similar to: Forcing a codec (take 2)

Hi, I'm having trouble explicitly forcing a codec between sip devices. Am I missing something or is this not really possible? I have a grandstream registering to asterisk, named sip0. Sip0 registers, via sip, to another asterisk box, sip1. When I place a call from the grandstream, it will travel through sip0 to sip1, where it is then placed to the PSTN. Nothing can reinvite, this path is

Sirs/Ladies, Not sure if anyone saw anything like that before... I was playing with an Asterisk setup with a Grandstream BT101 (1.0.5.11) and www.voipjet.com (IAX2). The other devices I have home (Sipura 3k and DTA-310) seem to work just fine, but the Grandstream seems to suffer from one-way voice (remote end can't hear me). The only workaround I found so far (have not spoken with VOIPJet

Hello to all. I have a couple of budgetones connected to Asterisk server. I can establish calls using budgetone with no problem, but when I hang up a Budgetone, Asterisk does not detect the hangup. It seems that the communication goes on in spite of budgetone's hangup. My sip.conf: [general] disallow=all allow=ulaw bindaddr=172.16.60.21 [sip1] callgroup=1 pickupgroup=1 type=friend

Grandstream and Global IP Sound have inked a deal in which Global IP Sound will provide its royalty free iLBC codec to Grandstream. GS will integrate this codec into the BT and HT product lines

Hello, I'd like to implement some public sip uri's that poeple can call into and get an echo test. Is there a way to force a codec so that users can test various codecs? Something like: echo-test at example.com (negotiates whatever codec, is there a way to figure out what codec was negotiated and tell the user) echo-test-g711 at example.com (forces g711) echo-test-g729 at

Hi, I am using R for microarray data anlaysis. When I normalize my data, it converts all my data in to log base 2 values. how can I convert back to log base 10..is there any function in R which I can use or how can I take anti log. or is there any function in R for antilog. Please let me know,..if anyone knows.. Thank you so much, Saurin ===== Saurin's WebWorld:

I have around 20-30 softphones behind NAT .. My sip.conf has nat=yes and they all are able to register and make calls with no problem . My voip carrier supports gsm as well as ilbc .. Server takes calls from sip phones , does call recording in between and forwards to voip carrier . My problem is that half of my softphones use ilbc and rest use gsm and my provider supports both gsm as well as

To me it's a error if I can't complete calls using the ATA configured to use the g726 codec. I just tried it usign a sipura SPA-2000 (preferred codec: G726-32) and I received NOTICES and WARNINGS, but I can't complete a call. On a zap channel: -- Executing Dial("SIP/2007-e4d8", "Zap/1/2217008") in new stack -- Called 1/2217008 -- Zap/1-1 answered

Hello all, I'm having a problem with getting incoming callerid to a lan-connected phone. The Asterisk server is connected to the Internet, and a Grandstream BT101 phone on a lan interface: INTERNET ----(eth0) Asterisk (eth1) ---- Grandstream (192.168.1.51) The phone registers with the Asterisk server as ext 20. I can initiate and receive calls from the Grandstream phone fine. The

Hi! As the upstream of my DSL-connection is very slow, I'd like my sip-phones to use iLBC to connect to my *. My gateway provider only allows ulaw. Hence, I'd like to use the follwing setup: SIP-phone <--iLBC--> Asterisk <---ulaw----> PSTN-Gateway I get the following error: "Unable to find a codec translation path from ilbc to ulaw" Setup SIP-phone: disallow=all

On Thu, 08 Apr 2004 10:14:09 -0400, Steve Kann wrote: >Gary wrote: > >>I have noticed lack of codec negotiation with calls thru a registrated >>asterisk box. >> >>No seen problems with outbound calls, (though I haven't specifically >>tried it), but the problem exists inbound. >> >>Easiest method for testing this was ring in via a sip client set

Need a little help understanding iax codec ordering.... Asterisk A (v1.07) -> iax2 -> Asterisk B (cvs-head, current) If both are configured with (I've purposefully left out other statements that don't pertain to the question): Type=user disallow=all allow=ilbc,gsm Type=peer disallow=all allow=ilbc,gsm a call between the two systems as shown above fails. But if this is arranged

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: --

Hi, how would i know what codec is being utilized? currently i have set allow=ilbc disallow=all. i unset all codecs on x-lite except ilbc. i tried to make a call and look at the channel i see these. does this mean it is using ulaw? how about writetranscode? does that mean there is no transcoding happening on the call? call is going thru, rtp is also going thru. what i would like to know is does

Hello list, I have always though codec translation table is dircetly connected to system speed, utill i came across this: in my lab, i have 2 boxes, First box is an Intel Celeron 1.7 GHZ with 256M RAM: show translation Translation times between formats (in milliseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw

Group, Just want to share with the group my recent findings regarding CODECs/Vocoders and the effect it has had on voice quality and the intermittent noise and breakup problem I have which I mentioned in a previous emailing with the u-law CODEC. Calls again are placed through a SIP phone to a TDM400P to the PSTN. A good reference on the reasoning behind the selection of a CODEC was found in the

hallo all, could sombody please help me, i dont know why nativ bridging is not working when i choose the ilbc codec, with speex it is working,?? iaxcomm (ilbc) ---> asterisk --> ( asterisk2 --> sip grandstream (alaw) ) \-----------------native bridge------------------/ 1. if i use on iaxcomm as default speex, nativ bridging between iaxcomm and my sip phone is working 2.

How can I see which codec is in use during conversation. I can see (for example) which codecs are negotiated before SIP connection, but I don't know which is chosen: 12 headers, 12 lines Found RTP audio format 3 Found RTP audio format 97 Found RTP audio format 2 Found RTP audio format 101 Peer audio RTP is at port 217.10.79.30:15666 Found description format GSM Found description format iLBC

All, Has anybody else experienced garbled voice between a phone using alaw/ulaw and one using iLBC? I have a Nokia E series phone with a preference to use iLBC and this works fine in Asterisk 1.2. However, since moving to 1.4 - I get garbled voice on Inbound (g711->iLBC). Outbound voice seems fine (iLBC->g711) though. It's not a 20/30ms framing issue as the phone uses 30ms