search for: wima

Displaying 11 results from an estimated 11 matches for "wima".

Did you mean: wimax
2009 May 21
2
MeetMe not working with GSM codec?
...k 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 [10000] type=friend secret=test host=dynamic nat=yes -------------------------- ----- extensions.conf: ----- [common] exten => 501,1,MeetMe(12,MI) exten => 501,n,Hangup() exten => i,1,Hangup() exten => h,1,Hangup() exten => t,1,Hangup() ------------------------------------...
2011 May 19
6
ConfBridge - Failed to find a bridge technology to satisfy capabilities
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
2009 Jan 15
0
Job as doctorial candidate
...er.koemm at ku-eichstaett.de LSQM - Lehrstuhl f?r Statistik und Quantitative Methoden Prof. Dr. Ulrich K?sters Auf der Schanz 49 85049 Ingolstadt Fon: +49 841 937-1847 Fax: +49 841 937-1965 -------------- next part -------------- A non-text attachment was scrubbed... Name: Ausschreibung-WiMa-Statistik-www ku-eichstaett.pdf Type: application/pdf Size: 14970 bytes Desc: not available URL: <https://stat.ethz.ch/pipermail/r-help/attachments/20090115/079efba3/attachment-0002.pdf>
2009 Mar 15
1
X-Asterisk-HangupCause - how to disable this?
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
2009 May 17
1
Capture "Server" header in SIP reply.
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:
2009 Jun 13
1
Dial with r option doesn't use 'ring' tone as defined in indications.conf
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
2009 Apr 16
7
How to send "404 Not found" SIP reply?
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
2009 May 17
1
SHARED() variables and <ZOMBIE> channel
Hi, I am using SHARED() function to push destination channel info (i.e. audio codec) into "source" channel, in order to record into a customer CDR field. My dialplan looks like: [default] exten => _X.,1,Set(_X-SRC_CHANNEL=${CHANNEL}) exten => _X.,n,Dial(SIP/user at domain.net,30,M(getCalledInfo)) exten => h,1,Set(CDR(DST_CODEC)=${SHARED(X-DST-CODEC,${CHANNEL})})
2009 May 22
1
Can't get G.726 to work.
Hi, I have both codec_g726.so and format_g726.so loaded: root at test:~# asterisk -r -x "module show" | grep 726 codec_g726.so ITU G.726-32kbps G726 Transcoder 0 format_g726.so Raw G.726 (16/24/32/40kbps) data 0 But when I try to dial into Asterisk with Twinkle softphone using G.726 codec: INVITE ..... [SIP headers omitted] v=0
2009 May 26
0
CDR after SIP blind transfer.
Hi, I can't get Asterisk to save CDRs for calls transferred via SIP blind transfer. My extensions.conf: [globals] __TRANSFER_CONTEXT = transfer [common] exten => 123,1,Playback(demo-congrats) exten => 123,n,Hangup() exten => _0X.,1,Dial(SIP/${EXTEN}@PSTN-GW,60) exten => _0X.,n,Hangup() exten => i,1,Hangup() exten => h,1,Hangup() exten => t,1,Hangup() [transfer] exten
2013 Mar 27
1
send_meta() called with NULL pointer!
Hi, I am running tinc (1.0.19) on a cluster of around 40 servers (Xen virtual machines in two geographical locations). Overall it works very well, however yesterday it crashed on two machines in odd way. I was still able to ping VPN IPs of other hosts from the two affected machines but "vpn" interface was not visible when I run ifconfig. There was also no "tincd" process