. 1. On the other hand, if i tested with OPENSER SIP server, registration is fine. Important details are below: sip.conf [global] nat=yes canreinvite=no localnet=192.168.1.46 externip=60.xx.xx.xx.xx [8000] type=friend secret=8000 nat=yes host=dynamic canreinvite=no How can i solve it? p/s : Network traffic capture in Ethereal are attached. ~ cobra client.cap - capture at client side ~ cobra server - capture at Asterisk server Thanks Regards kum Live Search: Better results, fast Try it now! Call and stay connected with your friends and family for free. Seen and be heard with high-definition video calls on Windows Live Messenger. Try it! cobra client.cap cobra server.cap ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] where is 1.4.12?
On Aug 30, 2007, at 8:49 AM, Matt wrote: impressions are everything).Digium also makes money off of the FXO/FXS/PRI cards, which you really wouldn't use unless you were running asterisk. So in this case, while Asterisk IS free, it is I have to comment here. If I recall all the zap hardware works with YATE and for sure with FreeSWITCH! /b___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to handle + prefix
On Aug 30, 2007, at 10:11 AM, Jared Smith wrote:
On Thu, 2007-08-30 at 15:42 +0100, Adrian Marsh wrote:
Is there a way of using variables within the dialplan, eg:
[globals]
SOMEVAR=0179344
[local]
exten = _${SOMEVAR}.,1,NoOp(Dialled own number)
No, unfortunately you can't use variables as part of the extension
name
or pattern match.
Since when? I knew you couldn't use them for pattern matches but in
1.2 you could at one point I tested this personally.
/b
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Re: [asterisk-users] Round robin behavior for dialing SIP trunks...
http://www.freeswitch.org/asterisk_stuff/app_distributor.c /b On Aug 30, 2007, at 7:38 PM, Paul Hales wrote: We found the 'random' dialplan function worked quite well for something similar a while ago. PaulH On Thu, 2007-08-30 at 17:38 -0500, Carlos Chavez wrote: I was wondering if anyone has an easy way to emulate dialing in a round robin fashion like when you use Zap/r1 for Zap trunks. At the moment what I do is simply make a macro that will dial the sip trunks in order so if the first one fails it goes to the second and so on. The problem with this approach is that the first few SIP trunks will always be busy because of outgoing traffic. Is there an easy way to randomize the trunks? I am guessing this will only be possible using AGI? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] where is 1.4.12?
On Aug 29, 2007, at 9:35 PM, Russell Bryant wrote: Another Digium software developer, Joshua Colp, has recently been working on an automated build farm with virtual machines for all of the different operating systems we support. It already has 64 and 32 bit versions of Linux (glibc and uclibc) and FreeBSD, building both asterisk 1.4 and trunk (development for the next major version). It is still growing, with planned support for Solaris 10 x86/x86-64/sparc, and Mac OSX PPC/Intel. Russell, I commend these efforts but if it compiles it doesn't mean it won't crash in certain conditions much less run at all. Proper unit testing is hard to do trust me I have been reading up on the subject and in this type of environment its hard to do proper unit tests without bring up the environment and performing all tests. That in itself is not easy. /b___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] calls being forwarded to neighbor?? please help, thx :)
On Aug 28, 2007, at 8:24 AM, Jody Gugelhupf wrote: -- Now forwarding SIP/9083XXX-0816b208 to 'Local/ [EMAIL PROTECTED]' (thanks to SIP/486-081d4738) Because SIP/486 issued a 302 redirect to 247110358. Check the phone for the forwarding setting. /b ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Astricon Meetup
Everyone, I will be attending Astricon in Phoenix and would like to have a little get together to discuss Open Source Telephony and the challenges we as developers and system integrators face. Exchange ideas and go over some use cases and see how we can all work together to improve our understanding of the dynamics of how everything works together. * Scaleability * Reusability of code * Standards (VoiceXML, MRCP and more) If anyone is interested please email me off list and we'll plan on having a meeting of minds. Thanks, Brian West FreeSWITCH.org ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Distributed System
On Aug 28, 2007, at 10:14 AM, Seysan wrote: Hi all, I'm kind a New to Asterisk.But I'm a Network Administrator with 5 years of experiance. I want to know for an installation with 90 clients, If I don't want to have just 1 server for it, then how is it possible to distribute it among about 3 servers. Should I do it in a cluster (kernel level) or something with SER? I would recommend SER plus Asterisk. I have had great success with using Asterisk with OpenSER. Best Regards, Seysan /b ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Astricon Meetup
haha you going to be there? /b On Aug 28, 2007, at 9:30 AM, Chris Childress wrote: oohs no! Whats up, haven't heard much out of you lately. Chris Brian West wrote: Everyone, I will be attending Astricon in Phoenix and would like to have a little get together to discuss Open Source Telephony and the challenges we as developers and system integrators face. Exchange ideas and go over some use cases and see how we can all work together to improve our understanding of the dynamics of how everything works together. * Scaleability * Reusability of code * Standards (VoiceXML, MRCP and more) If anyone is interested please email me off list and we'll plan on having a meeting of minds. Thanks, Brian West FreeSWITCH.org ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax Problems with SpanDSP
On Aug 28, 2007, at 3:49 PM, Doug Lytle wrote: Christian Peter wrote: Can anybody help me with this issue. Please no switch to Hylafax mails, because I'm very happy with SpanDSP, it integrates nicely and It just show you how many people on this list are pleased with HylaFAX+ Doug -- I'm rather pleased with it. /b ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Load testing/burn-in for Sangoma quad PRI card
Having calls connected for that duration is worthless testing... What you need to do is call setup and tear down many times per second... I recommend trying to accomplish 20-30cps at 1ms to 10ms variable durations. That will expose any bugs quickly. And that my friend is how you expose any bugs... leaving calls up for days is easy... its the setup and tear down that you'll have bugs in. /b On Aug 28, 2007, at 4:11 PM, Erik Anderson wrote: Hello all - I'm about to deploy an asterisk server here at work. Before deploying, I'd like to do an extended load test on the system. I currently have T1 crossover cables connecting ports 1-2 and 3-4. Would there be an easy way to script generating a bunch of calls across these spans? I envision generating 23 calls over the 1-2 span and 23 over the 3-4 span. I'd like to start the calls and then let them stay connected for several days to make sure things are in order. This number of calls would be a *lot* higher load than this system would ever see, but I just want to be safe. Is there currently any script out there that would facilitate this sort of testing? Here's my current config: linux-2.6.21 asterisk-1.4.10 zaptel-1.4.4 wanpipe-3.1.3 libpri-1.4.1 Thanks! -Erik -- Erik Anderson http://andersonfam.org ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Distributed System
This fails to take into account total failure of a machine. NAT mappings and various other variables that are not covered by Dundi or realtime... Best thing is to use OpenSER in the front then failure isn't a huge issue. /b On Aug 28, 2007, at 4:40 PM, Bruce Reeves wrote: Realtime and DUNDi covers all the bases. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax Problems with SpanDSP
On Aug 28, 2007, at 6:28 PM, Matt Riddell wrote: Sorry to hijack the thread, but its great to see you here again Brian! - -- Kind Regards, Matt Riddell Director Thanks... /b ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stereo Conferences?
The HD Codec is just G.722 /b On Aug 27, 2007, at 7:52 AM, Matthew Rubenstein wrote: Do any softphones run the HD codec? What exactly is the HD codec technically called, and is there any info about its codec running inside Asterisk? On Mon, 2007-08-27 at 08:47 -0400, C F wrote: although not stereo i believe its the closest you will get if the codec is supported by asterisk. polycom has now HD codec On 8/27/07, Matthew Rubenstein [EMAIL PROTECTED] wrote: Are there any speakerphones or other conferencing HW phones that play the audio in stereo? Either their own speakers, or jacks for an amp with room speakers? Is there any way for Asterisk to deliver call legs with stereo channels in the RTP stream? If not, is it possible for Asterisk to keep 2 separate calls, or pairs of legs in a conference call, synced exactly enough (including traveling over the Net between the same 2 IP#s) for them to arrive as a stereo pair at the endpoint? -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stereo Conferences?
The 601 has g722 (and its not g722.1 or .2) /b On Aug 27, 2007, at 8:14 AM, Bruce Reeves wrote: The codec is G722 I believe. and Polycom has a conference speaker phone with a subwoofer option that has HD voice. On 8/27/07, Matthew Rubenstein [EMAIL PROTECTED] wrote: Do any softphones run the HD codec? What exactly is the HD codec technically called, and is there any info about its codec running inside Asterisk? On Mon, 2007-08-27 at 08:47 -0400, C F wrote: although not stereo i believe its the closest you will get if the codec is supported by asterisk. polycom has now HD codec On 8/27/07, Matthew Rubenstein [EMAIL PROTECTED] wrote: Are there any speakerphones or other conferencing HW phones that play the audio in stereo? Either their own speakers, or jacks for an amp with room speakers? Is there any way for Asterisk to deliver call legs with stereo channels in the RTP stream? If not, is it possible for Asterisk to keep 2 separate calls, or pairs of legs in a conference call, synced exactly enough (including traveling over the Net between the same 2 IP#s) for them to arrive as a stereo pair at the endpoint? -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stereo Conferences?
FreeSWITCH supports 16k wideband conferences and supports G.722, speex 16k and should work great with the phones that support it. I have personally tested it with grandstream phones. /b On Aug 27, 2007, at 7:47 AM, C F wrote: although not stereo i believe its the closest you will get if the codec is supported by asterisk. polycom has now HD codec On 8/27/07, Matthew Rubenstein [EMAIL PROTECTED] wrote: Are there any speakerphones or other conferencing HW phones that play the audio in stereo? Either their own speakers, or jacks for an amp with room speakers? Is there any way for Asterisk to deliver call legs with stereo channels in the RTP stream? If not, is it possible for Asterisk to keep 2 separate calls, or pairs of legs in a conference call, synced exactly enough (including traveling over the Net between the same 2 IP#s) for them to arrive as a stereo pair at the endpoint? -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sending digits from SIP to Asterisk's VoiceMailMain
If you can get an rtp debug while your pressing digits I can see if maybe your device is sending the digits incorrectly. /b On Aug 19, 2005, at 1:46 PM, Innocent Evil wrote: my sip phone have dtmf relay: rfc2833 asterisk sip.conf have dtmf relay: rfc2833 in associated context. I tried with Inband.. but g729 doesn't support it. I have g729 liscence from digium I havn't try with INFO yet. I prefer to have rfc2833 as dtmf relay. Is there any other thing that can cause this issue? Thanks, -Original Message- From: [EMAIL PROTECTED] Sent: Fri, 19 Aug 2005 14:21:27 -0400 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Sending digits from SIP to Asterisk's VoiceMailMain It sounds to me like an issue of transmitting DTMF tones from the SIP phones. There are several methods that can be used to accomplish DTMF from SIP phones. Of course, you may ask why it isn't just sent as audio (like a regular POTS phone would.) What happens if you are using a SIP phone, hold down the number 4 button for two seconds (so it sends 2 seconds worth of DTMF on the audio stream) and there is some packet loss during that time? You'll have an audio dropout (thus, tone followed by brief silence and tone again.) The remote end will see this as two tones, not one, which obviously can cause undesired results (and is why it's not a good idea to send DTMF in the audio stream.) That being said, look in your sip.conf for a dtmfmode parameter. You can use inband (in the audio stream, not recommended), RFC2833, or SIP INFO. Your SIP phone should also allow you to set how DTMF is sent (although it may not support all of these formats.) Preferably, use RFC2833 or SIP INFO. Find a setting that is available on your phone and on *, and make sure they're set to match. Once you do that, it should work. Jeremy Innocent Evil wrote: Hi, I am using Asterisk cmd VoiceMailMain to manage voice mail. Problem is, voice mail box can't read password sent from SIP phone, but I don't have any problem to read password from the handset attached to my asterisk box. Your help will be greatly appreciated. Thanks,___ 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 -- Jeremy Gault, KD4NED[EMAIL PROTECTED] Network Administrator, WinWorld Corporation Member: Bradley County ACS/RACES/SkyWarn voice: +1.423.473.8084 fax: +1.423.472.9465 fwd: 461771 msn msgr: [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
Re: [Asterisk-Users] codecs order
The way I said is the "gospel" of how it happens. /bOn Aug 16, 2005, at 1:42 AM, Erik Versaevel wrote:That should be controllable by a weight, for example 2 peers: ___ 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 Fax
Just an FYI http://www.groklaw.net/article.php?story=2005080914234645 /b On Aug 16, 2005, at 4:50 AM, Tamas J wrote: Joseph wrote: I'll second that. Hylafax has can handle the job. If you put asterisk in between you are looking for problems. I've the following setup working with asterisk NVBackgroundDetect implemented. PSTN -- asterisk -- hylafax It woks, I would say 90% of the time. There seems to me some timing problems with asterisk, see my posting with subject: real-time priority Hello Joseph, how did you connect asterisk with hylafax? Could you share that? Regards, Tamas ___ 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 Fax
Although Groklaw seems to think that these suits are about faxing, I don't think that they really are. See: http://www.hylafax.org/archive/2005-08/msg00107.htmlLee.No it is really about faxing. As someone that has first hand knowledge of the case outlined on Groklaw, it is in fact about faxing.Go read the two patents very carefully! If you email it you break 638, if you store it you break 021./b___ 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] codecs order
As someone that spent a week or more with anthm refactoring this code I can tell this is how it was when we were done and the code was accepted. So I do know a bit about this area of sip and iax./bOn Aug 16, 2005, at 3:03 PM, Pavel Jezek wrote:I remember many discussions about inteligent codecs negotiation in asterisk, but seems, however, this isn't as simple to implement as it looks... :-( PJ ___ 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] TE410P + SPANDSP fax problem
Time and time again. CHECK YOUR Span clock src./bOn Aug 16, 2005, at 10:18 PM, Ma Zhiyong wrote: Hi, I just setup a fax server by spandsp. But it doesn't look good. Because each fax I received from my fax machine is not completed. I use te410p work with it. While the voice call is good. Any ideas? Trace shows that the fax is received successfully. Aug 17 12:01:10 VERBOSE[19571]: -- Executing RxFAX("Zap/94-1", "/var/spool/asterisk/FAX/1124251267.284.tif") in new stackAug 17 12:01:46 DEBUG[19571]: ==Aug 17 12:01:46 DEBUG[19571]: Pages transferred: 1Aug 17 12:01:46 DEBUG[19571]: Image size: 1728 x 355Aug 17 12:01:46 DEBUG[19571]: Image resolution 7700 x 3850Aug 17 12:01:46 DEBUG[19571]: Transfer Rate: 9600Aug 17 12:01:46 DEBUG[19571]: Bad rows 66Aug 17 12:01:46 DEBUG[19571]: Longest bad row run 22Aug 17 12:01:46 DEBUG[19571]: Compression type 2Aug 17 12:01:46 DEBUG[19571]: Image size (bytes) 0Aug 17 12:01:46 DEBUG[19571]: ==Aug 17 12:01:49 DEBUG[19571]: ==Aug 17 12:01:49 DEBUG[19571]: Fax successfully received.Aug 17 12:01:49 DEBUG[19571]: Remote station id: xxAug 17 12:01:49 DEBUG[19571]: Local station id: Aug 17 12:01:49 DEBUG[19571]: Pages transferred: 1Aug 17 12:01:49 DEBUG[19571]: Image resolution: 7700 x 3850Aug 17 12:01:49 DEBUG[19571]: Transfer Rate: 9600Aug 17 12:01:49 DEBUG[19571]: ==Aug 17 12:01:51 VERBOSE[2999]: -- Channel 0/1, span 4 got hangup___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 ___
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Re: [Asterisk-Users] codecs order
Here is an example: Call comes in via PSTN... ulaw is the native format of the channel. On the sip side you have g729,ulaw as the codec order. That call will end up being ulaw because we send the native format as our first choice above all because we don't want to transcode. /b On Aug 15, 2005, at 1:10 PM, Tony Hoyle wrote: Pavel Jezek wrote: Hi, asterisk will negotiate codecs for both parties independently (use sip show peer peer and look for codec order entry), so, if you have prefered codec g729 for your sip phone/peer, asterisk will use them (regardles of codec setting for other party - if codecs does not match, asterisk will try to transcode between) imho ;-) It does seem to be a weakness of asterisk.. it's creating load on the server when it doesn't need to. Really it should look at the capabilities of both ends and see if there's a common set, and only start transcoding if there's no overlap. Tony ___ 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] T.38 decoding
You do realize that t.38 is the act of taking the t.30 stream and stuffing into UDPTL packet and sending it over a network with a little ASN.1 header added and some reliable delivery kinda like how IAX has reliable delivery of UDP packets used for signaling. This is a very basic description of how its done Go check how t38modem does it.. it emulates a modem and just intercepts the t.30 stream and transports it. /b On Aug 13, 2005, at 11:55 AM, Roger Schreiter wrote: Hi, I searched a while about T.38 decoding, and learned about the bounty for T.38 support for asterisk and some softdecoders and some hardware de- and encoding T.38. Now I wonder, if there is already any (almost) ready to use solution for decoding of T.38 faxes? My szenario would be: - Receiving a SIP call (containing the T.38 fax) by my provider with my asterisk box. - asterisk would forward that SIP call to the converter. - The converter would send the SIP call back to my asterisk box, but now with the fax deocoded to an ordenary anolog fax. Has anyone experience with a working solution, maybe a foreign service provider doing it, or a working (asterisk independent) software? Thanks for any hints! Roger. P.S. Currently I'm trying to understand, what ionidea's T.38 software is already able to do, but I'm still confused. ___ 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] inbound caller id name pri - tnt - asterisk
The TNT can't pass callerid name as far as I know./bOn Aug 9, 2005, at 5:17 PM, Damon Estep wrote: Anyone out there have success getting caller id name from a pri, through a lucent tnt, to asterisk? What about from other media gateways? ___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 ___ 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: OPAL now supports IAX2
What are the advantages of using woomera IAX2 instead of native IAX2?Put woomera aside right now, This is something that brings a cross platform IAX2 stack that can for example be used in Gnomemeeting or anything else that uses OPAL, using a closed and open familiar API. This can be used on windows, linux and anything that OPAL and PWLIB can be used on without any changes. Its a step in the right direction in my opinion./b ___ 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] OPAL now supports IAX2
August 5th, 2005: Craig Southeren announced today that OPAL (http://www.voxgratia.org) now provides support for the IAX2 protocol(Written by Derek Smithies and released under the MPL). This support allows you to use chan_woomera (http://www.pbxfreeware.org) driver developed by Anthony Minessale II to interconnect your asterisk systems and use the IAX2, SIP, and H.323 protocols. I would like to thank everyone involved in Cluecon for all their support! Thanks guys! Brian West Asterlink.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk fails to start
Its very clear your zaptel.conf and/or zapata.conf is wrong.Make sure your devices are registered.. re-run ztcfg -vvv/bOn Jul 28, 2005, at 7:48 AM, Dr. Marios Moutzouris wrote: Hello, This is debug output I get: Jul 28 15:05:49 WARNING[8249]: chan_oss.c:239 sound_thread: Read error on sounddevice: Resource temporarily unavailable [chan_zap.so] = (Zapata Telephony w/PRI)Jul 28 15:05:49 WARNING[8249]: chan_zap.c:924 zt_open: Unable to specify channel 1: No such device or addressJul 28 15:05:49 ERROR[8249]: chan_zap.c:6460 mkintf: Unable to open channel 1: No such device or addresshere = 0, tmp-channel = 1, channel = 1Jul 28 15:05:49 ERROR[8249]: chan_zap.c:10247 setup_zap: Unable to register channel '1-15'Jul 28 15:05:49 WARNING[8249]: loader.c:345 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered channel type 'Tor' == Unregistered channel type 'Zap'Jul 28 15:05:49 WARNING[8249]: loader.c:440 load_modules: Loading module chan_zap.so failed!Ouch ... error while writing audio data: : Broken pipeWarning, flexibel rate not heavily tested! -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.9.6/59 - Release Date: 27/7/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.9.6/59 - Release Date: 27/7/2005 ___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 ___ 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-Dev] Digium to Sponsor a Pizza party at Cluecon
Digium, the creator and primary developer of Asterisk, the industrys first Open Source PBX, will be hosting a pizza party from 4pm to 6pm on the first day of Cluecon. We look forward to everyone coming out to enjoy this opportunity to meet fellow developers and users in a more casual environment. I would like to personally thank Mark Spencer and Digium for their support. Thanks, Brian West Asterlink.com ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ 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] CRITICAL PATCH for anyone using the L option in dial.
http://bugs.digium.com/view.php?id=4760 If you use the L() option on dial and say the latest CVS-HEAD in the past month you're potentially getting screwed out of a lot of money. We originally wrote the L() option for dial and it worked great till someone came along and hijacked the timer for something else thus causing the L option to fail/reset the timer to zero thus causing it to never timeout if someone were to say press a DTMF digit. So if you use this please test this and report back to the bug ASAP. Thanks, Brian West ___ 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] mpg123 - two processes
If you use mp3nb from the sample configs you will have exactly 1 per class. /b On Jul 26, 2005, at 9:38 PM, MF Hulber wrote: Yes, I always have two. MARK. Billy Dunn wrote: Does everyone have two processes running for mpg123? I always have them when I'm running an idle Asterisk box. No calls going in or out and nothing off hook. Is this normal? Thanks! 5008 ?S 0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 4096 fpm-calm-ri 5015 ?S 0:00 /usr/sbin/asterisk 5061 ?S 0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 4096 fpm-calm-ri ___ 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
Re: [Asterisk-Users] super high bandwidth codec
http://www.globalipsound.com Try there. /b On Jul 25, 2005, at 8:15 AM, Eric Wieling aka ManxPower wrote: Steve Underwood wrote: Steve Kennedy wrote: On Sun, Jul 24, 2005 at 11:54:17PM -0700, Storm D. J. Petersen wrote: I don#8217;t know if I have the same experiences. Usually my Skype calls are very garbled at first. I find that my G729 Asterisk calls are better quality. You can try using ULAW if you have the bandwidth. It. might make the #8220;quality#8221; sound better. Maybe it#8217;s your SIP client/hardware phone that is giving you troubles. Skype uses ilbc, and g.729 for PSTN breakout. Skype uses wideband-ilbc. Do yu have a link for wideband-ilbc info? -- 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 ___ 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] We are giving away 3 A101 single-port T1 cards during Cluecon!
Cluecon's ( http://www.cluecon.com ) premier sponsor, Sangoma ( http://www.sangoma.com ), will be giving away 3 A101 single-port T1 cards during Cluecon. The A101 is Sangoma’s next generation hardware designed for optimum support of data and voice over T1 and E1. Register for Cluecon now for your chance to win.Be sure to register by this wednesday, it's the last day I can squeeze in room registrations so please register and pay by that date if possible.Thanks,Brian WestAsterlink.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: [Asterisk-Dev] Cluecon - Who's going ?
I'm going to be speaking about how to use valgrind, gdb and strace to help debug issues... it can be applied to more than just asterisk. /b On Jul 25, 2005, at 10:29 AM, Terry Moore-Read wrote: I'm relatively new to Asterisk and I'm hoping attending Cluecon will be a good way to get up to speed on the project and hear about what others are doing with it. We currently use a Cisco IP phone system at my office, although I just added an asterisk box to provide soft phones to our travelling users. (IAX2 is a lot easier to get through firewalls than cisco's protocols). Terry Moore-Read Lukins Annis, P.S. Spokane, WA -- This message has been scanned for viruses and dangerous content by Lukins Annis, P.S. NOTICE: This email may contain confidential or privileged material, and is intended solely for use by the above referenced recipient. Any review, copying, printing, disclosure, distri- bution, or any other use, is strictly prohibited. If you are not the recipient, and believe that you have received this in error, please notify the sender and delete the copy you received. Thank You! ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ 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] ClueCon Giving Away Voice Hardware (even more than before)
In addition to all the great speakers, ClueCon will be giving away prizes during the conference. You may end up with a chance to go home and put your knowledge to good use with free hardware and VOIP services. Prizes Day 1: 1 Sangoma A101 single-port T1 card 1 Digium Wildcard TE205P or TE210P 2-port T1/E1 cards (winner picks) 2 Pre-Paid Asterlink accounts with 1000 minutes of talk time. Day 2: 1 Sangoma A101 single-port T1 card 1 Digium Wildcard TE110P 1 Digium S101/IAXy 2 Pre-Paid Asterlink accounts with 1000 minutes of talk time. Day 3: 1 Sangoma A101 single-port T1 card 1 Digium TDM400P 4 port analog card. (winner picks configuration) 2 Pre-Paid Asterlink accounts with 1000 minutes of talk time. Tickets will be issued with your ID badge at registration. The drawing will take place at Noon each day right before we break for lunch. Good Luck! Thanks, Brian West Asterlink.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [Asterisk-Dev] We are giving away 3 A101 single-port T1 cardsduring Cluecon!
Cluecon's ( http://www.cluecon.com ) premier sponsor, Sangoma ( http://www.sangoma.com ), will be giving away 3 A101 single-port T1 cards during Cluecon. The A101 is Sangoma’s next generation hardware designed for optimum support of data and voice over T1 and E1. Register for Cluecon now for your chance to win.Be sure to register by this wednesday, it's the last day I can squeeze in room registrations so please register and pay by that date if possible.Thanks,Brian WestAsterlink.com___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev___ 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-Dev] Cluecon - Who's going ?
I'm going to be speaking about how to use valgrind, gdb and strace to help debug issues... it can be applied to more than just asterisk. /b On Jul 25, 2005, at 10:29 AM, Terry Moore-Read wrote: I'm relatively new to Asterisk and I'm hoping attending Cluecon will be a good way to get up to speed on the project and hear about what others are doing with it. We currently use a Cisco IP phone system at my office, although I just added an asterisk box to provide soft phones to our travelling users. (IAX2 is a lot easier to get through firewalls than cisco's protocols). Terry Moore-Read Lukins Annis, P.S. Spokane, WA -- This message has been scanned for viruses and dangerous content by Lukins Annis, P.S. NOTICE: This email may contain confidential or privileged material, and is intended solely for use by the above referenced recipient. Any review, copying, printing, disclosure, distri- bution, or any other use, is strictly prohibited. If you are not the recipient, and believe that you have received this in error, please notify the sender and delete the copy you received. Thank You! ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ 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] ClueCon in 2 Weeks!
I'll talk to your boss if he has a problem! ;) /b On Jul 23, 2005, at 11:03 PM, Terry Moore-Read wrote: Mine did. [EMAIL PROTECTED] 7/21/2005 2:54 PM Brian West wrote: ClueCon is coming in 2 weeks so we urge everyone who plans on attending to register today so we get a proper headcount! snip Thanks, Brian West Asterlink.com snip Anyone else think that was a joke at first impression? Good luck convincing the boss to pay for your way to ClueCon ;-) ___ 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 message has been scanned for viruses and dangerous content by Lukins Annis, P.S. NOTICE: This email may contain confidential or privileged material, and is intended solely for use by the above referenced recipient. Any review, copying, printing, disclosure, distri- bution, or any other use, is strictly prohibited. If you are not the recipient, and believe that you have received this in error, please notify the sender and delete the copy you received. Thank You! ___ 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] Queues and timeouts
PLEASE FOR THE LOVE OF GOD put a NAME in your email program.. I'm sure it makes going back and finding stuff in the archives when you and about 100 other people use Asterisk in their names This goes for anyone that uses Asterisk, Asterisk PBX or any form there of .. lets put a name in there. /rant /b On Jul 24, 2005, at 1:44 AM, Asterisk wrote: Joseph wrote: [snip] exten = _6XXX,2,Busy exten = _6XXX,3,Hangup But the whole point is that I don't want the caller to hear a busy signal or get hung up, I want the Queue to try the next available agent. Which it does at the moment, just with the errors mentioned in the error log file. This busy means, tell the queue app that the agent is busy. The queue app willl go try someone else. The caller will keep hearing music. :) damn, that's so obvious when you say it - I'm sorry that I questioned you, but it smelt wrong ;) Many thanks. I'll go try that now. Julian ___ 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] Re: Re: Business Edition
But I guess I'm wondering ... does the present licensing model discourage other vendors from contributing to *? I'm not sure Sangoma developers could sign the disclaimers even if they wanted to ... but then again I don't know if there's anyone there with anything to offer. I would think that that fact that they're selling hardware that supports * means that there's _some_ sharp cookies there, but perhaps they're just kernel module/driver hackers out to make a quick buck off of Digiums's back without contributing to the core?Sangoma does have a disclaimer on file with Digium as well as a few of their resellers that I know of. app_dictate was sponsored by Sangoma.. written by anthm. Competition is great for hardware vendors regardless of who did what... This is the nature of open source. 99% use it... 1% help out!/b___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Re: Business Edition
Aidan isn't a troll he does raise a very valid point. /b On Jul 23, 2005, at 5:55 PM, Brian Capouch wrote: Aidan Van Dyk wrote: Is this indicative to how Digium people respond to everything (including the company that built the first asterisk-supporting hardware still continuing to make hardware which Asterisk works on)? Nothwithstanding the almost-unparseable syntax, let's not feed this troll . . . b. ___ 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] RE: Business Edition
Or better yet.. modify the disclaimer like I and a few others did to say that the only thing you will disclaim are things you post on the bug tracker! NO UPDATES, NO CHANGES, NO NOTHING! If its not posted under your user on mantis IT IS NOT DISCLAIMED! /b On Jul 23, 2005, at 2:59 PM, William Lloyd wrote: On 23-Jul-05, at 11:22 AM, Kevin Walsh wrote: On Fri, 2005-07-22 at 18:18 +0100, Kevin Walsh wrote: Adam Goryachev [EMAIL PROTECTED] wrote: On Fri, 2005-07-22 at 04:15 +0100, Kevin Walsh wrote: For this reason, I believe that if a fork were ever necessary, it would struggle to beat a distinct path away from the Asterisk Binary Edition Correct, until the point where there is MORE features being added to the forked version of asterisk than the digium version of asterisk. That can't happen, because the ABE could, and probably would, absorb all of the advances in the fork, while forging ahead with the original. Since the fork would be GPL only, if ABE 'absorbed' the new features, then it would 'become' GPL, and therefore would need to be released as GPL, and hence would no longer by ABE :) So, that can't happen. Any other ideas? You're forgetting about the disclaimer documents. Anyone who signed the perpetual agreement and made changes and/or enhancements to the Asterisk code (a fork would still be using Asterisk code) would firstly be obliged to inform the owner, and would secondly have a prior agreement with the owner to allow them to use and close the code. That would neatly bypass the GPL and allow the new code to be folded into the Asterisk Binary Edition. It's unlikely that the current pool of asterisk developers will remain static however. People change jobs, new people find asterisk interesting, people that have not contributed before start to contribute. Assuming a fork were to happen one day. Lots of current developers would stay with the Digium tree because they know it, are digium partners, think it's a better idea, already signed the disclaimer and don;t have an issue with it etc. Many new developers submitting smaller patches would not bother to sign a legal disclaimer and just submit the patch to the full GPL tree. The splinter GPL tree would likely integrate the changes faster and obviously don;t care about a disclaimer. The practicalities of tracking the changes between two source trees would just get more and more time consuming for Digium. They will want to make 100% legal sure that every change they bring into their tree comes from somebody with a disclaimer. Rewriting the missing bits with other programmers would just help the tree's diverge faster. Meanwhile a full GPL tree can just plow ahead without concern. Many companies successfully manage the commercial GPL gap. MySQL for example. The difference in this case is selling a binary only version instead of making money off just hardware and support services/contracts. At the end of the day Digium own the Asterisk trademark and in the world these days, brand name recognition is often more important than the product behind it. -bill ___ 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] Stupid hold music
I'm shocked nobody put the new charlie and the chocolate factory soundtrack on the list... /b On Jul 22, 2005, at 9:57 AM, Michael Graves wrote: I was thing about XTCs stupidly happy M. On Fri, 22 Jul 2005 15:57:07 +0200, Simone Cittadini wrote: Happy Tree Friends' theme is all you need to annoy who's on hold (even if actually I don't know if you can use it for business purpose) Anyway, isn't time to split this list in strictly technical questions-asterisk-users and what's the best provider/hardware/moh/book/distro/ Iwanttocomplaincostheydidn'tsendmethecdrom-asterisk-users lists ? Simone Cittadini IT Manager == COMVERT S.R.L. via F.lli Bressan, 21 20126 Milano - ITALY Tel +39.02.27006796(aspetta un beep)105 [EMAIL PROTECTED] http://www.comvert.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 -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ 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] ClueCon in 2 Weeks!
ClueCon is coming in 2 weeks so we urge everyone who plans on attending to register today so we get a proper headcount! ClueCon was put together by Asterlink, the same team of people who helped shape Asterisk into what it is today by writing features, fixing bugs, offering IRC support and assisting with the management of the development effort. We have produced several real-world solutions based on Asterisk and to top it off, we have invited many of the greatest names in VoIP to attend and share their experiences with you. * Mark Spencer- President of Digium * Kristian Kielhofner - AstLinux * Craig Southeren - co-author of OpenH323 * David Sugar - Lead developer of Bayonne To name a few (there are many more). We have done extensive research and development and had success in enterprise deployment, configuration and clustering of Asterisk. We don't just daydream about it we actually have it in production and send back our changes to the Asterisk CVS. In addition, Asterlink has donated may resources to the community such as: * The Infamous 996 Audio Conference * The Digium CVS mirror * PBXFreeware.org and all it's free asterisk add-on's * The dev.pbxclue.com forum Asterlink developers have produced a great deal of technology that we hope to offer as a service in the near future: * Speech Recognition IVR Input * IMAP Based Voicemail * The app_confcall software that powers the 996 conference. We have produced 2 embedded language modules that are freely available: * res_perl - Embedding Perl into Asterisk * res_js - Embedding JavaScript into Asterisk All of this for the modest cost of $350.00. You could learn enough the first day to justify the price and then you get 2 more days on top of that! Thanks, Brian West Asterlink.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] AstLinux creator to speak at Cluecon
Kristian Kielhofner, the lead developer of the AstLinux project, will be speaking at ClueCon. His latest AstLinux Version 0.2.6 is a complete Asterisk distribution built to run from Compact Flash and uses less than 32mb. Thanks, Brian ___ 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] Support needed
Do you even know what e.164 is? http://www.numberingplans.com/index.php?goto=guidetopic=E164 /b On Wednesday, July 13, 2005, at 09:27PM, Julian J. M. [EMAIL PROTECTED] wrote: Have you tried googling for asterisk e164 ? Julian. On 7/13/05, Will Velez [EMAIL PROTECTED] wrote: Hi my name is Will Velez. Does Asterisk support E164? Thanks ___ 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] Vikrant Mathur lead developer for the open source OSP Toolkit to speak at Cluecon.
Vikrant Mathur is the lead developer for the open source OSP Toolkit available on SIPfoundry. Mr. Mathur began his career in telecommunications as a software engineer at Hughes Software Systems where he focused on softswitch development. After completing his Masters degree in Electrical Engineering at North Carolina State University he joined TransNexus as a senior software engineer developing solutions for secure peer to peer routing, access control and accounting of VoIP traffic on the Internet. If you haven't registered yet please do so ASAP so we can make sure to reserve you a room! Thanks, Brian West http://www.cluecon.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
