similar to: cli command to check the codec use for the connected calls

Good afternoon, I'm trying to use the CLI command file convert on an Asterisk 1.4.26 server with a TC400B transcoding card. The transcoding card is working well for calls but I have some trouble converting sound files from alaw to g729. The command creates empty file as you can see below... CLI> file convert /var/lib/asterisk/sounds/fr/service_notactivated.alaw

For reference... http://www.voip-info.org/wiki-Asterisk+zap+channels Not sure it is relevant but go ahead and remove the spacing on the channel line so it will read.... channel=>1 Here is the original incoming context you showed. [incoming] exten => s,1,Answer ; Answer the line exten => s,2,Playback,demo-thanks ;for playing a file The Playback looks malformed based upon the wiki

Hi, This seems to be a fairly common question, but I have Googled for this quite a bit and looked at the Asterisk documentation/book and haven't been able to find an answer. My question is: Can I get my IP phone to select a different codec depending on the final destination of each call? I've got these things connected to my Asterisk box: - Snom 300 phone (supports g729 and

You should see something like this.... (I have 8 channels) tuxpbx*CLI> zap show channels Chan Extension Context Language MusicOnHold pseudo incoming en 1 incoming en 2 incoming en 3 incoming en 4 incoming en 5 incoming en 6

Hi there, Here at my company we are willing to use the asterisk IVR system. The problem we are having rigth now is that all our GWs use G729. I've read that in order to asterisk be able to make transcoding from the GSM audio files to G.729, it is necesary to purchase a license from digium. Is this correct? I've seen that licenses are purchased on a per-channel basis. Could

Hi there, I had a problem, basically, I have 4 different types of end users (gsm, ilbc, g729, ulaw). However, I only have one user with my DID provider. My provider supports all 4 codecs. The issue is then: When an incoming call comes in, a codec is negotiated (usually ULAW), later on, when the extension is dialed, we'll see we're doing GSM, and thus transcode. Here's an example

This may seem obvious or silly but have you tried using a different phone cord? A bad phone cord played havoc on a colleague of mine during initial config. Fro what you show as your output from ztcfg, you should have one channel configured successfully then. Your example shows you have the channel set to one so no problem there. If you run ZAP SHOW CHANNELS from the CLI what do you see? W

Hello! I'm facing the following scenario: - Initial call opened to asterisk: SDP g722,alaw,ulaw - Outgoing call to provider started with Invite / SDP alaw, g726 and g729. - Provider sends 183 Session progress SDP: g729, alaw - Provider sends g729 rtp packages But: there is no license to transcode g729. What is asterisk doing? Asterisk decides to stop the call at all: - Sends cancel

I'm communicating a softphone (SJPhone) to a Grandstream phone GXP-2000. The codec order on each one is the next: SJPhone: GSM - iLBC - PCMA - PCMU GXP2000: G729 - GSM - PCMA - PCMU (I have a G729 license, so there's no problem with transcoding G729) In my sip.conf, I've defined the following codec order: disallow=all allow=g729 allow=gsm allow=g726 allow=alaw allow=ulaw And my

We're having an odd issue with codec negotiation from one of our SIP providers. Here's the basic situation. We receive an invite from them advertising support for G711, G729, and G723. In our response, we send back that we support G711 and G729. In about half the cases, this results in no problems, with audio being encoded with G711. The other half of the time, they send us a second

Hi all, i'm trouble with codec setup on an asterisk machine 1.4.18 and some Thomson ST2030 as extensions. In the users.conf file for internal extension i have: disallow=all allow=g729 allow=alaw allow=ulaw Without any codec installed (i mean with original g729 of asterisk) all go fine, calling from an extension to one other: Peer User/ANR Call ID Seq (Tx/Rx) Format

Hi, I would like to change the sequence of messages that was playbacked when I call to the voicemail extension, VoicemailMain. Where can I changed it, can't find the definition for VoicemailMain in extension.conf and voicemail.conf __________________________________________________ Do You Yahoo!? Download the latest ringtones, games, and more! http://sg.mobile.yahoo.com

Hi, I'm after two very specific ringtones for the 79xx's... A dog barking, and a horse either galloping or neighing. I've tried making the sounds, but for some bizarre reason they're not working. I used to make quite a few ringtones for the 79xx's, but I seem to have forgotten how to do it! And to top things off, I can't even find the documentation on Cisco's site

Hello, I have transcoding card TC400P installed in server running Debian with Asterisk 1.4.23. Everything seams to be fine and after I boot up server I see in dmesg: 7.590966] Zapata Telephony Interface Registered on major 196 [ 7.590966] Zaptel Version: 1.4.12.1 [ 7.590966] Zaptel Echo Canceller: MG2 [ 7.610963] zttranscode: Loaded. [ 7.618969] wctc4xxp: tc400b0: Attached to

Hello Michael, you are referring to the following behavior - did I get it correctly?: outbound broken: asterisk offers g722 / g711 to provider (callee), callee answers g711. Asterisk now transcodes between caller and callee (g722 <-> g711). inbound works: call from provider: g711 -> asterisk drops g722 and passes g711 to internal callee -> no transcoding. As far as I know,

Hello, Does asterisk do transcoding when the call goes through the system, codecs are the same but signaling protocol is changed. example: SIP with GSM ---> IAX with GSM What quality destruction happen when I use transcoding? I know this is not a concrete/precise question, but I would like to know how is it in general. What CPU performance is needed for transcoding 30 channels e.g. from

We are evaluating starting a small VoIP consumer based platform. What is the best codec to use with customers using primarily DSL as internet connectivity? I know that g729 is the king-all, but I want to know what the rest of the professional are using out there. g729 has a cost involved, so does the cost really offset the performance? Or is it better to go with g711 to start off? We plan

I received a sample IP/Speakerphone from my friends in China today. Asterisk setup was fairly uncomplicated and I had it running as an extension on my server within a few minutes. Sounds quality of both the receiver and the speakerphone are fine (wife's opinion). Are there any tests I should run with this phone? Following are the specs: - Single line appearance - Alpha display, 2x16 chars -

Can't find an answer to this, but maybe I've not looked hard enough ... Does MixMonitor work without transcoding? ie. if I have a g729 stream passing through and I'm recording it with e.g. MixMonitor(/dump/filename.g729,b) and specify g729 in the filename, does MixMonitor transcode both legs of the stream to a format it can then "mix" then transcode it back to g729 to

If you don't want to proxy the media through * the put this setting: canreinvite=yes this will allow the 2 end points to connect directly for the RTP bypassing you. otherwise I have noticed the same when I try to proxy I have to make sure everyone is using the same codec or it doesn't work well. .o-------------------------------------------------------o. Brian Fertig NOC/Network