similar to: Unable to find a codec translation path from ilbcto ulaw

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

I'm doing some final check-outs before upgrading from 1.4.x to 1.6.x and I've encountered a problem playing back a .wav file to an Ekiga client: My dialplan looks like: exten => 730,1,answer exten => 730,n,playback(/home/phones/common/moh/moha/Sovereign) exten => 730,n,hangup Sovereign.wav is a .wav file that plays nicely on my 1.4 server. Here is what the console displays:

Dear Folks, I'm running Asterisk 1.4.31 server, on an Ubuntu 9.10 system. My scenario is simple: connection to the PSTN directly via SIP, using g729 codec, and connection to the softphones (X-lite 3.0 build 56125) trought local network, using ulaw codec. Sometimes, I got messages like: [Jul 1 15:26:16] WARNING[26483]: chan_sip.c:5514 process_sdp: Unsupported SDP media type in offer: image

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

Hi Users, I was wondering if someone of you have the same thing on CentOS 64x? asterisk01*CLI> core show translation Translation times between formats (in microseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 siren7 siren14 slin16 g723

Does anybody has got the some problem? The grandstream ATA 486 schould support almost all codecs, but it doesn't work. I get the following message when I force the use of different codec WARNING[9529]: chan_sip.c:2765 process_sdp: No compatible codecs! Feb 3 11:17:15 NOTICE[9529]: chan_sip.c:7395 handle_request: Unable to create/find channel What could I do to see some more detailed

Hi, I cannot get SIP channel working with folowing codec configuration: [sip] disallow=all allow=g723.1 ;I need this codec between sip phones (BT100) allow=ilbc ;Use this codec to others Calling between BT100 SIP phones is OK - asterisk makes native bridge (with g723.1) between them. When I'm calling from SIP to other channel (iax,zap,...), asterisk is not able to chose right codec

Dear all, I'm trying to use ulaw and alaw with Diax and Asterisk but I'm not able to, I got the following error message: Mar 20 11:47:59 NOTICE[7099]: chan_iax2.c:6350 socket_read: Rejected connect attempt from 192.168.0.55, requested/capability 0x8/0xc incompatible with our capability 0xfe02. I do not understand why because my Asterisk box load these codecs properly! Does somebody

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

I seem to be having a problem with transcoding and/or agreeing on a valid codec. I am running a new image pulled from CVS at 1:30 PM CST. The issue occurs when I try to make a call to a toll-free number over sipphone.com. Here's what I see in the console: NOTICE[1259545280]: File channel.c, Line 1478 (ast_set_read_format): Unable to find a path from G729A to ULAW NOTICE[1259545280]: File

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

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

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

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

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

I'm reposting this to the list.. My spam filters didn't like the list host. :( If anyone was able to respond to the mail below, can you send it again please? Thanks. ------------------------------------------------------------------------- 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

Justin Killen wrote: <snip> > > Whenever I try to copy this callfile into /var/spool/asterisk/outgoing/ > I get these 3 lines repeating over and over (I?m not 100% sure which > entry is first): > > [2015-02-16 16:56:02] WARNING[9737][C-0000f8a7]: channel.c:5353 > set_format: Unable to find a codec translation path from (nothing) to (slin) > > [2015-02-16

Hi all, I'm using 'show translation' to help dimension my system, but I confused by the results I get. My 2 test systems (results below): an AthlonXP 2000+ (1.3GHz) and a Pentium D930 (duo-core, 3.0GHz) produced similar results (D930 is slightly faster). Googling shows that others have similar results running on other CPU speeds >2.0GHz. At first glance, it would look like the