@ninjoan wrote:
Hi, guys, i have a question FreePBX has the option to return an incoming call to the extension that answers the call the first time?
Example
Caller ID 555-555-3211 Call and that call was pick by Extension 1005
later the same number call 555-555-3211 i would like that, in this case, the call go again to the same extension 1005 if is possible.
Posts: 1
Participants: 1
@BlazeStudios wrote:
[mod - this post and the 15 that follow were split from an unrelated thread here]
A little FYI. The concept of an AOR having multiple contacts is not new. It’s been around since the start of SIP. As well, PJSIP is not new. It’s been around for some time as well. So the fact that Asterisk, in the last 5-6 years, has moved to support both of these things it seems new to you if Asterisk has been your only experience.
It would also be nice to know the source of your statement that headers get split when going over the public Internet when having “too many contacts” for PJSIP. Because that sounds more like a networking/router issue as it makes no sense that nothing bad happens locally but the moment it goes out a router to the Internet headers get split and this is a PJSIP/Asterisk problem.
Posts: 16
Participants: 3
@rabeh wrote:
bonsoir a vous
récemment en plein communication on entends une music d’attente ou un transfert non aboutie
je suis sur asterisk 13merci a vous
Posts: 2
Participants: 2
@shokoienia wrote:
I defined my first Sip Trunk and assigned it to an predefined outbound route, but when I dial a number that match the route dial plan from an extension, I see that sip invite to asterisk and then 100 Trying and Next 183 Session Progress and after RTP for busy lines form asterisk, server received Cancel from Ip Phone and after ACK from extension Call ends. There is no Sip Invite has been send from asterisk to other side of Sip Trunk. only after the calls ends, I see a OPTIONS sip:10.0.100.1:5060 to opsite side and it answers with 200 OK.
It seems , there is a problem with Sip Trunk setting. I think that asterisk can find outbound route and it’s assigned sip trunk, but call does not properly proceed.
Can anyone help me, Why?
Thanks in advanced
Posts: 3
Participants: 1
@Richard_at_F1 wrote:
Heard this static burst on a customer’s system yesterday, they’ve been having a number of these plus some garbled speech and dropped calls, we haven’t been able to find any errors in the Asterisk logs/console related to this and we’re running tcpdumps hoping to capture one and get a closer look. Anyone have any idea what this might be?
http://s000.tinyupload.com/index.php?file_id=00871346650226676798
Posts: 3
Participants: 2
@first0ne wrote:
Hi guys. Can you help me with such trouble: when I try to create new extension, freepbx return me an error
FATAL ERRORINSERT INTO users (extension,password,name,voicemail,ringtimer,noanswer,recording,outboundcid,sipname,noanswer_cid,busy_cid,chanunavail_cid,noanswer_dest,busy_dest,chanunavail_dest) values (“5011”, “”, “5011”, “novm”, “0”, “”, “”, “”, “”, ‘’, ‘’, ‘’, ‘’, ‘’, ‘’) [nativecode=1305 ** PROCEDURE asterisk.Maximum_user_limit_exceeded does not exist]SQL -
INSERT INTO users (extension,password,name,voicemail,ringtimer,noanswer,recording,outboundcid,sipname,noanswer_cid,busy_cid,chanunavail_cid,noanswer_dest,busy_dest,chanunavail_dest) values (“5011”, “”, “5011”, “novm”, “0”, “”, “”, “”, “”, ‘’, ‘’, ‘’, ‘’, ‘’, ‘’)
Trace Back
/var/www/html/admin/libraries/sql.functions.php:25 die_freepbx()
[0]: INSERT INTO users (extension,password,name,voicemail,ringtimer,noanswer,recording,outboundcid,sipname,noanswer_cid,busy_cid,chanunavail_cid,noanswer_dest,busy_dest,chanunavail_dest) values (“5011”, “”, “5011”, “novm”, “0”, “”, “”, “”, “”, ‘’, ‘’, ‘’, ‘’, ‘’, ‘’) [nativecode=1305 ** PROCEDURE asterisk.Maximum_user_limit_exceeded does not exist]SQL - <br /> INSERT INTO users (extension,password,name,voicemail,ringtimer,noanswer,recording,outboundcid,sipname,noanswer_cid,busy_cid,chanunavail_cid,noanswer_dest,busy_dest,chanunavail_dest) values (“5011”, “”, “5011”, “novm”, “0”, “”, “”, “”, “”, ‘’, ‘’, ‘’, ‘’, ‘’, ‘’)/var/www/html/admin/modules/core/functions.inc.php:5552 sql()
[0]: INSERT INTO users (extension,password,name,voicemail,ringtimer,noanswer,recording,outboundcid,sipname,noanswer_cid,busy_cid,chanunavail_cid,noanswer_dest,busy_dest,chanunavail_dest) values (“5011”, “”, “5011”, “novm”, “0”, “”, “”, “”, “”, ‘’, ‘’, ‘’, ‘’, ‘’, ‘’)/var/www/html/admin/modules/core/functions.inc.php:7294 core_users_add()
[0]:/var/www/html/admin/libraries/components.class.php:448 core_users_configprocess()
[0]: extensions/var/www/html/admin/config.php:273 component->processconfigpage()
FreePBX 2.11.0.43
Asterisk (Ver. 11.8.0)
PHP Version 5.3.3
Apache/2.2.15 (CentOS)
Posts: 4
Participants: 2
@ghurty wrote:
I had this set up a long time ago, but that server died and I need to recreate it.
I have an old avaya pbx that is hooked up via a T1 line to my freepbx.
All the extensions on the avaya box start with 3XXX.
So in order to all calls from the freepbx to the avaya, I would make an outbound route to that trunk.
How would I though allow calls from the avaya device to go through the freepbx to be able to make outbound calls?Thank you
Posts: 1
Participants: 1
@sorvani wrote:
I tested out Zulu a long while ago and found it lacking. Well it has had more years to improve and I wanted to look at testing it out again.
But first, I want to know how to clean up this crap. My extension is 103, and I know 99103 should be Zulu. Where are the rest from? Ext 108 was delete a year ago and yet that 9999108 is still hanging out.
Posts: 1
Participants: 1
@otaro wrote:
Hello Dear
Some Operator request user=phonelike this how to insert it on freepbx 14 (asterisk 16) I use PJSIP
Posts: 1
Participants: 1
@SiP1701 wrote:
I changed my admin password today and for some reason I am unabel to log back into my FreePBX.
I have access via CLI
There are so many articles saying different things as to how to change this password via CLI
Can anyone give me clear instructions on how to do this via CLI?12.7.6-1904-1.sng7
Posts: 5
Participants: 3
@AndrewZ wrote:
Is there an option to enable/disable all the trunks with fwconsole at once?
Everything works fine with the individual trunks but I’m wondering what should be the bulk syntax for the command (if any).
Posts: 3
Participants: 2
@a0429 wrote:
Could somebody advise if the below is even possible at all:
CHAN_SIP extension to another extension that is in DND it shows a message ‘Busy Here’
PJSIP extension to another extension in DND, I get Q.850;cause=17.I understand this is cause codes build into the PJSIP driver, however is there any easy way to change PJSIP extensions so that a ‘Busy Here’ message is displayed instead of the SIP response code?
I want to continue using PJSIP for all my extensions, however I want the CHAN_SIP cause codes that is clearer for the end user to see why the other extension is not available.
Posts: 1
Participants: 1
@mkaye wrote:
i have my main system Freepbx15 running in a VM on my server
it has 2 trunks, i for business, 1 for personali have raspbx15 running on a Pi
i have added the IP of my Pi to my 2-line voip phone, and my ATA
i see that these extensions are now registered with both systemsat the moment i have the 2 trunks disabled in my Pi
for testing i disabled the trunk in my main system before enabling on the Pifor automatic failover to work i need the trunks connected from both systems
will this be a problem with my providers (voip.ms & freephoneline)?mark
Posts: 1
Participants: 1
@nav_singh wrote:
So I have an Algo 8373 paging adapter which works fine, until I try to use the “Expanded Zones” option. Algos have the option to use “Basic Zones” that go from zone 1-6 for paging. If you want more zones you can enable “Expanded Zones” which range between 10-50. I’m able to page fine using zones 1-6 (Basic Zones), but the Expanded Zones are giving me trouble. The Algo is set to use DTMF selectable zones, which means after I dial the paging extension, I have to enter 1 for zone 1, 2 for zone 2 and so on.
The thing with Expanded Zones is that you have to dial *10 in order to page zone 10, then *11 for zone 11 and so on. Problem is when I dial the paging extension, then enter *10 to page zone 10, I get an “Access Denied” message. I know that *10 is a feature code for “Contact Manager Speed Dial”, but dialing *11, which isn’t a feature code also gives me “Access Denied”.
I’ve looked around and I can’t figure out what I should be looking for to fix this. Wasn’t able to find anything on any forums either. If anyone has any suggestions, I’d love to hear them.
Thanks!
Posts: 1
Participants: 1
@AndrewZ wrote:
Is there an option to enable/disable all the trunks with fwconsole at once?
Everything works fine with the individual trunks but I’m wondering what should be the bulk syntax for the command (if any).
Posts: 3
Participants: 2
@jsmith wrote:
Please let me first apologize for the impact that yesterday’s outage of Sangoma infrastructure had on your business. We understand the trust you have placed in us to perform reliably, and we did not live up to those expectations yesterday. For that we are truly sorry, and commit to using this event as a learning experience to improve our service to you.
In summary, yesterday we suffered a major outage of our internal database cluster that powers many of the public facing services we provide. This outage had many impacts on our customers including the Sangoma Portal, Support Ticketing systems, FreePBX and PBXact updates, and many other business transactions with Sangoma. While many of Sangoma’s Cloud Services customers were unaffected, the event did create a service disruption for some SIPStation customers, as well as an outage on our FAXStation, VPN and SMS services.
As to what specifically happened, at approximately 2:08PM CST yesterday, one of our production database clusters crashed, and indicated data corruption as a result of the crash. Subsequently we were unable to simply restore it to a working state. Our internal infrastructure teams worked through the night to restore the data, redeploy the database (in a manner that would not be susceptible to another similar problem), and return the services to normal operation. All services were fully restored by 6AM CST today.
In addition to the outage, there was also a lack of effective communication and support for our customers and partners during the event. While there was not a lot of definitive information we could provide during the early portion of the outage, we should have done a better job of keeping you informed about the scope of the outage, and our progress. Some of you have expressed your displeasure with our lack of communication, and we certainly heard you, and commit to doing better in the future. As we continue our investigation into the root cause of this event, we will provide more details over the next few days, along with a plan for corrective actions.
We know that you depend on Sangoma products and services to run your business, and our goal is to exceed your expectations. Yesterday, we fell short of that goal, and for that we apologize. We take the availability and reliability of our services and infrastructure very seriously, and will take steps to prevent this in the future. Thank you for your patience, understanding and continued support.
Posts: 1
Participants: 1
@MC1 wrote:
Hello
Can some one please tell me where I can find the latest firmware release for an S500 phone? I prefer to browse to the IP address of the phone and upload the firmware from within the browser as opposed to using EPM. I therefore need to locate and download the firmware to my desktop before uploading it to the phone.Thanks in advance for any help that can be provided.
Michael
Posts: 2
Participants: 2
@covici wrote:
Hi. I just installed freepbx under DEbian buster with asterisk 16.6. After install I went to the html/admin URL and it asked me for admin username, password and email address and I said set up system. Now, there are two choices, ucp which when I click the browser says not found, and operator panel which waits indefinitely. I am using php 7.3. How do I even troubleshoot this one?
Thanks in advance for any suggestions.
Posts: 2
Participants: 2
@subiacOSB wrote:
I followed this guided:
https://wiki.voip.ms/article/FreePBX_/_PBX_in_a_FlashOutgoing is working, incoming is not. All incoming calls get: The number you have dialed is not in service.
When I look at the CDR I see incoming call:
Playback s [from-sip-external] ANSWERED 00:06
Congestion s [from-sip-external] ANSWERED 00:12Inbound Routes are set to:
DID: Any
CID: Any
Destination: Extention 125Peer Details:
host=sanjose2.voip.ms
username=xxx
fromuser=xxx
secret=xxx
transport=tls
encryption=yes
qualify=yes
qualifyfreq=50
nat=yes
type=peer
directmedia=no
context=from-trunk
insecure=invite
sendrpid=yes
trustrpid=yes
disallow=all
allow=g729&ulaw&gsmRegister String: tls://xxx:xxx@sanjose2.voip.ms:5061~300
Thanks for your help!
Posts: 2
Participants: 2
