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How can I manage double doorphone/doorlock?

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It happens that doorlocks open and call clears normally , nice !

But I find a more elegant solution:

With the current one, once the “featurecode_doorsopener” dss key is pressed it’s like to call another extension, thus a transfer is initiated (person at the door get MOH).

Instead I created a /var/www/html/doorsopener.php script like this:

<?php
    shell_exec("curl -i 'http://admin:admin@192.168.10.30/cgi-bin/ConfigManApp.com?key=F_LOCK&code=*' && curl -i 'http://admin:admin@192.168.10.31/cgi-bin/ConfigManApp.com?key=F_LOCK&code=*'"); 
?>

associated to a Yealink phone “URL” dss key (http://pbx_ip/doorsopener.php).
While talking, once key is pressed doors are opened without loosing connection with waiting person.

Of course, as-is, it could have security issue…


How to check yesterdays channel count on particular time in asterisk

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Did you look at Reports >> CDR Reports?

You can have so many options to filter / group your results and you can export as csv file

How to check yesterdays channel count on particular time in asterisk

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Mysql can COUINT SELECTed data from the ‘cdr’ table in asteriskcdrdb

SELECT COUNT(uniqueid)  FROM  cdr WHERE calldate >= 'YEAR-MO-DA HR:MN:SE' AND  addtime(calldate,DURATION) <= 'YEAR-MO-DA HR:MN:SE'

YEAR-MO-DA HR:MN:SE is of course numbers to identify the point of interest

Chan-SCCP-B not working / installing

How to check yesterdays channel count on particular time in asterisk

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It depends on what the OP means by channel count. I read it as the number of channels concurrently open, but even if you want the cumulative number closed, that query will underestimate.

If you can get by with the cumulative number created, and there are not too many, there is an implementation detail of unique ID that will give you that, in that the part of unique ID after the dot is a serial number modulo some power of two. The other part is the time of creation.

I don’t think the number concurrently open is logged.

How to check yesterdays channel count on particular time in asterisk

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Only the OP can answer what he means by ‘channel count’ (@moussa854 ?)

I supplied a query to simply count the number of concurrent calls at a singular second in time that have been opened but not closed, If he wants absolute channels he can use a similar method on the ‘call event log’ (cel) table using eventtype and counting uniqueid’s but bearing in mind any linkedid , both of these records are committed on the state change of the channel.

The granularity is 1 second (asterisk not mysql) , so I guess you are correct the result might be either an over or under estimation.

Actually the mantissa of uniqueid is simply the number of ‘channels’ previously opened since Asterisk was last started ++ and the significant part is the number of seconds since the ‘epoch’ (Jan 1 00:00 1970 in Greenwich London) , which is why it is always a ‘uniqueid’,

No, the number of open channels is not logged, however Asterisk’s SNMP mib can show ‘the number of active channels’ as ‘asteriskChannels’ in a contemporaneous fashion if he needs it.

Asternic Does Not Fully load

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What error do you see in the apache logs?

Freepbx17 - running OK, rebooted, PHP out-of-memory error in GUI

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Freepbx seems to be running, but get error revery time i access the webpage

  1. if(extension_loaded(‘ionCube Loader’)){die(‘The file ‘.FILE." is corrupted.\n");}echo("\nScript error: the ".(($cli=(php_sapi_name()==‘cli’)) ?‘ionCube’:’ionCube’)." Loader for PHP needs to be installed.\n\nThe ionCube Loader is the industry standard PHP extension for running protected PHP code,\nand can usually be added easily to a PHP installation.\n\nFor Loaders please visit".($cli?“:\n\nhttps://get-loader.ioncube.com\n\nFor”:’ get-loader.ioncube.com and for’)." an instructional video please see".($cli?“:\n\nHome - PHP Encoder, protection, installer and performance tools from ionCube”:’ http://ioncu.be/LV ').“\n\n”);exit(199);
  2. ?>
  3. HR+cPmsZT0iZEaqjEcZXAFaErVISEguECyoVvxEufFHIP56SjO5CSKSrCaa6iOVajTZSG7XgmkUK
  4. +bUXvEW9+17J+BdwT6WV/63eaGw7XzqO/JvaY7txJesqr39lW0qNRFyNmMxBmyVNxWetfmX49CSD
  5. xE9LQYVAbq54zGjWAKGXBbUTDejjfj0Vt1UxOgcFHicWH+1s2P6/TYdhqnCQAyxuTaEYUPueA2tV
  6. gjChKiZlpsusT+krEUV+KqTz43cCIhH6BTTr42QqaKkBynriHdQTrOkaMxfeT9Q5Tai/xrGbn1jH
  7. Jka77JWI6uY9XFwLhw5oNxXsi/tj3ebFgQX9NSEyWaUwXoejO/T6b1Nj3YqtQMY2FsMOiZqmE83y
  8. 5AUxZQxRc39Rs07CcValf9XyB8VcLbkrJdPtS96YyzShUrgdd8je/X/ryta0p9SPNBsTPpJ0zcpT
  9. 0nSJDYVXLEbJFZd+JOfo9qRIYIDwhurzC7se/ObZILkyCMUKpVsW0RqN00bYE+MIlvjLqPCefoot
  10. PRWWTbgrUIzow4hZMbg3kNRvJwNZmIpmW2PQpNXdCbtlW2xYFkwJ98dKd5Ct2V2q6urvypa/MeA9
  11. lbs5ZyNEOpBqqTAvqLa5spMiBgnhfzIosSumtM3tTCD+EHezwIZ/icCiMzLlo4LRXZTh8FilJmts
  12. 4bUZGWvGPstuN5OCm24/lOB2DlwqGs3ZGeE9SnjhGKTeLgWFQ33ecq2s5Ha+czoAwxCQcOqgDOE0
  13. rLf4HC5jgvaCHdcMa9PUrvkwmGyKSYZ44cxikz7j0pETgVHDkR6hfpryqKsPFJ8QgGVZ9DsDJa+A
  14. keuT3eozsJIoVELO08wBGaY5UdH09m4wex6l26Uynm1Pxk9aoE/oUchEkCmsZF4419v+5h3vwJSI
  15. K5UQXjoGDR+qjzA1Hs+lrHzpHOW8S+vXpJZPc5+zYEGZhm7oNLMC5IWaG1xa8wv9iK3W8SNj7l5T
  16. OhKTJoAkq1/rNnctLxOummg4V4QGgzz5pytu8F3VaOqeogXvdn0jvKMSg9O3JKiPufnGWQVaNf2n
  17. xRxWTUTNyfP3CSdRbxvx6ke7XkDI1mc1O31xcTduLxzVWuDuwXlPrVhbxsOrWmibfAbL9uISmBwW
  18. +u529K9aizT/uns5lBypxUdK855tuoVLne8X7aGsz4EUG+witNhh6VHy45S/+2bMRMTUR34GuJrx
  19. mYeW/GY9hdk7UIbywx/34NlMrR3nwmN5C3haVHgBI98hFf+uJgsnvVKXILkVR8MngK8UndxbHU+0
  20. 67NImL4VGTHHJLWArfdQJkXsg7e7QgT+kT49xjs1uuaMeRd0tbZcQGvoHhl3wNYJpuazKLUh/ZNB
  21. 32YdcQsbuPO7GFX7ZXpx1qAt25Rr1XLs7E/qxDWPoRyMC9I1GdOtnAe6urMPFKK566h2VEw3L7AI
  22. TZyO8901ObJyTFdxmtg3R4Iq3d1gXIO1KugRd+zidxKu7jYFyxs9E0ezqHYil8goBkK3oq9GBh4h
  23. rZKM9lILq3l8aZc7XFyCgRqLMIu0Jr5sgRpSpKAxXn95sdft936sj13dt9SPMR2qyI9FtvRaH3v+
  24. HTQKvhIxqfmeT0i+CpdvfES5t/FHtKuDLt9CBxKNgvp4cPYZ9UjDEBK4cGmPDqV9j8gNXIzzCgD3
  25. nNpTuGt95OSj1VH1p0zAdhYcbth/0vbURRxFx0bUKxw5EwFAocOzKICoaGXhzeibifYGf5dH3CrM
  26. wqHyAanvJLfkcqfasMqdmUZloJQALG77RSfaFaTB8BVbmMlLIaFD/tEWeqLDz2zm+bXU/eQA4DY0
  27. wF+t1Je6HoNzuuM7/0TUIi93z7pSNNOa5gnPGhCdBtutQVZszFUcChvrpFWtKo5xz8o9UixiaSYL
  28. yKwKJKlTgWTf0jVdLbVPqJjEdUd+6zykTxnn2DIlHWXXKgjxUCIsM75OZeim54SR/kznQnoLnrq5
  29. W/aG9I8Q25XnTFwlRvGsITpCT70DFUheLmKvuYQuLqAFeDPqT6JrTiEajElPOdZS7V+mkJuMGtSN
  30. egT0p8YMUt+XipYSjGo3kZABZeXxRTE9R5uvpSfh0YTgpx0WNNglWx4jzJZQq1H/ChyfqYe/Fl/s
  31. Qdmw/4z6XgIcwBEtfUg1h6KYpOBxhrctc0Qjci/bDbpuypqNdaWnB31adJ+0T+BLyEJFXXhLVVEd

Arguments

  1. “Allowed memory size of 536870912 bytes exhausted (tried to allocate 1281717103 bytes)”

How to check yesterdays channel count on particular time in asterisk

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Thank you for your input. I use the cdr table to get some information. I was trying to help . Will see what the op @kiranpk is trying to achieve

FreePBX- Voip.ms Inbound Route Configuration

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Assuming you use IP Auth, then yes, you would need to forward SIP and RTP ports used.

It is recommended to forward only from trusted sources and not publicly expose these.

Forward phone call to first available Stasis-based extension

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For anyone reading this: after trying a few solutions I realized I started with a completly wrong assumption. When using Statis, there is no “busy state”. One call reaches the Stasis App and it gets assigned a channel. If a second call reaches the Stasis App (and the first call is still ongoing) it simply gets another channel assigned.

So, any concurrency must be implemented at the level of the Stasis App.
One needs to ensure the (automated) answers are simply sent to the right channel back.
I hope that helps anyone in the future.

How to check yesterdays channel count on particular time in asterisk

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Sorry, I shoulda said @kiranpk, I was just pointing out how cdr/cel tables can more specifically filtered

Trying to convert to pjsip - trouble with Grandstream devices

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You’ll need to provide some packet captures or asterisk logs of the failing registrations for anyone to give you really any useful advice here, otherwise everyone will be just trying to guess what your problem actually is and you may have something miss configured and not realize it.

Freepbx17 - running OK, rebooted, PHP out-of-memory error in GUI

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Please try to restart the apache2 ie. systemctl restart apache2 .

Php error after clean install and activation

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Hi,

I just installed FreePBX 17 on Debian.

The script worked without any problems and I was able to open and use FreePBX until I activated it. After activating the installation and rebooting the machine, I received the following error:

php

Code kopieren

Whoops \ Exception \ ErrorException (E_ERROR)
Allowed memory size of 536870912 bytes exhausted (tried to allocate 1281717103 bytes)

File path: /var/www/html/admin/modules/sysadmin/functions.inc/Ioncube.php

php memory Limit is set to -1 so it should not be a Problem

I´m using an Lenovo Thinkcenter M710q, with a 4 Core CPU at 2,4 GHz and 24G of memory.

I formatted the machine and installed everything again, but I encountered the same result after activating.

I hope i´m not reopening a known problem, i just seach for it but wasn´t able to find it.

Best Regards
Chris


Php error after clean install and activation

Call busy from one number to extension

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Goodmorning,
when I call from a number and choose the extension via IVR, it always gives me BUSY, while if I choose others it works. I did other tests from another number and it makes me call that extension. until a few days ago it worked, then to do some tests from that number (I’ve done a lot) I think he banned me or put me on some blacklist and I’m always BUSY. I tried using the Blacklist module but nothing appears.
how to solve?
Thank you

Configurazione Trunk Irideos

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Buongiorno a tutti,
sperando di fare cosa gradita indico qui di seguito la conifgurazione del trunk SIP in caso di full-voip con Irideos. Ci ho sbattuto la testa per una settimana senza successo con PJSIP ma alla fine la configurazione seguente in un trunk chan_sip funziona all’istante:

Outgoing Peer Details
type=peer
insecure=port,invite
contex=inbound-trunk ;context for incoming call
srvlookup=no
host=vnpublic.irideos.it
port=5060
transport=udp
fromdomain=vncluster.irideos.it
username=user fornito da Irideos
secret=password fornita da Irideos
canreinvite=no
nat=no
disallow=all
allow=g729
allow=alaw
qualify=yes
dtmfmode=rfc2833
dnisontoheader=no
defaultexpiry=120
minexpiry=60
maxexpiry=3600

Incoming Register string
user fornito da Irideos@vncluster.irideos.it:password fornita da Irideos@[vnpublic.irideos.it:5060/user fornito da Irideos](http://vnpublic.irideos.it:5060/***user fornito da Irideos***)

One way audio inbound calls / no audio from remote

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Mine has a public IP address, and the ISP provided credentials that have no registration for both inbound and outbound calls. But when a call comes in, I see it but when picked both ends have no audio.
And when I do an outbound call, the extension instead calls itself. Is there any idea on this?

Tim business, freepbx e errore SIP/2.0 500 CSCF Server Internal Error 020350306 in uscita

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Buongiorno, abbiamo un account Tim Business, con due numeri associati. Abbiamo un setup con FREEPBX 16 e Asterisk 20.5.2

Abbiamo fatto la configurazione suggerita qui config e abbiamo lo stesso problema riportato qui:

https://community.freepbx.org/t/problemi-con-numero-aggiuntivo-tim-business-e-freepbx/97103/3

Randomicamente (da pochi minuti a poche ore) le chiamate che funzionano, iniziano a fallire in uscita, via tcpdump si vede un pacchetto ritornato dal server TIM riportante:

SIP/2.0 500 CSCF Server Internal Error 020350306

Abbiamo contattato la TIM (supporto a pagamento) che, pur avendola messa “spalle al muro”, mostrando l 'errore serverside, di fatto non ci ha aiutato, dicendo che anche se rilasciano i parametri VOIP “loro supportano connettività SIP solo su porte FXS”. Ci hanno proposto di passare al loro centralino proprietario (ovviamente a pagamento).

E’ chiaro che l’errore 500 sia serverside, ma sospetto che in qualche modo lo provochi Asterisk (o che i loro router abbiano dei settaggi di registrazione più robusti/aggressivi che lo aggirano, altrimenti avrebbero dei disservizi massivi per tutti i loro clienti che usano i normali router, cosa che non succede).

L’unico modo per risolvere provvisoriamente è disabilitare/riabilitare i trunk per farli riregistrare, oppure riavviare direttamente la macchina che fa girare il PBX.

L’errore lo otteniamo con diversi server Tim VOIP (provato i due ritornati dall’nslookup e un terzo “fuori zona” suggerito dal tecnico Tim). Qualcuno ha mai visto questo errore nelle sue prove, e ha dei suggerimenti di dei settaggi per aggirarlo/evitare di provocarlo?

O in alternativa cosa avete detto al support Tim a pagamento per convincerli a prendere in carico la questione? O sapete cosa è stato fatto per risolvere il problema?

Se non ne usciamo stiamo pensando di utilizzare un adattatore con due porte FXO, oppure cambiare direttamente provider telefonico passando al voip puro.
Grazie

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