When you fwconsole reload, you overwrote the astdb value you just set with what is in the mysql meetme table.
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Change password (CLI)
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See the number of the person calling, not the SIP number
All my incoming calls are forwarded to external numbers (mobile phones).
When receiving a call on the mobile phone, I can see the SIP phone number.
Is there a way to see the number of the person calling?
Thanks
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See the number of the person calling, not the SIP number
Depends on your carrier and if they allow the transmission of CID info.
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Quick calling via names
I was hoping moving to VOIP would help eliminate the need to print out extension lists.
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Steps to getting FreePBX to work with Office 365 email
Where is this change made? Within Office365 or elsewhere? Great guide!
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Questions about soft-clients
If the endpoints were simple enough, I think it would be nice to have just one. Less objects to manage. If the endpoints were different enough, you might have to split them out due to functionality conflicts.
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Configure Astersk
Please guide me to configure the calling system
what is next step i do to configure my asterisk server after installation?
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Configure Astersk
Rohit: I answered several of your questions, it seems you are trying to build a FreePBX server by yourself without the needed knowledge, I suggest you read the wiki and try to move forward and ask specific questions when you find a specific issue that you can’t resolve. Asking general questions without providing any info will not get you the help you need. Just my humble opinion.
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Outgoing call are not working, icoming working fine.
Hi all
Outgoing call are not working, icoming working fine.
I’m useing Freepbx 14.0.1.19, Asteriks version 13.17.2 and my SIP Channel driver is chan_pjsip.
If I make a outgoing call, the asterisk is setting in the contact vield the username “asterisk”.
My sip trunk provider says that this is not valid.
But how can I change it?
I tried to change it by setting the field - pjsip settings, advanced, contact user, the SIP trunk username. But if I put this there, then nothing has changed at the next call.
An other way to find a solution was, that I change my SIP Channel driver to chan_sip, then the pbx will no longer send any register to my trunk provider.
Have anybody any idea, what do I make wrong?
Thanks in advanced
Thomas
Session Initiation Protocol (INVITE)
Request-Line: INVITE sip:2171001@xx.xx.xx.xx:5083 SIP/2.0
Message Header
Via: SIP/2.0/UDP xx.xx.xx.xx:5083;rport;branch=z9hG4bKPj2dccfcff-68c8-4549-b051-d1b856dcb725
From: sip:2171140@192.168.33.11;tag=ae49fdf0-2dd5-4230-826d-feb7ccee7118
To: sip:2171001@xx.xx.xx.xx
Contact: sip:asterisk@xx.xx.xx.xx:5083
Contact URI: sip:asterisk@xx.xx.xx.xx:5083
Call-ID: 68b2025f-e073-41f7-bb2c-a70069053d5a
CSeq: 29351 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: FPBX-14.0.1.19(13.17.2)
Content-Type: application/sdp
Content-Length: 309
Message Body
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See the number of the person calling, not the SIP number
is there something I can try to check? Without asking the carrier? Is this a setting somewhere in FreePBX?
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Configure Astersk
How to install EndPoint Manager in our FreePbx server modules?
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Voicemails hangs up
This topic was automatically closed 24 hours after the last reply. New replies are no longer allowed.
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Incoming calls only last 20 seconds
Since last update of the FREEPBX server, incoming calls only last approximately 20 seconds.
-- Channel PJSIP/UNE-0000000b left 'simple_bridge' basic-bridge <a3af9e42-f657-4c8f-998b-5ad14e8cac14>
-- Channel PJSIP/110-0000000c left 'simple_bridge' basic-bridge <a3af9e42-f657-4c8f-998b-5ad14e8cac14>
== Spawn extension (macro-dial-one, s, 53) exited non-zero on ‘PJSIP/UNE-0000000b’ in macro ‘dial-one’
– PJSIP/110-0000000c Internal Gosub(crm-hangup,s,1) start
== Spawn extension (macro-exten-vm, s, 20) exited non-zero on ‘PJSIP/UNE-0000000b’ in macro ‘exten-vm’
– Executing [s@crm-hangup:1] NoOp(“PJSIP/110-0000000c”, “Sending Hangup to CRM”) in new stack
– Executing [s@crm-hangup:2] NoOp(“PJSIP/110-0000000c”, “HANGUP CAUSE: 16”) in new stack
== Spawn extension (from-did-direct, 110, 2) exited non-zero on ‘PJSIP/UNE-0000000b’
– Executing [s@crm-hangup:3] ExecIf(“PJSIP/110-0000000c”, “0?Set(__CRM_VOICEMAIL=)”) in new stack
– Executing [h@from-did-direct:1] Macro(“PJSIP/UNE-0000000b”, “hangupcall,”) in new stack
– Executing [s@macro-hangupcall:1] GotoIf(“PJSIP/UNE-0000000b”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,3)
– Executing [s@crm-hangup:4] NoOp(“PJSIP/110-0000000c”, “MASTER CHANNEL: 1513363238.12 = 1513363226.11”) in new stack
– Executing [s@crm-hangup:5] GotoIf(“PJSIP/110-0000000c”, “1?return”) in new stack
– Goto (crm-hangup,s,8)
– Executing [s@crm-hangup:8] Return(“PJSIP/110-0000000c”, “”) in new stack
== Spawn extension (from-internal, , 1) exited non-zero on ‘PJSIP/110-0000000c’
– PJSIP/110-0000000c Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=
– Executing [s@macro-hangupcall:3] ExecIf(“PJSIP/UNE-0000000b”, “0?Set(CDR(recordingfile)=)”) in new stack
– Executing [s@macro-hangupcall:4] NoOp(“PJSIP/UNE-0000000b”, "PJSIP/110-0000000c monior file= ") in new stack
– Executing [s@macro-hangupcall:5] AGI(“PJSIP/UNE-0000000b”, “attendedtransfer-rec-restart.php,PJSIP/110-0000000c,”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/attendedtransfer-rec-restart.php
– <PJSIP/UNE-0000000b>AGI Script attendedtransfer-rec-restart.php completed, returning 0
– Executing [s@macro-hangupcall:6] Hangup(“PJSIP/UNE-0000000b”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 6) exited non-zero on ‘PJSIP/UNE-0000000b’ in macro ‘hangupcall’
== Spawn extension (from-did-direct, h, 1) exited non-zero on ‘PJSIP/UNE-0000000b’
– PJSIP/UNE-0000000b Internal Gosub(crm-hangup,s,1) start
– Executing [s@crm-hangup:1] NoOp(“PJSIP/UNE-0000000b”, “Sending Hangup to CRM”) in new stack
– Executing [s@crm-hangup:2] NoOp(“PJSIP/UNE-0000000b”, “HANGUP CAUSE: 16”) in new stack
– Executing [s@crm-hangup:3] ExecIf(“PJSIP/UNE-0000000b”, “0?Set(__CRM_VOICEMAIL=)”) in new stack
– Executing [s@crm-hangup:4] NoOp(“PJSIP/UNE-0000000b”, “MASTER CHANNEL: 1513363226.11 = 1513363226.11”) in new stack
– Executing [s@crm-hangup:5] GotoIf(“PJSIP/UNE-0000000b”, “0?return”) in new stack
– Executing [s@crm-hangup:6] Set(“PJSIP/UNE-0000000b”, “__CRM_HANGUP=1”) in new stack
– Executing [s@crm-hangup:7] AGI(“PJSIP/UNE-0000000b”, “sangomacrm.agi”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
– <PJSIP/UNE-0000000b>AGI Script sangomacrm.agi completed, returning 0
– Executing [s@crm-hangup:8] Return(“PJSIP/UNE-0000000b”, “”) in new stack
== Spawn extension (from-did-direct, h, 1) exited non-zero on ‘PJSIP/UNE-0000000b’
– PJSIP/UNE-0000000b Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=
Help!!
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Configure Astersk
Rohit I agree with Arielgrin, there is a tremendous amount on online help and information. I self-taught by creating a lab version of Freepbx at home before going live at my business.
Follow this link;
https://wiki.freepbx.org/display/PPS/FreePBX+Distro+First+Steps+After+Installation
Also use the search box at the top of this page.
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XML-API Call Park Default Lot
Hello,
We use a custom Parking lot instead of 70. When a user parks a call with the xml-api button it goes directly to slot 71 instead of 101. Is there a default parking lot setting I’m missing?
Thank you,
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[SOLVED] Forced MODULEADMINWGET to true
how did you solved this?
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How to receive email notifications from voicemail
Hello,
I’m new at FPBx, I’ve set up a trunk and it’s working great for incoming/outgoing calls.
I was wondering how to set up voicemail notifications when an extensions have a new VM.
Do I need to have a mail server?
I’ve just set up the basic information at the extension settings, I mean I’ve enabled the VM notifications and type an email but nothing happens when I left a new VM
Thanks in advance
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Anyone found a solution to the 44 bytes recording issue?
I found plenty of posts and bug submissions describing the problem and most of them suggest that the issue was resolved at some point after upgrading. My system is fully upgraded, and all modules are up to date, but still…
with every valid recording, I also have a 44 bytes wav file containing…nothing.
I know I can schedule a cron job to delete these files, but that’s just a workaround.
Any suggestions?
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How to receive email notifications from voicemail
First check you spambox. My freepbx used to send notifications from something like asterisks@freepbx (so no valid email/fqdn) which were refused by my mail server. I did some tweaking which resulted in this working.
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New FreePBX 14 install crashing every hour after a reboot
We have setup a distro install of FreePBX 14 on a VPS from the official distro. We’ve done this with 12 and 13 with no issues to speak of.
On 14, if we spin it up and leave it alone, it will run for days. However, if we do anything that requires a reboot (like changing the time zone), the system becomes unreachable almost hourly. We can’t see the GUI, we can’t SSH/ping the box, nothing.
We’ve blown it away a few times and started over and the problem persists. The key is always a reboot.
Is there something hourly that 14 is trying to do that 12 & 13 didn’t? The logs we have looked at so far don’t reveal anything.
I am working to get some detailed logs but hoped someone would reveal some random job/process that we might need to check.
Thanks!
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