This is what I found today googling on the Web, also I post it now in order to save time to others:
The G726-32 codec: * It has been determined that previous versions of Asterisk used the wrong codeword packing order for G726-32 data. This version supports both available packing orders, and can transcode between them. It also now selects the proper order when negotiating with a SIP peer based on the codec name supplied in the SDP. However, there are existing devices that improperly request one order and then use another; Sipura and Grandstream ATAs are known to do this, and there may be others. To be able to continue to use these devices with this version of Asterisk and the G726-32 codec, a configuration parameter called 'g726nonstandard' has been added to sip.conf, so that Asterisk can use the packing order expected by the device (even though it requested a different order). In addition, the internal format number for G726-32 has been changed, and the old number is now assigned to AAL2-G726-32. The result of this is that this version of Asterisk will be able to interoperate over IAX2 with older versions of Asterisk, as long as this version is told to allow 'g726aal2' instead of 'g726' as the codec for the call. Now, I can complete a call but I have now audio yet. I believe that I need to restore the original rtp.c and recompile it. I hope this help someone else... Today before to find the new information: Since nobody answer my previous question (It looks like g.726 is a bad word). I have this scenario: One box with Asterisk 1.4.0 beta 2 IAX to anothers Asterisk working properly. As an ATA I have only one Grandstream HT496. Two lines on the ATA 727 & 726. >From outside I can call any of those two extensions if: I defined both as ulaw (G.711) One as ulaw and the other as G.729 Only one at the time if I define both as G729 Only the G711 if I define one as G726 and the other as G711. No way to make calls between the two extensions if both are G726, or both are G729 Or if one is G726 and the other is G729 or ulaw. I allways get codec not match or invalid on the console. Grandstream claims that they can handle two g729 calls at the time ( I never was able to do this), and even when I change the rtp.c the G726 is not working at all. (I 'm not sure but it looks like is more an Asterisk problem, since the modification was made for Sipura boxes, but even when other phones, after the 1.2.6 we have fast busy problems on those devices with that code) So at this point. if someone has any other experience that want to share I'll appretiate it. Thanks Carlos Alperin -----Original Message----- From: Carlos Alperin [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 05, 2006 2:29 PM To: '[email protected]' Subject: G.726 on Asterisk 1.4.0 Importance: High I'm trying to make a new box with Asterisk 1.4.0, work with one ATA GrandStream 496 and G.726. However I modified the rtp.c as suggested for the Sipura's ATA with USE_DEPRECATED_G726=1 is not working. Someone knows about this? Thanks, Carlos Alperin _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
