similar to: No compatible codecs

If this has been covered before - I appologize. We use some Sipura SPA-2000's with the g711 codec and all seems fine (except for the occasional failure to register errors in my asterisk logs - but I will save that for another post). g711 call quality is on par with our Cisco 7960's. However, when using the g729 codec, the call quality on the Sipura device goes downhill on the PSTN side

Hi All, I have been reading through the list quite a bit, and I am going to post this more as a poll than anything else. I am working on setting up a very small business with something that resembles a professional voice system. My idea is to use Asterisk with a SIP provider and SIP clients. I currently have a Vonage account already. So adding the 9.99 a month Soft Phone would be easy.

Hi, I have an Asterisk Server with a Public IP (No NAT) and I'm trying to register an adapter SIPURA 2000 behind a NAT Linksys Router WRT54G, but It has been impossible. Into the SIPURA Port #1 I have a Termination with MutualPhone and it works perfect. Into the SIPURA Port 2 I have connection to Asterisk Server but I can't reach it. The configuration on both ports is the same.

We are looking for a flatrate long distance provider to integrte with a hosted Asterisk implementation. I would like to have a SIP/IAX handoff to whatever carrier. We want the ability to turn LD to any of our subscribers for a flat rate as well. This request is for USA only. Int'l would be a per minute basis. Any sugggestions are greatly appreciated. Thanks, Ty Carter

Anyone having problems with LiveVoIP lately? I am seeing failed outgoing calls. Calls that are being routed to wrong numbers. DID's that ring busy. For the pass 2 days I am unable to pass CID. Is anyone else have these problems? Can anyone recommend a Quality VoIP provider?

Hello, I've encountered the following error, trying to start a domain: error: internal error Child process (PATH=/usr/local/sbin:/usr/local/bin:/usr/sbin:/usr/bin:/sbin:/bin:/usr/X11R6/bin LIBVIRT_DEBUG=2 LIBVIRT_LOG_OUTPUTS=2:syslog:libvirtd /usr/libexec/libvirt_lxc --name vm_14 --console 18 --handshake 21 --background --veth veth1) status unexpected: exit status 1 I see nothing relevant

Hi, I have licensed the digium G.729A codec. But for some reason incoming and outgoing calls will ALWAYS use G.711a. When I force my phone to only accept G.729 then an incoming call from ZAP goes straight to my voicemailbox as the phone doesn't accept the codec Asterisk wants, even if I force it in sip.conf. Is there anywhere else I need to look to tell ZAP to use G.729 preferrably ? The

Hi, I am trying to replace my Cisco 5300 gateway with my new Zap TE405P card. Incoming calls and outgoing calls between my cisco and my SIP phone works fine on G.729. Recording messages in the asterisk voice-mailbox also works fine from both my SIP phone as well as PSTN -> Cisco -> Asterisk. I have licensed the digium G.729A codec. When I connect my ISDN PRI to my Zap card and I call

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

Hi there, im having some problems with my asterisk box, it seems codec is the principal cause of it. Reading in some forums i found that i can get the new codec_g729 from ftp://ftp.digium.com/pub/telephony/asterisk/g729/new_codec_binary/codec_g729b.so i checked it but the directory new_codec_binary doesnt exist. Anybody knows where can i found it?? Thanks for your help. Carlos Andres Medina

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

Got one weird one and one prob easy one. 1. I have upgraded our BT101's to Program--1.0.5.0 Bootloader--1.0.0.17 HTML--1.0.0.34 VOC--1.0.0.6 after doing this i have some phones on different subnet's ie 255.255.255.248 or .192 or .252 and i am now unable to login to these phones from different subnet's . I have one at home which is on a .248 ( Using an external IP for the phone )

Hello, Has anyone noticed that if you don't pick up a BT101 phone in 60 seconds it stops ringing and acts like it was never called ? Or is it just something I missed ? If it matters for something I have call waiting enabled on the phone. Product Model: BT100 Software Version: Program--1.0.5.11 Bootloader--1.0.0.18 HTML--1.0.0.37 VOC--1.0.0.6 Custom Ring Tone:

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,

Hello, I have that problem with codec G729. Please can somebody help me! WARNING[16384]: File codec_g729b.c, Line 413 (load_module): Unable to initialize va stuff: -1 == Detected 4 licensed G.729 transcoders WARNING[16384]: File translate.c, Line 219 (calc_cost): Translator 'g729tolinb' does not produce sample frames. == Registered translator 'g729tolinb' from format G729A to

In the following scenario, the user calling from a SIPphone registered phone is able to transfer the called user to another extension. sip.conf: [general] port = 5060 context = from-sip register => number:password@proxy01.sipphone.com extensions.conf: [from-sip] exten => s,1,Dial(SIP/111&SIP/117) exten => 111,1,Dial(SIP/111,20) exten => 117,1,Dial(SIP/117,20) 1. The calling user

I've done all the googling I can on this, and have come to the conclusion that a Grandstream BT101 can be abused to be a door phone. Could someone with access to one, confirm that the following is possible? Researched: 1. When set to auto-answer, dialing the phone will result in a short beep and instant speaker-phone connection. 2. When pressing the "message" button while

Hi, I am wondering if anyone has gotten the BT101's to work with the paging in Asterisk? I know that the phones themselves have an auto-answer option and if I turn it on every call is auto answered. I want to be able to call the extension normally and have it ring normally but if someone dials # and the extension to have it auto answer for intercom purposes. Anyone have this working?

I think I am missing the whole purposes of *. i see that it can do mainy things, but in laymans temrs I am not sure what it does. I am very proficient in Linux and would like to use * for the following: 1) I would like to get rid of my landline(verizon) and use voip as my main means to communicate on the telephone. I would like to be able to plug in my plain old phone into my linux box and

hi, I'm currently trying to use Speex with Asterisk from my OpenH.323 client. It seems to mismatch the codecs, below is my log from Asterisk. My Openh323 client crashes in responding to a Speex request for bits per frame. I'm guessing it either isn't running the codec correctly or doesn't support the same subset of speex codecs as openh323. (I'm using speex-1.0.1 with