similar to: SHARED() variables and <ZOMBIE> channel

Hello, for your consideration. The following patch moves the channel_pos() function from within the #if !defined(DISABLE_FLOAT_API). This change is required when compiling with FIXED_POINT and DISABLE_FLOAT_API defined. #### ### diff --git a/src/opus_multistream_encoder.c b/src/opus_multistream_encoder.c index 3efab53..6f3eb53 100644 --- a/src/opus_multistream_encoder.c +++

Apparently there are subtle diferences between meaning of MeetmeJoin event depending on channel type. Problem is: after originating a call from channel to MeetMe room i.e.: [meetme] exten => 1,1,Answer exten => 1,2,Meetme(kolejka|dqM) than: Context: meetme Exten: 1 Priority: 1 ActionID: 1077925740-00000004 Timeout: 5000 Action: Originate Async: true Channel: somechannel I get eventually

Hi, I'm trying to access audionativeformat / other codec variables in the hangup handler of a call (with ${CHANNEL(audioreadformat)}), but I get no response. Also 'core show channel ...' doesn't list those variables. Are they always set by asterisk, or only in some scenarios? It's a simple SIP-SIP call with audio passing through asterisk, same codecs on both sides. I see that

Hi, Is there a variable that contains the Sip Peer name? I was using ${CALLERID(num)} for outgoing calls, but when a call is being transferred, that variable contains something else. I need a variable that is always set to the SIP Peer's name. Thanks Dan -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Friends, Hope you would help me out here, I have searched the asterisk user list for hours and also read the readme and test files that comes with the driver. I need a very simple scenario. I have SIP clients and want to use oh323 to dial out to PSTN using a h323 gateway. a)If I set the extention.conf like this: exten => _87.,1,Dial(OH323/16.52.153.206) oh323 dials out (I can ring a

Sorry if this is an obvious question and perhaps my Google foo isn't right on this one: I have calls coming into an Asterisk server that may be using 2 different codecs. I am recording audio in both cases but the challenge is knowing which codec was negotiated at call setup. I need to pass the proper format to the record command as the codecs cannot be transcoded and are only supported for

What is the current best practice to avoid transcoding on an outgoing call to a party whose codec preference is not known in advance? In other words, incoming calls are easy since codecs are negotiated from least-known (the remote party) to most-known (my endpoint) and my codecs can simply be preferred accordingly to match the remote. Outbound calls seem harder. Our endpoints always negotiate

Hi, I am not sure if I am doing something wrong, but I can't get MeetMe to work with GSM codec (Asterisk 1.6.1 SVN r190371). My config files below: ---- sip.conf: ---- [general] context=common canreinvite=no bindport=5060 bindaddr=78.105.1.127 disallow=all allow=alaw allow=gsm rtptimeout=600 rtpholdtimeout=3600 rtpkeepalive=30 nat=no jbenable=yes tcpenable=no realm=dev-sip.wima.co.uk

Thank You . But now i get solved with that error since I had some mistakes in installing googletts.agi Now when calling from my softphone i have written dialplan with an AGI script to convert from text to speech. It get executed without error but there is no sound getting played. My output, == Using SIP RTP CoS mark 5 -- Executing [1310 at Client-dial-Menu:1]

Hi, I am trying to send "404 Not found" reply, without any luck with the following: exten => 555,1,Playback(you-dialed-wrong-number,noanswer) exten => 555,n,Playback(check-number-dial-again,noanswer) exten => 555,n,Congestion() However the above results in "500 Service Unavailable" being send out. What would be the correct application/function to generate "404

Hi, I am trying to use ConfBridge application, but it throws "Failed to find a bridge technology to satisfy capabilities 0x4 (ulaw)" error. Please see console output below. -- Executing [501 at services:9] ConfBridge("SIP/OpenSER-00000005", "1001") in new stack [May 19 13:36:05] DEBUG[7452]: app_confbridge.c:404 join_conference_bridge: Trying to find conference

Hi, I am trying to find a way to add the following info in CDRs (with asterisk 1.4.23.1): 1. Codec used 2. RTP QoS statistics 3. RTP IP of remote host 4. For answered calls, the peer that requested to end the conversation I have managed to get 1 and 2 for the caller, like that: exten => h,1,Set(CDR(userfield)=RIP=${SIPCHANINFO(recvip)}

Hi, Is there any way to tell Asterisk not to generate additional headers like: X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 I can't find any relevant option in sip.conf file :-( Thanks for help. Chris

Hi, I am trying to capture "Server" header in a 200 OK reply message. My idea was to use Dail(SIP/user at domain,30,M(GetOtherPartyInfo)), and inside of GetOtherPartyInfo macro use SIP_HEADER function. For example: [default] exten => _X.,1,Dial(SIP/user at domain,30,M(GetOtherPartyInfo)) exten => _X.,n,Hangup() [macro-GetOtherPartyInfo] exten => s,1,NoOp(SIP Server:

Hi, Just noticed Asterisk is not playing 'ring' tone as defined in indications.conf when Dial command is used with 'r' option. For example: [test] exten => 123,1,PlayTones(ring) exten => 123,n,Wait(5) exten => 123,n,Playback(demo-congrats) exten => 123,n,Hangup() exten => 321,1,Dail(LOCAL/123 at test/n,60,r) When I now dial with a SIP phone - 123 I can hear nice

There is always two zombie process in my system.I don't kill them.Is there the same problem in your system. PID TTY STAT TIME COMMAND 2444 ? Z 0:00 [Xsession] <defunct> 2551 ? Z 0:00 [netstat] <defunct> -------------- next part -------------- An HTML attachment was scrubbed... URL:

At the moment xm shutdown --all --wait hangs with a zombie domain. This means it hangs xend stop on dom0 shutdown.. I see no use for this behaviour, it would be better to skip Zombie-Domain-* domains. _______________________________________________ Xen-devel mailing list Xen-devel@lists.xensource.com http://lists.xensource.com/xen-devel

on sip show channels I do get a lot of entrys like 192.168.1.47 11 07ba5a490b3 00102/00000 unkn No Init: INVITE 192.168.1.47 11 19090f115b8 00102/00000 unkn No Init: INVITE 192.168.1.47 11 7d8b8fde46f 00102/00000 unkn No Init: INVITE How do they appear? How can they be removed? "core show channels" does not list them. Elmar

Dear listers, I'm facing a puzzling situation with Digium TDM2400 card (12 FXO / 12 FXS). Once a day or so we detect 1 or 2 zombie FXO channels. These can be either outbound or inbound calls. I thought this could be related to obsolete DAHDI or Asterisk versions, so I upgraded to 2.4.0 and 1.6.2.15 respectively (OS: Ubuntu 10.04 64 bits). To no avail; the zombie channels keep showing up.

I've looked it up on the database here and have tried the suggests there such as setting the DXlevel to 81 and adding zps then setting it to load under windows 98 but still it continues to crash on me. Not completely clued up in regards to what information I can post. Is there a log file I can attach maybe submit my experience of this game on the current version of Ubuntu ? In regards to