API for Third-Party application to make calls

April 16, 2020, 5:01 am

Hello,

We are a portuguese IT entreprise that inplements Asterisk systems in our costumers.

We need to implement a new PBX system on one of our custumer, but have the requirement that can initialize a call from a third-party application. In the actual asterisk PBX client (but old version) we use a webService (on the PBX) to make a callBack to extension (desktop Phone) and automaticaly make the call to the destination.

Do FreePBX have webServices (or other funcionality) that can permit this possibility?

Thank you.

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Cisco x-cisco- asterisk

April 16, 2020, 6:44 am

≫ Next: Upgrade to 14.0.22? Restapps?

≪ Previous: API for Third-Party application to make calls

@jcadman wrote:

I understand that this is asterisk however I am trying to get the softkeys on my Cisco 8841 IP Phones and have seen a website http://usecallmanager.nz/sippeer-options.html. An issue is that it doesn’t specify where the code should go or if a new context should be used. Currently, all phones are on from-internal and adding the codes to the extensions.conf. Has anyone know where the code should be?

I have patched asterisk 13.32.0 so that the Cisco 8841 IP Phones will work with asterisk.

Please do not say that this is not recommended as I understand that many people have got this to work.

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Upgrade to 14.0.22? Restapps?

April 16, 2020, 1:04 pm

≫ Next: Asterisk hanging with channel.c: Exceptionally long voice queue length queuing to Local using PRI

≪ Previous: Cisco x-cisco- asterisk

@pbx_noob wrote:

Just curious how to resolve this - I’ve upgraded as far as I can, it says “up to date” but it won’t enable because it isn’t up to date. Am I supposed to use the “rollback” to upgrade? Seems counterintuitive? Thanks

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Asterisk hanging with channel.c: Exceptionally long voice queue length queuing to Local using PRI

April 16, 2020, 1:55 pm

≫ Next: Start of system recording is sometimes cutoff

≪ Previous: Upgrade to 14.0.22? Restapps?

@dbaddorf wrote:

Hello!

I have had a customer system in place for about 1 1/2 years and seemingly all of a sudden it has hung three times in two days. Calls can’t be received or placed and even local phones don’t connect. “Core show channels” shows active calls even with the PRI disconnected.

When this condition happens I am seeing messages in the Asterisk log file like:

[2020-04-16 14:10:06] WARNING[22605][C-00000070] channel.c: Exceptionally long voice queue length queuing to Local/##########@from-internal-00000671;1

I have to restart Asterisk to get everything working again.

Here is the setup:

SNG7-PBX-64bit-1805-1, Release Date: May 2018, FreePBX 14 • Linux 7.5 • Asterisk 13.

Asterisk 13.22.0 built by mockbuild @ jenkins7 on a x86_64 running Linux on 2018-07-25 22:30:39 UTC

FreePBX Distro: 12.7.5-1807-1.sng7

Sangoma PRI card A101DE w/ Comcast PRI

The system hasn’t been updated since the install except a few weeks ago I updated the FreePBX Framework and the Core modules because follow-me confirm calls wasn’t working. I don’t understand how this could cause any stability issues with the underlying Asterisk.

Does anyone have any suggestions on the least intrusive way to resolve this issue? If I don’t hear from anyone I am planning on doing a yum update to update the OS & Asterisk, in addition to updating the FreePBX modules. If anyone has a less intrusive suggestion, I would love to hear it.

Thanks! Dave

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Start of system recording is sometimes cutoff

April 16, 2020, 4:18 pm

≫ Next: Outgoing Calls drop after 15 minutes

≪ Previous: Asterisk hanging with channel.c: Exceptionally long voice queue length queuing to Local using PRI

@bhegardt wrote:

I’m using FreePBX 15.0.16.49 / Asterisk 16.9.0

I have a system recording “Thank you for calling…” used in the primary IVR.

When testing this, the words “Thank you” are sometimes cut off and not heard by callers.

I searched for issues like this and found the advice to add 1 second of silence before the recording starts. This does seem to work around the problem.

I notice that on the CrossTalk Solutions video covering system recordings, Chris does not do this and in fact he trims the start of his test recording so that it starts immediately. So I don’t believe adding silence is always necessary.

My FreePBX server is a new installation hosted in the cloud. I’m wondering if I’ve got a problem somewhere.

Any ideas what the root cause of this could be?

Thanks

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Outgoing Calls drop after 15 minutes

April 16, 2020, 4:04 pm

≫ Next: Outbound Calls

≪ Previous: Start of system recording is sometimes cutoff

@joelones wrote:

Hi,

I know this is a common issue and have done the following in an attempt to fix the problem.

Added ‘session-timers=refuse’ and ‘timers=no’ to SIP Legacy Settings.

Using pfSense as a firewall and set Firewall Optimization Options to ‘Conservative’ which is supposed to increase the udp timeout setting

No port forwarding.

This is a call log (unfortunately forgot to turn on debug, will attempt to grab another) https://paste.ubuntu.com/p/Csj88yV8Sq/

Outgoing SIP settings:
username=XXX
type=friend
secret=XXX
qualify=yes
nat=yes
insecure=port,invite
host=voip2.freephoneline.ca
fromdomain=voip2.freephoneline.ca
dtmfmode=auto
disallow=all
context=from-trunk
canreinvite=no
allow=ulaw&g729
keepalive=yes
session-timers=refuse
Incoming SIP settings:
type=user
secret=XXX
insecure=port,invite
fromdomain=voip2.freephoneline.ca
context=from-trunk
Other Info:

SIP provider (freephoneline)

Hardware (Fresh install of Raspbx on a RPi3)

Any suggestions would be greatly appreciated.

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Outbound Calls

April 16, 2020, 8:29 pm

≫ Next: Understanding the CDR reports

≪ Previous: Outgoing Calls drop after 15 minutes

@mwillcut13 wrote:

Getting busy signal when making outbound calls. It only happens when the caller id matches the inbound route they are calling. If we change the caller id from their public dept number the call goes thru. Has been working fine for long time until today. I’m new at this so any help will be appreciated.

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Understanding the CDR reports

April 17, 2020, 4:46 am

≫ Next: Isymphony softphone integration help

≪ Previous: Outbound Calls

@markcrobinson wrote:

Can anyone refer me to a USEFUL (not asterisk Wiki) explanation of the CDR report fields?

For example:
lastdata - what do these all mean?

SIP/131&SIP/220&SIP/224&SIP/225&SIP/226&SIP/233&SIP/234&SIP/603&SIP/606&SIP/607

301,tC,custom/2011FromTheCartQueue,

SIP/606&SIP/607,20,HhtrM(auto-blkvm)Ib(func-apply-sipheaders^s^1),

How do I identify what inbound route a call came in on?
… and a dozen other questions…

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Isymphony softphone integration help

April 17, 2020, 6:55 am

≫ Next: NFON Trunk

≪ Previous: Understanding the CDR reports

@Raven5650 wrote:

hello,
I’m trying to create a custom pbx for our call center. Freepbx seem a good product for this. Now what i need is a user web interface for my operators to login and start receiving calls but also i need some common feature like change status, see other statuses, transfer call, ecc…
iSymphony seems good but from what i see it lacks the softphone (or phone) to make and receive calls.
Has someone already deal with that and in what way?

i have tryed adding a softphone like sipml5 but i see 2 problems:

the user is force to login 2 times, one in which you have to know sip extension, port, ip

doing outbound call the extensions itself (maybe a bug?)

ofc all i have wrote came from my search in internet and some test, and i’m not very expirence with Voice Software so if there are some mistake feel free to correct me.
Thank for the help

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NFON Trunk

April 17, 2020, 7:51 am

≫ Next: Upgrading from FreePBX 13.0.143

≪ Previous: Isymphony softphone integration help

@felonmuc wrote:

Hello Everyone

i’m trying to register my freepbx instance to connect to nfon.
I’ve successfully registered the trunk, but have problems by dialing out.

Does anyone have experience with nfon?

Best regards,
Ralph

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Upgrading from FreePBX 13.0.143

April 17, 2020, 10:39 am

≫ Next: OUTAGE: Exception (1045) SQLSTATE[HY000] [1045] Access denied for user 'freepbxuser'@'localhost' (using password: YES)::

≪ Previous: NFON Trunk

@fisher15 wrote:

Just inherited a FreePBX server version 13.0.143

Looking for the best path to get this machine updated to the latest. Currently supports 15 stores over MPLS connection. I am afraid to just click update.

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OUTAGE: Exception (1045) SQLSTATE[HY000] [1045] Access denied for user 'freepbxuser'@'localhost' (using password: YES)::

April 17, 2020, 10:48 pm

≫ Next: Help: Contacts deleted

≪ Previous: Upgrading from FreePBX 13.0.143

@comtech wrote:

FreePBX/Asterisk 14

Hopefully someone can help as we are suffering an outage. We were noticing a voice quality issue so we rebooted our FreePBX server. When it came back the GUI displayed the following error:

_Exception (1045) _
SQLSTATE[HY000] [1045] Access denied for user ‘freepbxuser’@‘localhost’ (using password: YES)::SQLSTATE[HY000] [1045] Access denied for user ‘freepbxuser’@‘localhost’ (using password: YES)

We found this post by Dave and followed the instructions, to no avail.

Freepbxuser access denied General Help

From the database server, you can do any commands you want as the ‘root’ database user (which shouldn’t have a password, but must be accessed from localhost). Connect to the database server as ‘root’ and log into the UI for MySQL: mysql -u root mysql Once logged in, use a command like: update user set password = password('thepasswordyouwanttouse') where user = 'freepbxuser'; The password is probably stored in one of the config files (I’m not sure which and my test machine is hosed right no…

The password in MYSQL was the same as what was in /etc/FreePBX.conf (the MySQL command said 0 changes made when I tried to update the password to what was originally in the conf file.

For fun I tried to update the password to a new password in the freepbx.conf file and using the SQL command, but it didn’t work. Any ideas? We are hurting without this box. Cant seem to restore without DB access too.

/var/www/html/admin/libraries/utility.functions.php line 204
Arguments
1.“SQLSTATE[HY000] [1045] Access denied for user ‘freepbxuser’@‘localhost’ (using password: YES)::SQLSTATE[HY000] [1045] Access denied for user ‘freepbxuser’@'loca ”

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Help: Contacts deleted

April 17, 2020, 8:33 am

≫ Next: Error on Restore?

≪ Previous: OUTAGE: Exception (1045) SQLSTATE[HY000] [1045] Access denied for user 'freepbxuser'@'localhost' (using password: YES)::

@StephanK wrote:

Stupid mistake. Deleted a private contact group with 50 contacts by mistake.

I have a full backup. Any way I can restore all contacts only. I cannot lose the text messages from today.

Or at least how to access them so I can print them and reenter.

Thanks for your help.

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Error on Restore?

April 17, 2020, 3:17 pm

≫ Next: Trunks won't re-register unless physical server is rebooted

≪ Previous: Help: Contacts deleted

@StephanK wrote:

I was trying to restore a full backup of my FreePBX 13.0.197.22 to a new setup distribution FreePBX 15.0.16.49.

We are a SipStation customer and use the SMS function a lot.

I got the following messages on restore, I believe this is an error:

Processing sipstation
Resetting sipstation
Removing AstDB EntriesChecking routes for trunks…ok
Generating CSS…Done
Restoring from sipstation [FreePBX\modules\Sipstation\Restore]
sipstation found no database definitions in module.xml
Reading Databases Table infromation using module name sipstation
Unable to run restoreLegacyDatabase on sipstation because NO Database information provided
Importing KVStore from sipstation
Importing Advanced Settings from sipstation
Done

Processing sms
Resetting sms
Dropping table sms_messages…Done
Dropping table sms_dids…Done
Dropping table sms_routing…Done
Dropping table sms_media…Done
Updating tables sms_messages, sms_dids, sms_routing, sms_media…Done
Generating CSS…Done
Restoring from sms [FreePBX\modules\Sms\Restore]
Reading Databases Table infromation using module name sms
Importing Databases from sms
Importing table ‘sms_dids’ from legacy sms
Cleaning table: sms_dids
Importing table ‘sms_media’ from legacy sms
Cleaning table: sms_media
PHP Fatal error: Allowed memory size of 536870912 bytes exhausted (tried to allocate 72 bytes) in /var/www/html/admin/libraries/Composer/vendor/doctrine/dbal/lib/Doctrine/DBAL/DBALException.php on line 166

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Trunks won't re-register unless physical server is rebooted

April 17, 2020, 10:38 pm

≫ Next: Yealink cid on outbound calls shows our did instead of number dialed

≪ Previous: Error on Restore?

@mcisar wrote:

Have an older FreePBX server here that I’m babying along until upgrade time. Mid last year it started doing something a bit odd, but never have had the time to worry about it (and even now it’s more of a curiosity than anything). This FreePBX instance is sitting as a virtual machine running under VMware ESXi 6.7. For some strange reason if I ever have to reboot the FreePBX VM it comes back up with no issue, all of the phones register with no problems… but the trunks (to two separate providers) simply sit in the “request sent” state pretty much forever and are not able to register (FWIW the extensions are all PJSIP and the trunks are all still running Chan_SIP). I can shut down the VM and still no-go… the only way I’m able to get the trunks to re-register is to physically restart the VMware server itself. Once everything comes back up and the VM restarts the trunks all register instantly with no intervention.

I’m kind of at a loss as to how that’s even possible, but it is 100% reproduceable.

Anyone ever seen (fixed ) anything like that?

Mike

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Yealink cid on outbound calls shows our did instead of number dialed

April 18, 2020, 5:44 am

≫ Next: PJSIP trunk with sipgate.de Basic account configuration

≪ Previous: Trunks won't re-register unless physical server is rebooted

@ccts wrote:

I’ve got a new fpbx 15 box and yealink t27 and t29g phones, incoming calls show external caller’s cid just fine, but when I make an outbound call on the phones, then I go to history, it only shows our DID as the cid of the outbound call. I suspect this is because I have a DID on our outbound route, but not sure…I thought that’s the way you’re supposed to do it?

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PJSIP trunk with sipgate.de Basic account configuration

April 18, 2020, 10:40 am

≫ Next: Queue Login - Change Feature Code

≪ Previous: Yealink cid on outbound calls shows our did instead of number dialed

@user404 wrote:

Hello everybody,
took me a while today to get FreePBX connected to sipgate.de basic, so I thought it might share my (finally) working configuration. I am using SNG7-2020-02 with todays updates to all modules and I’ll just note what I changed and omit (almost) all the unchanged default values.

PJSIP trunk settings for a Sipgate Basic account in Germany:

Tab General:
TrunkName: SipgatePJSIP (or whatever you want to call it)
Outbound caller ID: <YourPublicPhoneNr> e.g. “<004932112345678>”
Tab pjsip Settings - General
Username: sipgate acount ID, e.g. “1234567a0”
Secret: sipgate account PW
SIP Server: sipgate.de
SIP Server Port: 5060
Context: from-pstn (default)
Tab pjsip Settings - Advanced
Expiration: 600
From User: sipgate acount ID, e.g. “1234567a0” (if you get this wrong, you’ll hear sth like “all lines are busy, pls try again later” on outgoing calls)

Hope this can be helpful for someone. Good luck

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Queue Login - Change Feature Code

April 18, 2020, 10:51 am

≫ Next: TLS error every minute on PJSIP trunk

≪ Previous: PJSIP trunk with sipgate.de Basic account configuration

@vaibhav wrote:

On my old Elastic server, I would login & logout to the queue as follows
XXXX* / XXXX**

With my latest FreePBX server, I need to use:
*45XXXX (toggle)

Is there anyway to change my queue login to XXXX* / XXXX** format? If yes, please point me to any documentation or steps here.

Thanks!!

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TLS error every minute on PJSIP trunk

April 18, 2020, 3:02 pm

≫ Next: Dialled Number (Called Number) in CDR

≪ Previous: Queue Login - Change Feature Code

@Marbled wrote:

Hi!

One of the ITSP provider I use offers TLS for its SIP connections so I have started testing it with one of my test DID.

I decided at the same time to make that trunk a PJSIP one since, AFAIK, this is the current recommendation since Chan_SIP is no longer actively maintained.

I am getting this error (well warning but I doubt this is normal) every minute in my logs, any knows how to fix this?
[2020-04-18 17:41:00] WARNING[2240]: pjproject: <?>:                       SSL 6 [SSL_ERROR_ZERO_RETURN] (Read) ret: 0 len: 32000
[2020-04-18 17:42:00] WARNING[2240]: pjproject: <?>:                       SSL 6 [SSL_ERROR_ZERO_RETURN] (Read) ret: 0 len: 32000
[2020-04-18 17:43:00] WARNING[2240]: pjproject: <?>:                       SSL 6 [SSL_ERROR_ZERO_RETURN] (Read) ret: 0 len: 32000
Thank you and have a nice day!

Nick

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Dialled Number (Called Number) in CDR

April 19, 2020, 9:37 am

≫ Next: ICE disabled on PJSIP trunks

≪ Previous: TLS error every minute on PJSIP trunk

@vaibhav wrote:

FreePBX 15.0.16.49 - ISO Install

My call flow is as follows:

=> Customer dials 888-888-8888 (inbound route)
=> Call enters Queue
=> Extensions 1001 answers the call from the queue

Any suggestions or pointer to documentation as to how I can log Called Number or Dialled Number (888-888-8888) into CDR?

Thanks,
Vai

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FreePBX 15.0.16.49 - ISO Install

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