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Hello everybody, and first of all sorry for my poor English. I'm having trouble with Quadbri installed on Asterisk 1.2.17-BRIstuffed-0.3.0-PRE-1y-e. Everything is working fine, except calling to switched off or "out of coverage" cell phones. In this case I have to wait 40 seconds until Asterisk tell me that "all circuits are busy now" instead of receive cell phone

Dear All, I have an asterisk server version 1.2.12.1 along with trixbox and I am having this nasty problem, I have a TE200P and have an E1 pri attached to it and zttool says it's OK, I have configured the whole 31 channels into one group as follow: Zapata-auto.conf: callerid=asreceived signalling=pri_cpe switchtype=euroisdn context=from-zaptel group=0 channel=>1-15,17-31

Every so often, and it seems that it happens only when a call is in progress, all 24 Zap channels get reset. All channels are opened and then timeout. This causes the in-progress calls to terminate. There are no corresponding Red/Yellow alarms on wither the PBX or Asterisk although we do receive a fair amount of Blue Alarms. The Asterisk server is connected to a legacy PBX through a Digium

hello list i have an issue related to outbound calls i can contact all the number except on number given by our provider in trunk the issue just when i configure my trunk in our server but when i configure the trunk directly in x-lite i can contact this number without issue below the cli == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [0149xxxxxx at

I am using asterisk version 1.2.6 with Zaptel version 1.2.5. I have a POTs line coming into a Digium TDM01B. It appears to not be getting CID at all. If I hook up a POTS phone to the line CID comes through fine. Inbound and outbound calls work fine but there is just no CID on inbound for this channel.The incoming route for the channel is Zaptel Channel 0. No DID or CID settings applied. My IP

Hi All, I have searched the various post and not able to find the solution. I am using Asterisk 1.4.21.2 for outgoing calls. Earlier i used ZAP trunk and it works fine. Now i need to move to SIP trunk and configured the same. When i try from softphone i got error as "Call rejected" and in the asterisk i got error as

Really stupid question, but have you wrapped this all in an App class? If you don''t have a wxApp subclass driving your applications then all sorts of badness would happen. > I''m writing a custom little application for my boss''s high school > football team to track there stats play by play. > > My problem is that when I start the application, it

I am trying to use the clusterer as stated in the README ( plugin is running well ...) and I got an error when displaying the map... addDescriptionToMarker is not defined (no name)()22 (line 169) (no name)()ym4r-gm.js (line 67) [Break on this error] map.addOverlay(new GMarker(new GLatLng (47.7377071331,-2.9257965088),{title : "aa... here is the generated script... <script

trying to use a Clusterer (other capabilities are running well....) I got an error : addDescriptionToMarker is not defined => line 169 my controller.. clusterer = Clusterer.new(@my_markers, :max_visible_markers => 2, :grid_size => 10, :min_markers_per_cluster => 2, :max_lines_per_info_box => 10, :icon => icons[2])

I have call center PCs that switch between an IBEAM SIP softphone and a NEBU IAX softphone (for reasons that aren't germane here). The SIP softphones work fine, but the IAX softphones get a fast busy unless I give them an IAX trunk to use, instead of the PRI trunks that all the other phones are using. I am using Asterisk 1.2.3. svn rev 47264. I've appended a sample call trace. The

i have posted this question couple of times and never really got any hits i wasn't able to provide any debug info Connected to Asterisk 1.6.0.21 currently running on phoneserver (pid = 3309) Verbosity is at least 4 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 == Using SIP VRTP TOS bits 136 == Using SIP VRTP CoS mark 6 == Extension Changed 117[ext-local] new

Hi, I am writing a small doc for people new to Xen. to be able to compile Xen from sources on Ubuntu. Please suggest some improvements for this how to or mistakes that you can point out. There is a technique of virtualization known as Xen and a hypervisor known as Xen. Like you have Firefox,Open Office,VLC and other softwares on your computer similary there is a software known as Xen.

thank you for your response below the asterisk -vvvr Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [0176XXXXXX at from-internal:1] Macro("SIP/300-00000192", "user-callerid,LIMIT,EXTERNAL,") in new stack -- Executing [s at macro-user-callerid:1] Set("SIP/300-00000192", "TOUCH_MONITOR=1427481319.470") in new stack --

I have a Asterisk @home ver 1.0 running with a TDMB11 card. Several sip extensions and a regular phone connected to the box. All routing works fine from the regular phone connected to the box, whether its going to FWD, broadvoice or the PSTN. The problem I am experiencing comes from making calls from the sip phones. They get routed correctly to the sip and iax trunks but when making calls

I hope this is the right place to submit libao patches -- the freshmeat page points to ogg vorbis as the homepage for libao. Currently, at least on my system (SB Live Value!, ALSA .9 branch), stuff going to alsa sounds *awful* because of too-small buffers. mpg321 and similar programs are unusable, and none of them seem to want to let you set the buffer size. This patch lets an environment

I am making some assumptions, but assuming the 217.195.xx.xxx is your provider, you are getting this back from them: "Got SIP response 556 "No address found" back from 217.195.xx.xxx:5060" Are you sure that "0033149xxxxxx" is the format the provider is expecting? You might try enabling SIP debug on the 217.195.xx.xx IP and seeing what the INVITE looks like, but

Hi, I use: Trixbox-2.2.4 FreePBX-2.3.1.0 Asterisk-1.2.17 BRIstuffed-0.3.0-PRE-1y-e Zaptel-1.2.19 ..with two ISDN cards, often but occasionally the dial out failed but is possible to receive external call. My zapata.conf conf is: [trunkgroups] [channels] language=it context=from-pstn signalling=bri_cpe_ptmp rxwink=300 pridialplan=unknown prilocaldialplan=local switchtype=euroisdn

Hi there, I've a training dataset and a test dataset. My aim is to visually allocate the test data within the calibrated space reassembled by the PC's of the training data set, furthermore to keep the training data set coordinates fixed, so they can serve as ruler for measurement for additional test datasets coming up. Please find a minimum working example using the wine dataset below.

Request: An additional hook in plot.new() that is called prior to the call to .Internal(plot.new()). Reason: To allow the hook to set up or modify a graphics device that the new plot will appear in. The code change needed for this is simple - just 4 new lines of R code in src/library/graphics/R/plot.R:plot.new() Current definition of plot.new() in src/library/graphics/R/plot.R: plot.new <-

Hi, i got Ubuntu 6.06 installed and theres a problem with asterisk. I've sucessfully installed it with the command: #apt-get install asterisk Then after installing FreePBX i get this error when restarting asterisk: root@hernandezz-laptop:/home/hernandezz# asterisk -rvvvvvvvvvv Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) After looking at the logs i