search for: app_page

I'm trying to use a simple page function. It starts a MeetMe conference with the devices I've listed, but the devices hang up after 3-5 seconds. After doing some research I found this was a problem, and I needed to remove a (5) from app_page.c Well, my app_page.c didn't have the (5). I did make clean; make install again just in case I had some weird compiled version installed that had the (5) in it. After compiling I restarted the asterisk service and tried paging again and still had the same problem. In the CLI I get the fol...

...rototypes -Wmissing-prototypes -Wmissing-declarations -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -DMAKE_VALGRIND_HAPPY -I/usr/local/include -L/usr/local/lib -I/usr/local/include/spandsp -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer -I/usr/local/include -L/usr/local/lib -fPIC -c -o app_page.o app_page.c cc -shared -Xlinker -x -o app_page.so app_page.o gmake[1]: *** No rule to make target `app_rxfax.o', needed by `app_rxfax.so'. Stop. gmake[1]: Leaving directory `/usr/ports/net/asterisk/work/asterisk-1.2.13/apps' gmake: *** [subdirs] Error 1 *** Error code 2 Anyone have...

using app_page on asterisk 1.6.1.6, as documented, the 'q' option only determines if the caller is sent a 'beep' tone when conferencing. is there a way (existing or someone sending me a patch) to also make app_page beep all of the extensions being called? someone adding an 'a' (announc...

...pscan.o app_zapscan.c gcc -shared -Xlinker -x -o app_zapscan.so app_zapscan.o gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer -fPIC -c -o app_page.o app_page.c gcc -shared -Xlinker -x -o app_page.so app_page.o gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer -fPIC -c -o app_curl.o app...

...om features. I have the module installed and I am able to successfully intercom with individual phones using *80xxx (extension number) but if I create a paging group it does not work. I receive a message that it is an invalid conference number. I did some digging and saw that I did not have the app_page.so file on my system - I have no idea why. I added this (copied from another instance I have running) but this did not fix the problem. I also noticed other *.so progs missing related to dahdi. What other modules need to be installed and loaded for paging to work properly? Thanks, jellydog...

...stem in intercom mode with about 90 phones overnight (624mins, CPU utilisation was running much higher than normal until i used the meetme kick <channel> all command, which restored everything back to normal). Would the Asterisk-PBX command Page() honour it (i had a quick look at the app_page.c source, nothing stood out to me)? i.e. ... n,Set(TIMEOUT(absolute)=60) n,Page(SIP/701&SIP/702&SIP/703&SIP/704&SIP/705&SIP/706&SIP/707&SIP/ 709&SIP/710|q) ... using; asterisk-pbx v1.2.7.1

I am trying to page my Grandstream GXP-2000 phones and keep getting the error message: Jan 27 12:55:04 WARNING[30401]: app_page.c:183 page_exec: Incomplete destination '' supplied. How can I fix this error? The two contexts below do either one-way paging or two-way paging to all Grandstream phones in a list. [One_Way_Page_GROUP] ; one to many page exten => **1,1,VERBOSE("ENTERING [default] **1 in exten...

I've been able to page to a specific phone (intercom type of thing), but I'd like to have a macro or agi that pages all phones but first checks if their on the phone. It looked like there used to be a pageall.agi type of script on the wiki, but that link isn't valid anymore. Does anyone have that script, or something else that would work? I would just do SIP/1000&SIP/1001, but

Hello, I wonder if there is a way to build my own asterisk application (let us say apps/app_myappl.c), and to launch other existing applications from it (for example, doing an apps/app_dial.c, or others). Could someone highlight me on that? thx Pirlouwi. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi list, i have and asterisk into a Pentium IV Server with 1GB of RAM about 75 Polycom Phones, one E1 for incoming calls. We have program a page system with the page command and the auto answer funtion of polycom. We have detect via diaplan if the phone isn't in call we place the call. All this via Macro. But in the our that they are not many calls. So much user that can be page.. The

I just upgraded to Asterisk 1.2.7 from 1.2.5. Page() is behaving differently. I'm getting an error - Incomplete destination '' supplied. -- Executing Page("SIP/2944093-5999", "SIP/3254107&SIP/3254105|") in new stack Apr 13 11:06:11 WARNING[9294]: app_page.c:193 page_exec: Incomplete destination '' supplied. -- Playing 'beep' (language 'en') -- Created MeetMe conference 1023 for conference '1592290043d' -- Hungup 'Zap/pseudo-1092580194' == Spawn extension (oneeighty_start, 1000, 1) exited non-zero...

...y info for both Zaptel and DAHDI in 1.4 and be sure to run dahdi_test in 1.6.x and trunk instead of zttest. - Closes issue 14132. Reported by and patch contributed by: dsedivec * Don't overflow when paging more than 128 extensions. The number of available slots for calls in app_page was hardcoded to 128. Proper bounds checking was not in place to enforce this limit, so if more than 128 extensions were passed to the Page() app, Asterisk would crash. This patch instead dynamically allocates memory for the ast_dial structures and removes the (non-functional) arbit...

Howdy, I am proud to announce that after a good bit of development, community feedback, testing, and code review, the brand new ConfBridge application has been officially merged into Asterisk Trunk!!! http://svnview.digium.com/svn/asterisk?view=revision&revision=314598 If you are already familiar with ConfBridge from Asterisk 1.6.X and 1.8, forget everything you know. This is a completely

I'm building a paging system composed of roughly 10 switches in daisy chain, with an embedded box with a speaker and a microphone for each switch. The embedded box runs my software. I need the system to be resilient to any network partition, so that anyone can send announces from any mic to all the reachable clients. I'd need also to page a subset of all the speakers. I'm

Hi asterisk-users, We are using Asterisk 1.2.12.1, and are trying to use the Page application. It seems to work but after approx 4-5 seconds the call is hung up. The dialplan code look like this: exten => _*2XX,1,AGI(get-paging-devices.agi,${EXTEN:2}) exten => _*2XX,n,GotoIf($[ "${PAGING_DEVICES}" = "invalid" ]?i,1) exten => _*2XX,n,SIPAddHeader(Call-Info:

..._exec.so app_externalivr.so app_festival.so app_flash.so app_followme.so app_forkcdr.so app_getcpeid.so app_hasnewvoicemail.so app_ices.so app_image.so app_lookupblacklist.so app_lookupcidname.so app_macro.so app_meetme.so app_milliwatt.so app_mixmonitor.so app_morsecode.so app_mp3.so app_nbscat.so app_page.so app_parkandannounce.so app_playback.so app_privacy.so app_queue.so app_random.so app_read.so app_readfile.so app_realtime.so app_record.so app_sayunixtime.so app_senddtmf.so app_sendtext.so app_setcallerid.so app_setcdruserfield.so app_settransfercapability.so app_sms.so app_softhangup.so app_sp...