That looks like exactly what I want, we are currently on 1.2, ill see if i can hack similar functionality into it, if not ill have to upgrade to 1.4 (probably best anyway)
Thanks for the pointers. ________________________________ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James FitzGibbon Sent: 11 May 2007 15:15 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Log CODECS in CDR's On 5/11/07, Morgan Gilroy <[EMAIL PROTECTED]> wrote: At the moment to find the codecs used I have to look though the sip trace or show channels/show channel (annoying when you have 50+ channels). Im just trying to find an easier and quicker way to keep track of the codecs used to help with debug etc. The closest variable iv found is, "${SIP_CODEC} Set the SIP codec for a call" Ill see if NoOp (${SIP_CODEC}) shows the codec that was used without me setting it though I don't think it will. Iv looked all over and I cant find anything so it looks like I may have to hack a ast_set_var into app_dial or chan_sip 1.4 has the CHANNEL function: pbxlab-01*CLI> show function CHANNEL pbxlab-01*CLI> -= Info about function 'CHANNEL' =- [Syntax] CHANNEL(item) [Synopsis] Gets/sets various pieces of information about the channel. [Description] Gets/set various pieces of information about the channel. Standard items (provided by all channel technologies) are: R/O audioreadformat format currently being read R/O audionativeformat format used natively for audio R/O audiowriteformat format currently being written R/W callgroup call groups for call pickup R/O channeltype technology used for channel R/W language language for sounds played R/W musicclass class (from musiconhold.conf ) for hold music R/W rxgain set rxgain level on channel drivers that support it R/O state state for channel R/W tonezone zone for indications played R/W txgain set txgain level on channel drivers that support it R/O videonativeformat format used natively for video When I put this in a dialplan with NoOps and called channel macros, I can kind of get what you're describing: [from-external-pbxtel] exten => 491,1,NoOp(${CHANNEL(audioreadformat)}) exten => 491,n,NoOp(${CHANNEL(audiowriteformat)}) exten => 491,n,NoOp(${CHANNEL(audionativeformat)}) exten => 491,n,Dial(SIP/491,20,M(logcodec)) exten => 491,n,Hangup [macro-logcodec] exten => s,1,NoOp(${CHANNEL(audioreadformat)}) exten => s,n,NoOp(${CHANNEL(audiowriteformat)}) exten => s,n,NoOp(${CHANNEL(audionativeformat)}) Console output is: -- Executing [ [EMAIL PROTECTED]:1] NoOp("IAX2/pbxtel-01-5", "ulaw") in new stack -- Executing [EMAIL PROTECTED]:2] NoOp("IAX2/pbxtel-01-5", "ulaw") in new stack -- Executing [ [EMAIL PROTECTED]:3] NoOp("IAX2/pbxtel-01-5", "ulaw") in new stack -- Executing [EMAIL PROTECTED]:4] Dial("IAX2/pbxtel-01-5", "SIP/491|20|M(logcodec)") in new stack -- Called 491 -- SIP/491-0a16d1c0 is ringing -- SIP/491-0a16d1c0 answered IAX2/pbxtel-01-5 -- Executing [EMAIL PROTECTED]:1] NoOp("SIP/491-0a16d1c0", "slin") in new stack -- Executing [EMAIL PROTECTED]:2] NoOp("SIP/491-0a16d1c0", "slin") in new stack -- Executing [EMAIL PROTECTED]:3] NoOp("SIP/491-0a16d1c0", "gsm") in new stack == Spawn extension (from-external-pbxtel, 491, 4) exited non-zero on 'IAX2/pbxtel-01-5' -- Hungup 'IAX2/pbxtel-01-5' This is a call coming in as ulaw over IAX2, then going to a SIP softphone configured for only gsm. Hope that helps. -- j. ______________________________________________________________________ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email ______________________________________________________________________
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