similar to: MeetMe not working with GSM codec?

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 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, 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

Hi All, I'm still having trouble trying to figure out if it is possible to define (in the dial plan) a context for meetme? I'm using 1.4.4 with dialplan logic of: exten => 123,1,Meetme(,Msa,) This defaults to conferences defined within the rooms context of meetme.conf Is it possible to specify another context as with voicemail? Or can any one think of another

I have moved from dovecot 1.x to 2.x and I have big problem with subfolders. When I'm moving subfolder with other subfolders is moving only main subfolder, without subfolders, example : mail-storage-1 /var/vmail/home/adamskitest/mdbox/mailboxes # find | egrep -e "janusz|jarek" ./jarek ./jarek/dbox-Mails ./jarek/dbox-Mails/dovecot.index.log ./jarek/jarek2 ./jarek/jarek2/dbox-Mails

We recently updated our Asterisk (1.4.24) box from Zaptel (1.4.12.1) to Dahdi (2.1.0.4). Everything seemed to go smooth with the exception of meetme. Meetme seems to not be able to find a zap channel for conferencing. We use voice introductions in our conference bridge and it seems to break that feature. The error from the console is.... # app_meetme.c:2593 find_conf: No Zap channel available for

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

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})})

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

I am having trouble getting Google Chrome to accept a WebRTC call coming from Asterisk, even though Firefox can (now) accept the same call without issue. My setup is as follows: Server: CentOS 7 x86_64 (Elastix 4 RC) with IP: 10.1.0.4 192.168.5.146 asterisk-11.21.0 patched to work around https://issues.asterisk.org/jira/browse/ASTERISK-25659 openssl-1.0.1e-51.el7_2.2.x86_64 [root at elx4 ~]#

> > 1. Do I need to activate jbenable in sip.conf? Or is it enough to call > > the JITTERBUFFER function? > > You only need to use the JITTERBUFFER function. > > The jbenable option will enable a jitter buffer on every channel > created for that peer (or, if global, for every peer in the system). > Depending on the version of Asterisk, it will also place the

Hello! I am going to use the JITTERBUFFER function in a SIP (and local channels) only setup, but have some questions of how to use it: 1. Do I need to activate jbenable in sip.conf? Or is it enough to call the JITTERBUFFER function? 2. What is the preferred way to invoke this function? Say I have channel A which is not in need of buffering, while channel B do need it. If A

I updated Asterisk to 1.6.2.7 and now the user introduction in the meetme application is no longer working: [May 27 09:26:51] WARNING[2407]: channel.c:4034 ast_request: No channel type registered for 'DAHDI' -- Created MeetMe conference 1023 for conference '800' [May 27 09:26:51] WARNING[2407]: app_meetme.c:3640 find_conf: No DAHDI channel available for conference, user

Hi list! When I receive an incoming call from a SIP peer where I've configured disallow=all allow=alaw (and no other codec) I can see the following NOTICE on the console: Dropping incompatible voice frame SIP/peer07-0000007c of format ulaw since our native format has changed to (alaw) My question is: where can I change the native format from ulaw to alaw (or something else)? Is ulaw, as

Hello, When I receive a call to my CounterPath Bria from Asterisk 1.4.18.1 and I place the call on hold, the call is dropped after 30 seconds. It looks like there is no RTCP/RTP sent to the client from Asterisk while on hold (music on hold playing to caller) thus client disconnects the call. During this time, I get the following messages in the CLI: NOTICE[24194] rtp.c: Unknown RTP codec 126

Hello! I use asterisk with TE420 as PRI switch for two channels : ;panasonic uplink group=3 context=panasuplink ; relaxdtmf=yes ; immediate=yes rxgain=0.0 txgain=0.0 mohsuggest=default jbenable = no ; jbenable = yes ; jbmaxsize = 200 ; display_send=name_initial display_send=name

Hi, I have asterisk 1.4.16 behind a NAT-FW which is using a hosted SIP trunk for PSTN calling. Asterisk is configured to support nat with nat=yes in sip.conf. Now the hosted PSTN Gateway supports symmetric RTP and early media using 183 Session Progress. So If I call a PSTN number which has IVR message played before the call is connected (via 183), those media RTP packets do not reach the

We have a customer on a wireless connection that has very bad jitter. They can hear people fine, but people have a very hard time hearing them. They are connected via a SPA-2102. It is a SIP client going to a SIP trunk. Something like this in sip.conf [general] would be in effect for all SIP clients: jbenable = yes jbmaxsize = 150 jbresyncthreshold = 1000 jbimpl = fixed jblog = yes I only want