similar to: G729 Registration unsuccessful

Hello I've been using VoIP over a not so reliable net: I usually get a 5% to 10% packet loss and a very high jitter. I tried several codecs and parameters, and the only thing left to test is PLC (Packet Loss Cancellement). Have the astesrisk and digium people implemented PLC?, Are they implmementing it now? and, if not, Where can i find an implementation? Thanks in advance -- Jorge

Hi I' ve just installed TE410P and asterisk-0.7.2 from tar.gz on fedora core 1. When i start asterisk it shows me this: /usr/lib/asterisk/modules/chan_zap.so: undefined symbol: ast_pickup_call Apr 19 16:52:32 WARNING[-1085304704]: loader.c:358 load_modules: Loading module chan_zap.so failed! Where do i look, how can i debug? Thanks in advance Jorge Verastegui G RedCetus S.R.L

Hi Please help! I have one X101P and TDM400P in my asterisk Box When i make a call from * to PSTN, everything goes Ok, When the PSTN hangups or * hangups, the busy tone is detected and * disconnects the channel without problems. The problem occurs when the call comes from PSTN. When * hangups, the other end (at pstn) does not hangup, it only presents silence. Please tell me how to solve this

Hi, i have a problem trying to register the codec G729, my licence is valid but when i try to Register i got the following error: "Registration unsuccessful... Error code: 110 ERROR! Your Internet connection is probably behind a proxy and the Registration program can't communicate with our server, however it has created the file: /var/lib/va-infoclient

Hi, I've connected a TE110P from digium with a 2 E1 to a Siemens PBX using asterisk from ubuntu linux. Everything is working as expected. This box is being used as a H323 gateway to the pstn. There are few complains but it is working pretty well overall. There is one thing that is bothering me. Asterisk says: May 22 05:03:39 WARNING[9360]: PRI: !! No channel map, no channel, and no ds1?

Hi I'm buying a new box and it brings the new PCI-e standard and not the old PCI slots. I would like to know if the Digium Wildcard TDM400P and <http://www.digium.com/index.php?menu=wildcard_tdm400p2>Wildcard TE405P will work with this PCI Express. <http://www.digium.com/index.php?menu=wildcard_te405p> Has anybody worked with PCI-e yet? As far as i understand, the Wildcard

Hi, I have a serious problem to configure Cisco AS5XXX and Asterisk , I trying to use asterisk for PSTN(A)----> Cisco AS5xxx ----> ASterisk---->PSTN(B) (No Nat, no Firewall) I hear (on the PSTN(A)) clearly what the other person is saying, but the other person (on the PSTN(B) side) hears nothing from PSTN(A). I use tcpdump for debug de rtp trafic, and ouput contains

Hi folks, where can I find the R2 beta code for Asterisk? Best, PauloHM -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030806/9c7a0660/attachment.htm

Hi, I've connected a two T100P from digium with a 2 Rhino channelBank. Everything is working as expected. but I have occasional Falls, asterisk take 99% of CPU resources, with the following report May 24 14:52:41 WARNING[11441]: We're Zap/1-1, not H8?F??n** May 24 14:52:41 WARNING[11441]: We're Zap/1-1, not H8?F??n** May 24 14:52:41 WARNING[11441]: We're Zap/1-1, not H8?F??n**

I have the following problem: I have installed two T1 digium card (old T100P cards), plus a TDM400 with 4 fxo modules. Several times in the week I have thousands of warnings like these in the log Aug 1 08:54:47 WARNING[2243]: We're Zap/18-1, not G\uffff\uffffG Aug 1 08:54:47 WARNING[2243]: We're Zap/18-1, not G\uffff\uffffG Aug 1 08:54:47 WARNING[2243]: We're Zap/18-1, not

chan_sip.c6524 reload_config= unable to get ip address from asterisk, sip disabled The ip address is working fine, Internet works great. Can anyone help...Thanks

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

According to digium's site, "Note: Please do not attempt to use the G.729 code in a SCSI-only system. We are currently working with VoiceAge to correct this issue." (found at http://www.digium.com/index.php?menu=asterisk_g729). Does anyone know what these issues are? Can anyone define SCSI-only system? I know this sounds kinda dumb, but I have a server with SCSI and IDE interfaces,

Original Post ---------------- I have an Asterisk related problem with mutualphone. I can connect to any number with any softphone that I am using (iaxcomm, SJPhone, and a few others). Also I have a Grandstream BT 101. But I cannot call (via Asterisk) to mutualphone destinations. Other destinations go fine. A working phone call (e.g. from iaxcomm) gives the following on the console: --

I own a G729b codec from voiceage which I had from Digium a couple of months ago , I friend of mine had the new Digium G729 codec which registers in the asterisk as a Annex A/B codec, the problem that we saw is that the call goes thru find but we cannot here any sound. Asterisk is showing this : -- SIP/10.10.1.1-babc is ringing -- SIP/10.10.1.1-babc answered

I've just setup a new asterisk server, and I need to have G.729 working on this system. The problem is I don't have any IDE drives (and therefore no /dev/hda etc), but only /dev/sda.   Is there really *no* way to license G.729 on a SCSI-only system? IMHO it's really stupid to replace an entire server because of a licensing issue. There *must* be a solution.   Anyone, please? Or at

I have multiple voiceage g729 licenses installed on a RH9 box, and have a remote C7960 configured to use it (low bandwidth). In calls like: Remote C7960 -> g729 -> asterisk -> g711 -> C7960 the audio is oftentimes rather choppy. Changing the remote 7960 to use g711 seems to eliminate/reduce the choppyness. Any ideas on what might be behind this?

Hello all, We have the following problem: When calling via iax, the sound is off after a while - most often after about 5 minutes (sometimes later or earlier) - at one end or at both ends. While the channel is up, and packages are still being transmitted, you just can't hear anything. Sometimes you can hear something just a little, but with the voice greatly distorted, sounding like a

Hi All, First of all, I'm sorry if my question is offtopic on this list. In such case please ignore this post and/or contact me directly. I'm asking my questions there because I feel you had similar problem before starting developing Speex. My story: my friend developed 3gpp content creator and he would distribute it in binary form. But there is problem with AMR licensing (the terms

Seeing that many people here hit problems with activating their G.729 licenses, I decided to post my opinion. I have purchased two G.729 licenses, for my private use. I did this even though VoiceAge makes G.729 free for private use, as Windows libraries. I guess a sufficiently motivated person could take the COFF libraries, run them through objcopy on cygwin (producing ELF .o files) and link them