I have come to realize that I don't have to have a g729a license in order to make use of an ATA-186 or 7460 connecting to another 7460. I just need to allow the codec in sip.conf. Now what ramification does that have when I dial out over one of my analog line (connected to * by a channelbank and a T100P) using my 7460 or ATA-186. The only benefit I am looking for is reduced bandwidth utlization when I call into my office. If I call through the analog lines, do I need to purchase g729 codec licenses? Also, does anyone know if iconnecthere can handle encoded G729 calls?
On Wednesday 10 September 2003 15:47, Kim C. Callis wrote:> I have come to realize that I don't have to have a g729a license in > order to make use of an ATA-186 or 7460 connecting to another 7460. I > just need to allow the codec in sip.conf. > > Now what ramification does that have when I dial out over one of my > analog line (connected to * by a channelbank and a T100P) using my 7460 > or ATA-186. The only benefit I am looking for is reduced bandwidth > utlization when I call into my office. If I call through the analog > lines, do I need to purchase g729 codec licenses?yes> > Also, does anyone know if iconnecthere can handle encoded G729 calls?yes... iconnect handles G729 just fine> > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users
Hi, I recently purchased some G729 licenses for asterisk. I'm concerned with the registration process. My build tools are not physically located on the same machine from which I build asterisk. I build RPMs on another machine and then install them on my production server. Am I going to cause myself trouble by runnning Registration on my non build host? Thanks -z -------------- next part -------------- A non-text attachment was scrubbed... Name: S. Zachariah Sprackett.vcf Type: text/x-vcard Size: 520 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20030912/982be33e/S.ZachariahSprackett.vcf
Hello, I've inherited a (now) broken asterisk implementation. It seems as if there are currently codec tanscoding issues in this box. Specifically I am receving calls from a SIP proxy in G.729 and attempting to transcode them to ULAW. My asterisk installation was working up until yesterday. The information I have found on the voip-Asterisk wiki is for a third party open-source implementation of asterisk (which doesn't seem to be what was deployed on this asterisk box at all); however there are glaring warnings all over the place about possible legal implications of using this codec. Moreover the words "GPL Violation" are printed and duplicated many times over in the form of an E-Mail from Mark Spencer attached to the Wiki. The documentation does not; however tell me the correct and 100% legal way to license and implement G.729. In short this server is broken and I don't know what to do because I'm afraid of possibly being in direct violation of the GPL by following the voip-Asterisk wiki's documentation. Can someone kindly point me to an RTFM in the form of Digium-supported licensing options for G.729 and some technical documentation on how to acctually do the implementation? Or perhaps someone could suggest a fix for my current issues: Here's a log snippet from my asterisk console: Feb 8 22:19:19 NOTICE[1125329728]: channel.c:1683 ast_set_read_format: Unable to find a path from ULAW to G729A Feb 8 22:19:19 NOTICE[1125329728]: channel.c:1650 ast_set_write_format: Unable to find a path from G729A to ULAW -- SIP/2101-aaf2 answered SIP/19544342000-375a -- Attempting native bridge of SIP/19544342000-375a and SIP/2101-aaf2 -- Attempting native bridge of SIP/19544342000-375a and SIP/2101-aaf2 Feb 8 22:19:19 NOTICE[1225991488]: channel.c:1683 ast_set_read_format: Unable to find a path from G729A to ULAW Feb 8 22:19:19 NOTICE[1225991488]: channel.c:1650 ast_set_write_format: Unable to find a path from ULAW to G729A Feb 8 22:19:19 WARNING[1225991488]: chan_sip.c:1797 sip_write: Asked to transmit frame type 256, while native formats is 4 (read/write 4/4) Now both channels in question have allow=ulaw and allow=g729 Any help at all would be appriciated. Regards, Daniel
http://www.digium.com/index.php?menu=software_products> -----Original Message----- > From: Daniel Corbe [mailto:daniel.junkmail@gmail.com] > Sent: Tuesday, February 08, 2005 9:00 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] g729 > > Can someone kindly point me to an RTFM in the form of > Digium-supported licensing options for G.729 and some > technical documentation on how to acctually do the implementation?
Has someone fixed g729 for use on FreeBSD as yet?
Hi All,
I have configured Line1 (2011) and Line2 (2012) in Sipura SPA-2000 (latest
Firmware) to use G729. In sip.conf I have set disallow=all, allow=g729
If Line1 is in use by an agent, then Line2 won't work and vice versa
(Inbound Calls Only). I have 40 license for G729. so there shouldn't be any
issue with the license.
I'm getting the following error msg:
-- Called 2012
-- Got SIP response 488 "Not Acceptable Here" back from
192.168.10.103
== No one is available to answer at this time (1:0/0/0)
== Auto fallthrough, channel 'IAX2/aaa@111.11.11.137:4569-5' status is
'NOANSWER'
-- Hungup 'IAX2/aaa@111.11.11.137:4569-5'
If I change 2012 to ULAW, it works fine. It seems that I can't have two
lines configured as a G729.
Do you guys have any idea why this happening?
Regards,
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On Fri, 2005-06-17 at 12:33 -0400, David wrote:> Hi All, > > > > I have configured Line1 (2011) and Line2 (2012) in Sipura SPA-2000 > (latest Firmware) to use G729. In sip.conf I have set disallow=all, > allow=g729 >Please take the time to read the Sipura documentation where it states that you can only do ONE G729 call at a time on a SPA 1000, 2000 and 3000. The processor in the unit is not powerful enough to do 2 G729 calls. You have to allow ulaw or alaw so the other line can make a call while the first is busy.
HI I loaded "codec_g729-gcc-pentium4.so" module for testing g729 with dialpad service. When I make an outbound, the destination rings, but the server exits and stops running when the destination party picks up the phone. Any clues on this Thnx Hitesh Sharma
I registered 5 g729 codec and the result was , I cant use these channels because all channels are not available even I have no call on the system 5/0 encoders/decoders of 5 licensed channels are currently in use Please help ********************************************* No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. ********************************************* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060809/13faad89/attachment.htm