search for: 8088

HI all, I'm new to R. Say I have a multi-layered list called newlist. ############ > str(newlist) List of 2 $ :List of 5 ..$ : num [1:8088] NA 464 482 535 557 ... ..$ : num [1:8088, 1:2] NA 464 482 535 557 ... ..$ : num [1:8088, 1:3] NA 464 482 535 557 ... ..$ : num [1:8088, 1:4] NA 464 482 535 557 ... ..$ : num [1:8088, 1:5] NA 464 482 535 557 ... $ :List of 3 ..$ : num [1:8088] NA NA 1386 464 482 ... ..$ : num [1:8088]...

Hello, I'm trying to have my first calls with WebRTC. My server has asterisk 13.7.0. I'm following the instructions from the wiki [1]. So I'm using [2] live demo from a Chrome navigator (v48) on Debian Jessie station. Whenever I type something like ws://123.123.123.123:8088/ws in Expert Mode form (see [1]), I'm getting this error : *2:SecurityError: Failed to construct 'WebSocket': An insecure WebSocket connection may not be initiated from a page loaded over HTTPS.* If I replace ws://123.123.123.123:8088/ws with wss://123.123.123.123:8088/ws, this error me...

Hi there, I just installed the GUI for Asterisk 1.4.4 and correctly set my settings but when I use my browser to access it, it gives me an error saying "Not Found. Nothing to see here, move along" with "asterisk" in the header and footer... anyone had this problemn before? greetz

...ing really smart. I get the SVN Edition of asterisk GUI interface, compiled and love to get it to run, what won't work. What am I doing wrong?! svn checkout http://svn.digium.com/svn/asterisk-gui/branches/2.0 make make checkconfig make install and If I open one of the URLs: http://localhost:8088/asterisk/static/config/cfgbasic.html http://127.0.0.1:8088/asterisk/static/config/cfgbasic.html I always get 404 not found! For any advises I would thank you. Here is the manager.conf and http.conf manager.conf: [general] displaysystemname = yes enabled = yes webenabled = yes port = 5038 httpt...

...ied to install debian as a new guest VM in paravirtualized mode on Xen the following error occurs. please help me to resolve the problem.. Unable to complete install ''<type ''exceptions.ValueError''> Invalid install location: Opening URL http://172.16.0.31:8088/debian1 failed.. Traceback (most recent call last):   File "/usr/share/virt-manager/virtManager/create.py", line 616, in do_install     dom = guest.start_install(False, meter = meter)   File "/usr/lib/python2.5/site-packages/virtinst/Guest.py", line 722, in start_ins...

...2] username=3002 secret=XXXXXXXXX host=dynamic type=friend context=test disallow=all allow=g729 ;allow=all ; Allow codecs in order of preference allow=ilbc allow=silk8 allow=gsm transport=ws I also modified the sipml5 library so that the URL looks like this: ws:// example.org:8088/ws with the /ws at the end, as instructed. Now, where I get confused is here: "You will need to change sipml5 to use http://<hostname or IP address of Asterisk>:8088/ws as the URL. WebSocket is only available on the /ws path." Did Joshua mean to say ws:// instead of http://? B...

..."--data" and "-H" Option as well as the "variables" section within the Body. I added the callerid function here as well as it is the sample in the asterisk wiki. curl -v -H "Content-Type: application/json" -u asterisk:asterisk -X POST " http://localhost:8088/ari/channels/newChannelId" <http://localhost:8088/ari/channels/1400609726.3/play?media=sound:hello-world> --data '{ "endpoint": "SIP/Alice", "variables": { "CALLERID(name)": "Alice" , "PJSIP_HEADER(add,P-Asserted-Identity":...

Should the ARI DELETE /playback/{playbackId} be able to stop a playback when a number is being played? Here is a test I am running. I am playing multiple portions (sounds and numbers). curl -v -u asterisk:asterisk -X POST http://localhost:8088/ari/channels/1596750578.46/play?media=sound:hello-world,sound:tt-monkeys,number:553,sound:demo-instruct,number:123,number:456,number:789 If I attempt to stop while any of the numbers portions is being played, it does not stop the playback. I hear a skip or hiccup, but it continues playing. When I...

...MI to ARI. Running into a small hiccup when I try to create (originate a call) with the caller id name and number I can pass the Name and Number if the name has no spaces in it and it shows up in my PhonerLite application. curl -v -u asterisk:asterisk -X POST http://asterisk:astersk at localhost:8088/ari/channels/mycallerid.1?endpoint=PJSIP/1003 at 1003&app=hello-world&extension=1000&context=mycontext&priority=1&channelId=mycallerid.1&formats=ulaw&timeout=30&callerId=Dan<291> However, when the caller id name has a space in it, I can't figure out how to...

...create (originate a call) with > the caller id name and number > > > > I can pass the Name and Number if the name has no spaces in it and it > shows up in my PhonerLite application. > > > > curl -v -u asterisk:asterisk -X POST > http://asterisk:astersk at localhost:8088/ari/channels/mycallerid.1?endpoint=PJSIP/1003 at 1003&app=hello-world&extension=1000&context=mycontext&priority=1&channelId=mycallerid.1&formats=ulaw&timeout=30&callerId=Dan<291 > > > > > > However, when the caller id name has a space in it, I ca...

Hi Dan, i did it wrong, sorry: curl -v -H "Content-Type: application/json" -u asterisk:asterisk -X POST " http://localhost:8088/ari/channels/newChannelId" <http://localhost:8088/ari/channels/1400609726.3/play?media=sound:hello-world> --data '{ "endpoint": "SIP/Alice", "variables": { "CALLERID(name)": "Alice" , "PJSIP_HEADER(add,P-Asserted-Identity)"...

...P) without the libata-tj patch. I'm just trying to determine the easiest route to go with minimal cost. I've found another enclosure that uses 4 mini-SAS connectors, but in addition to the extra $320 for the enclosure, I'd need to buy a $300 card and at least one $120 cable (either SFF-8088 to SFF-8470 or a straight SFF-8088). While this is for at home, I would consider this a "mission critical" device and don't want a flakey setup, which is what I have now...I've become an expert at recovering from a 2 drive failure in a 4 drive RAID5 setup using mdadm. Any ideas,...

Sonny Rajagopalan wrote: > Is there a specific place where I can set logger to log incoming TCP > segments from L4? > > $ netstat -tulpn | grep asterisk | grep LISTEN: > > tcp 0 0 0.0.0.0:8088 <http://0.0.0.0:8088> > 0.0.0.0:* LISTEN 10313/asterisk > tcp 0 0 0.0.0.0:5060 <http://0.0.0.0:5060> > 0.0.0.0:* LISTEN 10313/asterisk > tcp 0 0 0.0.0.0:2000 <http://0.0.0.0:2000> > 0.0.0.0:*...

...20:47 asterisk.pem -rw------- 1 root root 161 nov. 18 20:46 ca.cfg -rw------- 1 root root 1781 nov. 18 20:46 ca.crt -rw------- 1 root root 3311 nov. 18 20:46 ca.key -rw------- 1 root root 124 nov. 18 20:46 tmp.cfg # cat /etc/asterisk/http.conf [general] enabled=yes bindaddr=0.0.0.0 bindport=8088 tlsenable=yes tlsbindaddr=0.0.0.0:8089 tlscertfile=/etc/asterisk/keys/asterisk.pem But, still I don't have any HTTPS server running: # asterisk -rx "http show status" HTTP Server Status: Prefix: Server: Asterisk/17.0.0 Server Enabled and Bound to 0.0.0.0:8088 Enabled URI's: /ht...

...; wrote: I have the ARI enabled on my Asterisk test box, and want to listen to all events. I can’t find the syntax to do that. Can I only listen to events related to a stasis app? I was hoping that a simple wscat command like this would show me all events: wscat -c "ws://localhost:8088/ari/events?api_key=asterisk:asterisk " This does not listen to all events by default. If you want to listen to everything you can pass subscribeAll=yes[1] like so: ws://localhost:8088/ari/events?api_key=asterisk:asterisk&subscribeAll=yes [1] https://wiki.asterisk.org/wiki/displ...

hi, i @asterisk - svn-ed asterisk from digium 1.6 - make install >> its running and i can access the CLI @gui then i -svned asterisk-gui from digium - installed - repointes apache /var/www/1234 >> /var/lib/asterisk/static_html >> now, i see the login box, but i dont have any credentials. tutorials are suggestion manager.conf, BUT I DONT HAVE that file, in fact /etc/asterisk is

...s openssl versions, https over TCP port 8089 initializes correctly with asterisk-11.7.0. After an upgrade to openssl[-devel]-1.0.1e-16.el6_5.4.x86_64 , I compiled asterisk-11.8.1 . When testing the exact same configuration, I noticed that TCP port 8089 was no longer listening, even though TCP port 8088 (standard unencrypted HTTP) was. After a patch coaxing some error message to be shown in the logs, I got the following: [Mar 27 10:25:47] DEBUG[10516] config.c: Parsing /etc/asterisk/acl.conf [Mar 27 10:25:47] VERBOSE[10516] config.c: == Parsing '/etc/asterisk/acl.conf': Found [Mar 27 1...

...bian Buster, I can't find a way to enable HTTPS. Asterisk is running as asterisk:asterisk: asterisk 11097 0.3 6.7 741352 67984 ? Ssl 17:53 0:06 /usr/sbin/asterisk -g -f -p -U asterisk # cat /etc/asterisk/http.conf [general] servername=Asterisk enabled=yes bindaddr=0.0.0.0 bindport=8088 tlsenable=yes tlsbindaddr=0.0.0.0:8089 tlscertfile=/etc/asterisk/keys/asterisk.pem ;tlsprivatekey=keys/asterisk.key # ls -lR /etc/asterisk/keys /etc/asterisk/keys: total 32 -rw-rw-r-- 1 asterisk asterisk 1229 janv. 6 16:00 asterisk.crt -rw-rw-r-- 1 asterisk asterisk 586 janv. 6 15:59 asterisk.c...

...MI to ARI. Running into a small hiccup when I try to create (originate a call) with the caller id name and number I can pass the Name and Number if the name has no spaces in it and it shows up in my PhonerLite application. curl -v -u asterisk:asterisk -X POST http://asterisk:astersk at localhost:8088/ari/channels/mycallerid.1?endpoint=PJSIP/1003 at 1003&app=hello-world&extension=1000&context=mycontext&priority=1&channelId=mycallerid.1&formats=ulaw&timeout=30&callerId=Dan<291<http://asterisk:astersk at localhost:8088/ari/channels/mycallerid.1?endpoint=PJSIP/...

...rancis Gerund wrote: <snip> > And FWIW, I do like, and use the CLI all the time. That's how I learned - using MS-DOS 3.2 on a 386sx box with 360k floppy drives. And 512k ram - what luxury! > > : ) You leaned on a 386 with (2?) floppy drives? I had to make do with DOS 3.0 on an 8088 w/ 2 floppy drives... and let me tell you how much fun it was to compile (I kid you not) basica, esp. with an encrypted library.... mark