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...gs and shows CID - BUT, when I answer the phone, there is no audio either way. I thought this was a firewall issue but the clients ring and I CAN leave and retrieve voicemail. My next assumption is that it is some codec issue. The Polycom defaults to G.711u. I've tried changing this to G.729AB - but there problem persisted. Any ideas? Thanks in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050602/f600ef01/attachment.htm

...with asterisk? We we're able to get the unit to register with asterisk, but when trying to place a call, no codec was compatible, even though I had all of the following enabled on the patton ... # G.711 A-Law/?-Law (64kbps) # G.726 (ADPCM 40, 32, 24, 16 kpbs) # G.723.1 (5.3 or 6.3 kbps) # G.729ab (8kbps) the link to this product is : http://commerce.patton.com/pe_products.asp?category=51&MiDAS_SessionID=e41363efa86e409caf79ab1fd9b32e49 ? thanks for any help, Michael Lyszczek

...G7231A6K3 - G.723.1A(6.3k) ; G726 - G.726(32k) ; G72616K - G.726(16k) ; G72624K - G.726(24k) ; G72632K - G.726(32k) ; G72640K - G.726(40k) ; G728 - G.728 ; G729 - G.729 ; G729A - G.729A ; G729B - G.729B ; G729AB - G.729AB ; GSM0610 - GSM 0610 ; MSGSM - Microsoft GSM Audio Capability ; LPC10 - LPC-10 ; Number of frames in RTP packet (if not specified) is 1. ; codec=G711U frames=20 codec=GSM0610 frames=4 codec=G7231 frames=2 codec=G729 frames=2 codec=G711A frames=20 langua...

...O. Box 941 Address2 : City : Sofia State : Outside the US, Mexico, or Canada Zip/Postal Code : 1000 Country : BULGARIA SupportType : DSP Tools Issue dsp_using : Other dsp_using_other : I don't know. G.729ab and other voice codecs platform : Other platform_other : PCI Card emulator : None software : None software_other : operating_system : Other operating_system_other: Linux target_os : Other target_os_other : Linux design_st...

I've got all three CODECs the 300 supports -- G.711u, G.177A, and G.729AB -- enabled, I've changed the order, I've got them all in allow lines in my sip.conf, as follows: disallow=all allow=ulaw allow=alaw allow=G729 From "sip debug" I get the following snippets: =================================================================================== Fou...

...- G.711 A-Law ; G7231 - G.723.1(6.3k) ; G72316K3 - G.723.1(6.3k) ; G72315K3 - G.723.1(5.3k) ; G7231A6K3 - G.723.1A(6.3k) ; G7231A6K3 - G.723.1A(6.3k) ; G728 - G.728 ; G729 - G.729 ; G729A - G.729A ; G729B - G.729B ; G729AB - G.729AB ; GSM0610 - GSM 0610 ; MSGSM - Microsoft GSM Audio Capability ; LPC10 - LPC-10 codec=G729 codec=G7231 codec=G711A codec=G711U

Hello, I'm trying to find out if Asterisk will support plain G729 or G729b. I've read all over that it supports G729, but I can't seem to find any explicit remarks regarding the specific versions of the codec Asterisk will support. I noticed that Digium allows Asterisk users to register and download G729a, but refers to it as G729 on it's pages. I also noticed that on

What is the configuration of H323.conf and openphone in order to run openphone and asterisk together ?

...values for the "codec" option are: ; G711U - G.711 u-Law ; G711A - G.711 A-Law ; G7231 - G.723.1(6.3k) ; G72316K3 - G.723.1(6.3k) ; G72315K3 - G.723.1(5.3k) ; G7231A6K3 - G.723.1A(6.3k) ; G7231A6K3 - G.723.1A(6.3k) ; G728 - G.728 ; G729 - G.729 ; G729A - G.729A ; G729B - G.729B ; G729AB - G.729AB ; GSM0610 - GSM 0610 ; MSGSM - Microsoft GSM Audio Capability ; LPC10 - LPC-10 ; Number of frames in RTP packet (if not specified) is 1. ; codec=GSM0610 frames=4 codec=G711A frames=20 ;codec=G7231 -------------- next part -------------- ; ; SIP Configuration for Asterisk ; [general]...

Hi All, For a small installation using ITSPs via DSL is G.729 a worthwhile exercise? I have G.729 capable SIP phones and my ITSPs cupport the codec so I could go end-to-end without transcoding. What's call quality like compared to G.711, GSM or iLBC? Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist www.pixelpower.com

...G7231A6K3 - G.723.1A(6.3k) ; G726 - G.726(32k) ; G72616K - G.726(16k) ; G72624K - G.726(24k) ; G72632K - G.726(32k) ; G72640K - G.726(40k) ; G728 - G.728 ; G729 - G.729 ; G729A - G.729A ; G729B - G.729B ; G729AB - G.729AB ; GSM0610 - GSM 0610 ; MSGSM - Microsoft GSM Audio Capability ; LPC10 - LPC-10 ; Number of frames in RTP packet (if not specified) is 1. ; ;codec=G711A ;frames=20 ;codec=G711U ;frames=20 ;codec=GSM0610 ;frames=4 ;codec=G7231 ;frames=2 codec=G729A frames=...

Folks! I would appreciate if someone could send me a simple working h323 configuration file oh323.conf that is part of asterisk@home installation. I have tried to use the oh323.conf content listed on WIKI but it is just not working as my H323 endpoint ( PA168 based ATCOM Phone) cannot register. I need a working example of this file for similar phone. Seshu

Hi all, i have big problems to make a h323 call over the gatekeeper from my provider. The provider demanded following account data: H323 ID: XXX-XXX-XX-X DetinationNumer: XXXXXXXXXXX I have configured the oh323.conf following: gatekeeper=XX.XX.XXX.XXX alias=XXX-XXX-XX-X Isx the alias equal to the h323id ? And how i have to make a call with the dial app ? I have following config: exten

...G7231A6K3 - G.723.1A(6.3k) ; G726 - G.726(32k) ; G72616K - G.726(16k) ; G72624K - G.726(24k) ; G72632K - G.726(32k) ; G72640K - G.726(40k) ; G728 - G.728 ; G729 - G.729 ; G729A - G.729A ; G729B - G.729B ; G729AB - G.729AB ; GSM0610 - GSM 0610 ; MSGSM - Microsoft GSM Audio Capability ; LPC10 - LPC-10 ; Number of frames in RTP packet (if not specified) is 1. ; ;codec=G711A ;frames=20 ;codec=G711U ;frames=20 ;codec=GSM0610 ;frames=4 ;codec=G7231 ;frames=2 ;codec=G729A ;fr...

...have in sip.conf in >>general and phone entries? >> >> > > Hi Wilson, Thanks for the reply. I didn't know anything about codecs but I've tried to look up what I can. The Polycom documentation (SIP admin guide) says the hone supports G.711u-law, G.711a-law, G.729AB, SID and RFC2833. The phone configuration files say that the preference is for u-law, a-law and AB in that order. My sip.conf file says: disallow=all allow=ulaw allow=alaw I would guess that means I'm ok (i.e. ulaw is good on both sides) but this is a new area of * for me. What do you...