@meluvalli wrote:

Hello. I am trying to use the goto function in extensions_custom.conf, however I am not getting it to work right.
When it hits the Goto function, I get error and it drops the caller.
Error:
Channel ‘SIP/mysip-0000002d’ sent to invalid extension but no invalid handler: context,exten,priority=Check-CallerID,s,1

It never gets to even the Wait(1) within the [Check-CallerID].

Here is my config:
[incoming-custom-config]
include => Check-CallerID
exten => s,1,answer
exten => s,n,Wait(1)
exten => s,n,Goto(Check-CallerID,s,1)
exten => s,n,Return()

[Check-CallerID]
exten => s,n,Wait(1)
exten => s,n,GotoIf($["${CALLERID(num)}" = “5555551212”]?ivr-1,s,1)
exten => s,n,Return()

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@hardocp wrote:

I am running Asterisk v16 and Freepbx v14 with a public static ip address

I have setup a PJSIP extension to operate with SIP TLS and a self signed certificate which i generated on my freepbx server.

I have test openssl by conencting to the server as follows:

openssl s_client -showcerts -connect xxx.xxx.com:5066 (yes TLS is running on port 5066)
CONNECTED(00000003)
depth=0 CN = xxx.xxx.com, O = xxx
verify error:num=18:self signed certificate
verify return:1
depth=0 CN = xxx.xxx.com, O = xxx
verify return:1

Certificate chain
0 s:/CN=xxx.xxx.com/O=xxx
i:/CN=xxx.xxx.com/O=xxx
-----BEGIN CERTIFICATE-----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-----END CERTIFICATE-----

Server certificate
subject=/CN=xxx.xxx.com/O=xxx
issuer=/CN=xxx.xxx.com/O=xxx

No client certificate CA names sent
Client Certificate Types: RSA sign, DSA sign, ECDSA sign
Server Temp Key: ECDH, P-256, 256 bits

SSL handshake has read 1411 bytes and written 435 bytes

New, TLSv1/SSLv3, Cipher is ECDHE-RSA-AES256-SHA
Server public key is 1024 bit
Secure Renegotiation IS supported
Compression: NONE
Expansion: NONE
No ALPN negotiated
SSL-Session:
Protocol : TLSv1
Cipher : ECDHE-RSA-AES256-SHA
Session-ID: A09A31CC6B1BB5157BA6C1D79CA0B566EA1D08CD6B4DD42C1CA85DF97E4ED9C3
Session-ID-ctx:
Master-Key: 1A91BF900C526132895D0511A99A0F23BE663A6032D7EA193886C7ED62018092 2785344CCDA58A2F6ABDED6E0D61DEEF
Key-Arg : None
Krb5 Principal: None
PSK identity: None
PSK identity hint: None
TLS session ticket lifetime hint: 7200 (seconds)
TLS session ticket:
0000 - f5 88 ac ee bf 6e 6d a3-30 68 19 a7 1d 51 ea 12 …nm.0h…Q…
0010 - 7b b1 7d 0f 0a f1 22 34-29 49 97 27 10 09 b9 46 {.}…"4)I.’…F
0020 - 70 c9 04 59 2b 1f f6 f3-51 23 62 3d 7e a4 ff 32 p…Y+…Q#b=~…2
0030 - 0b 36 3c 85 ae f0 66 2f-7b 95 b3 2c 94 71 b4 4b .6<…f/{…,.q.K
0040 - 14 ae 76 5f 97 01 9a 62-0b a1 87 75 d8 f5 6c 5e …v_…b…u…l^
0050 - 4e f6 71 c1 5c 85 8c ae-e4 4a 83 27 fc de dd 09 N.q…J.’…
0060 - 18 85 1b f5 fb ef 47 7b-c6 0f fe bc 92 ff 0a 24 …G{…$
0070 - 01 43 dc cb ca 7a 1b 3d-75 d7 12 b4 16 48 ec f6 .C…z.=u…H…
0080 - a6 f0 0f f2 d6 a6 f9 9c-be 86 91 47 1f 16 03 f2 …G…
0090 - 4d ee 6d d8 ad 79 9e 5a-ba bd d7 50 d7 1b ae dc M.m…y.Z…P…

Start Time: 1542556245
Timeout   : 300 (sec)
Verify return code: 18 (self signed certificate)

​

So it connects and everything looks ok on the server side (although i does not seem to like that the certificate is self signed)

Then i went into the phone (Yealink T54S) and set the account setting according to the server and port listed above and set the transport to TLS

Then i went to security and trusted certificates – uploaded the .pem file from the server – disabled only accepted trusted certificates – confirm and reboot and i am unable to get the phone to register the extension with asterisk

i have also tried connecting the same extension using UDP and that works fine

there is some issue getting the SSL certificates to negotiate a connection – i have been working on this for a few days and really need to get this working as we can not roll out this server until we can secure calls between remote phones and the server

Can you please help me?

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@digitalo wrote:

Probably a really silly question but I’m unable to find the options menu for the print extensions module.
According to this page there should be an options menu on the right.
I only have a button “print”.

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@Samwheel wrote:

I have been tasked with upgrading an old FreePBX system. I plan to set up a system running the same version and import the config then step trough the upgrade process.
The system has no access to the internet.
I have not been able to find any version of FreePBX 10.13.66-6.

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@Bedrock wrote:

FreePBX voicemail to email spamming issue.

I’ve gone over this a lot, and can not find the problem. Email is using the standard Postfix Mail server. The Postfix appears to be setup, like all the other boxes i’ve done. It is setup to relay email. It does authentication, and everything works fine with the authentication, and emailing.
FreePBX 12.0.76.4

The Problem
It is sending all messages to the forwarding host, so all the garbage messages - like root and asterisk messages. So the relaying host gets spammed from the server ~ 1000 emails a day, when only a few are actual voicemails. It would seem there’s a local delivery option or something I’ve overlooked, but I can not find it, and I’ve done this several times.

I have Webmin installed, for ease of management.

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@floriandinca wrote:

Hello everyone, sorry for my post, I’m new in this area and I need help to configure the trunk provided by the ISP, i will try to give all the information below. Thanks and sorry for the post again.

1.I have a Freepbx machine in Hyper-V environment(FreePBX 14.0.4.5).
2.Eth0 is configured with the ip provided from the ISP(10.103.49.194)
3.Eth1 is configured with my local ip address(10.0.7.15)

Freepbx Version: FreePBX 14.0.4.5

The ISP provider gave me the fallowing settings:

IP: 10.103.49.194
Mask: 255.255.255.252
IP GW: 10.103.49.193
Local DNS : IP SBC 172.29.62.97 => imt.orange.ro
Add: static route Gateway to imt.orange.ro
Add: “User = phone” in header SIP “TO”

-        user=phone parameter must exist  in To/From fields

•    Proxy IP: 172.29.62.97
  

•    allowed codecs: G.711a/u, G.729
  

•    numbering plan is international (+40)
  

•    the entry 172.29.62.97 -> imt.orange.ro must be added in the local dns table
  

•    used domain: imt.orange.ro
  

Invite Example:
Request-Line: INVITE sip: +40744448157@ imt.orange.ro ;user=phone SIP/2.0

Message Header

From: <sip: +40374442608@ IP :5060;user=phone>;tag=dc60464184a1e3c3

To: <sip: +40744448157@imt.orange.ro;user=phone>

Contact: <sip:+40374442608@ IP :5060;transport=udp>

I’ve added imt.orange.ro to my localhost table
My trial to the sip configuration is:

type=friend
t38pt_udptl=yes
host=imt.orange.ro
insecure=port,invite
dtmfmode=rfc2833
disallow=all
allow=alaw&ulaw
usereqphone=yes
qualify=yes
context=from-pstn
nat=no
externip=10.103.49.194

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@PitzKey wrote:

Hello everyone,

Lately two separate PBX servers (1 v13 and two v14’s) started sending occasionally a single disk full alert (usually if your disk is actually full, you’ll receive the alert every hour), but when checking in the morning the disk space is far from 75% full.

Yesterday, a third PBX started doing this. email says disk is 77% used, and when I checked the PBX it’s 42% used.

I suspected that this is happening while backup is running, but after checking, I saw that backup completed Sunday 12:?? AM and we received the alert on Sunday 1:59 PM

Can anyone please advise which logs I should start looking at?

Thanks

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@anty506 wrote:

I attempted to move our FreePBX to a new server and all seemed to work except for outbound calls from the desk phones.

So I made sure both servers were same versions and I did a backup on the old server and a restore on the new server.

Everything moved over just fine. I changed our DNS on our side to point hostname to the new public IP.

Inbound calls work just fine. Outbound calls work if I test from a softphone. BUT! The desk phones are not working. They register just fine and inbound flows to the phone but when you try to dial out to either internal or external, nothing happens. If you keep trying over and over, you will get maybe on call out to *98 but then the rest of the time you get “No destination” on the grandstream phone.

Provisioner also works just fine, only outbound.

Am I missing a setting somewhere or a config file that needs to be updated with the new IP or etc?

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@bamio wrote:

Hi all

I have a system with FreePBX 2.11.0.11, with several Cisco 7970 phones.

Now, i’m trying to add a Cisco cp-8851 with very similar xml configuration, only changing fiwmare (sip88xx.12-1-1SR1-4), but i always get the same message in the phone: “phone is registering” … and then unprovisioned

Here is a portion from the log on the phone:

9679 DEB Nov 19 19:23:20.722080 (677:901) JAVA-SIPCC-SIP_CC_PROV: ccsnap_gen_deviceEvent: g_deviceInfo.ins_state=2
9680 NOT Nov 19 19:23:20.722101 (677:901) JAVA-SIPCC-SIP_CC_PROV: ccsnap_gen_deviceEvent: event type : SERVER_TRANSPORT
9681 DEB Nov 19 19:23:20.722120 (677:901) JAVA-SIPCC-SIP_CC_PROV: ccsnap_gen_deviceEvent: ref_count=1 
9682 DEB Nov 19 19:23:20.722141 (677:901) JAVA-SIPCC-SIP_CC_PROV: ccsnap_gen_deviceEvent: name=SEP5006AB0876B6 : privacy=0 : hlog=1 : dnd_state=0 : mwi_lamp=0 
9683 DEB Nov 19 19:23:20.722162 (677:901) JAVA-SIPCC-SIP_CC_PROV: ccsnap_gen_deviceEvent: dnd_type=01 : ins_state=02 : cucm_mode=FFFFFFFF : ins_cause=01 
9684 DEB Nov 19 19:23:20.722233 (677:901) JAVA-SNAPSHOT-RELEASE: CCAPI_Device_releaseDeviceInfo:  reference pointer=4580cfd0
9685 NOT Nov 19 19:23:20.722406 (677:902) JAVA-SIP : sip_transport_init_ti_addrs : ip_mode is:0.
9686 NOT Nov 19 19:23:20.722438 (677:902) JAVA-SIPCC-SIP_TRANS: sip_transport_init_ti_addr:  Entered
9687 DEB Nov 19 19:23:20.722469 (677:902) JAVA-PLAT : MED_API : platGetLocalIPAddr :Hi,It contains IPV6_INTEGRATION
9688 DEB Nov 19 19:23:20.722498 (677:902) JAVA-SIPCC-SIP_TRANS: sip_get_local_ip_addr: dst_addr: 8.8.8.8
9689 DEB Nov 19 19:23:20.722524 (677:902) JAVA-SIPCC-SIP_TRANS: sip_get_local_ip_addr: src_addr: 192.168.1.227
9690 DEB Nov 19 19:23:20.722598 (677:902) JAVA-[getDeployMode] deploy-mode:1 
9691 DEB Nov 19 19:23:20.722636 (677:902) JAVA-SIPCC-SIP_TRANS: sip_transport_getaddrinfo: 192.168.1.234 is already an IPv4 address
9692 DEB Nov 19 19:23:20.722660 (677:902) JAVA-PLAT : MED_API : platGetLocalIPAddr :Hi,It contains IPV6_INTEGRATION
9693 NOT Nov 19 19:23:20.722681 (677:902) JAVA-SIPCC-SIP_TRANS: sip_transport_init_ti_addr:  Entered
9694 NOT Nov 19 19:23:20.722702 (677:902) JAVA-SIPCC-SIP_TRANS: sip_transport_init_ti_addr: Unexpected value specified for ip_type : 0
9695 ERR Nov 19 19:23:20.722725 (677:902) JAVA-SIPCC-SIP_TRANS: sip_transport_get_ti_addr: No active CUCM found using primary CUCM
9696 ERR Nov 19 19:23:20.722746 (677:902) JAVA-SIPCC-SIP_TRANS: sip_transport_get_ti_addr: No active CUCM found using primary CUCM
9697 NOT Nov 19 19:23:20.722769 (677:902) JAVA-SIPCC-UI_API: ui_set_ccm_conn_status: ***********CUCM 192.168.1.234 Not connected***********
9698 DEB Nov 19 19:23:20.722873 (677:901) JAVA-SIPCC-SIP_CC_PROV: ccsnap_update_ccm_status: entry ccm 192.168.1.234 status=0
9699 DEB Nov 19 19:23:20.722908 (677:901) JAVA-SIPCC-SIP_CC_PROV: ccsnap_update_ccm_status: Find cucm with addr= 192.168.1.234,one of the cucms' name=192.168.1.234 (ipv6:INVALID_IPV6)
9700 DEB Nov 19 19:23:20.722934 (677:901) JAVA-SIPCC-SIP_CC_PROV: ccsnap_update_ccm_status: server(ipv4) 192.168.1.234 is now status=0, index=0
9701 DEB Nov 19 19:23:20.722968 (677:901) JAVA-SIPCC-SIP_CC_PROV: ccsnap_update_ccm_status: Find cucm with addr= 192.168.1.234,one of the cucms' name=INVALID (ipv6:INVALID_IPV6)
9702 DEB Nov 19 19:23:20.722992 (677:901) JAVA-SIPCC-SIP_CC_PROV: ccsnap_update_ccm_status: Find cucm with addr= 192.168.1.234,one of the cucms' name=INVALID (ipv6:INVALID_IPV6)
9703 DEB Nov 19 19:23:20.723015 (677:901) JAVA-SIPCC-SIP_CC_PROV: ccsnap_update_ccm_status: Find cucm with addr= 192.168.1.234,one of the cucms' name=INVALID (ipv6:INVALID_IPV6)
9704 DEB Nov 19 19:23:20.723035 (677:901) JAVA-SIPCC-SIP_CC_PROV: ccsnap_update_ccm_status: Find cucm with addr= 192.168.1.234,one of the cucms' name=INVALID (ipv6:INVALID_IPV6)
9705 DEB Nov 19 19:23:20.723067 (677:901) JAVA-SNAPSHOT-CREATE: CCAPI_Device_getDeviceInfo:  g_deviceInfo.ins_state=2
9706 DEB Nov 19 19:23:20.723097 (677:901) JAVA-SNAPSHOT-CREATE: CCAPI_Device_getDeviceInfo:  deviceInfo->sis_name=
9707 DEB Nov 19 19:23:20.723118 (677:901) JAVA-SNAPSHOT-CREATE: CCAPI_Device_getDeviceInfo:  reference pointer=4580cfd0
9708 DEB Nov 19 19:23:20.723144 (677:901) JAVA-SNAPSHOT-CREATE: CCAPI_Device_getDeviceInfo:  deviceInfo->ins_state=2

May i need to upgrade to a newer freepbx version? Or what’s the problem? I can give you further details or logs

Could you help me? Thank you

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@bajramia wrote:

Hi all,
I have a remote connection over site to site vpn and i have 3 phones there the issue is when they put peoples on hold i wont hear MOH also they cant park im using sonicwall firewalls.

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@bajramia wrote:

Hi all,
Im having issue with ucp keep reloading i get this message on the debug.

Mon Nov 19 18:25:42.809722 2018] [authz_core:error] [pid 169237] [client 192.168.xxx.xxx:43763] AH01630: client denied by server configuration: /var/www/html/ucp/index.html, referer: http://XXX.XXX.XXX.XXX:81/

Thank you

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@digitalo wrote:

Hi Guys,

I was wondering when this feature would be available? (or is it already?)
I found this document online which stated that the feature would be available in an upcoming release.

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@mpforum wrote:

I’ve created several extensions for testing but have found that I can get X-Lite and MicroSIP to connect to SIP successfully, but when trying to connect to my extension configured for PJSIP, Asterisk CLI tels me that it’s the wrong password.

SIP is listening on port 5060 (UDP)
PJSIP is listening on port 5160(UDP)

I just tried using Zoiper and have the following when trying to connect to PJSIP
SIP TCP not found
SIP UDP not found
IAX UDP not found
“We couldn’t detect any transport type”

Yet, with SIP, SIP UDP was found straight away.

What am I missing? Am I using the wrong port? Wrong softphone?

I have been told that we need to use PJSIP with FlowRoute. Is this correct?

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@Tommaso wrote:

Hello everyone, I founded some treads with the same question but they are all reeeeally old (like around 2010)

i read about “FOP2” but i was just wondering if there is an up-to-date way to monitoring the status of my estension (like see wich are actually busy, and what number they are caling and stuffs like this)

regards!

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@mvogel4949 wrote:

I have a FreePBX 13 system that just won’t update pm2. I access the update via the CLI and it had me look in the install logs. There I found the following warning

npm ERR! fetch failed http://mirror1.freepbx.org/npm/pm2-2.10.5.tgz

The IT guy said he can see traffic going to the mirror1 site but there is no reply coming back. any ideas?

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@mvogel4949 wrote:

I think it is fairly well known that if a system loses internet that asterisk grinds to a halt trying to resolve URLs for SIP Trunks. Is this just with a certain version of asterisk or all versions? Is the best option to replace URL in SIP Trunk with IP?

We had a school system using SIP and POTS lines for backup in case of emergency. Internet was lost so Asterisk stopped and the phones at that point couldn’t reach the system or use the POTS lines.

Besides the conversion of URL to IP in the SIP Trunks is there anyway to keep this from happening?

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@Oleg32 wrote:

Please help!
Not applies Config FreePBX 13.0.195.13
Backup module not work

Reload failed because retrieve_conf encountered an error: 1

exit: 1
Exception: could not find driver::could not find driver in file /var/www/html/admin/libraries/utility.functions.php on line 204
Stack trace:

1. Exception->() /var/www/html/admin/libraries/utility.functions.php:204
2. die_freepbx() /var/www/html/admin/libraries/BMO/Database.class.php:131
3. PDOException->() /var/www/html/admin/libraries/BMO/Database.class.php:128
4. PDO->__construct() /var/www/html/admin/libraries/BMO/Database.class.php:128
5. FreePBX\Database->__construct() /var/www/html/admin/libraries/BMO/FreePBX.class.php:72
6. FreePBX->__construct() /var/www/html/admin/bootstrap.php:141
7. require_once() /etc/freepbx.conf:9
8. include_once() /var/lib/asterisk/bin/retrieve_conf:9

ministr@RegSRV:~$ sudo fwconsole ma download framework
Exception: could not find driver::could not find driver in file /var/www/html/admin/libraries/utility.functions.php on line 204
Stack trace:

1. Exception->() /var/www/html/admin/libraries/utility.functions.php:204
2. die_freepbx() /var/www/html/admin/libraries/BMO/Database.class.php:131
3. PDOException->() /var/www/html/admin/libraries/BMO/Database.class.php:128
4. PDO->__construct() /var/www/html/admin/libraries/BMO/Database.class.php:128
5. FreePBX\Database->__construct() /var/www/html/admin/libraries/BMO/FreePBX.class.php:72
6. FreePBX->__construct() /var/www/html/admin/bootstrap.php:141
7. require_once() /etc/freepbx.conf:9
8. include_once() /var/lib/asterisk/bin/fwconsole:12
   ministr@RegSRV:~$

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@alexkg1 wrote:

Please choose which filter to set for importing users only from the group “pbx_users”. He made the group himself in the domain and added there those who would use SIP.
Group object filter -?
if so: (& (objectCategory = Group) (sAMAccountName = pbx_users)) does not work.
Please tell me how to select users only from the group “pbx_users” in my Active Dirrectory.

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@vespino wrote:

A client is complaining about dropped calls. How can I best troubleshoot this? I have started asking to log when this happens, so I have date, time, incoming or outgoing, from, to, duration of the call. Is this a good start? But then what?

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@chrischevy wrote:

Quick question here:

Currently, we have a customer that manually enables a call flow control at the end of the day for the night IVR. The person in charge of activating this call flow forgot twice to activate it in the last week, which led to some unanswered emergency calls.

At the time of the original installation of the system, the customer didn’t want a to use a Time Condition because the start time in the morning is not always the same. Sometimes, they even leave the night IVR for days.

They are now asking me to automatically activate the night IVR at 5pm but they don’t want it to disable itself automatically

My question is:
Can I create a Time Condition with only a start time ? I want it to enable itself automatically but never disable itself automatically. I thought about enabling it at 5pm and disabling it at 4:59pm but this creates a 1 minute windows where calls can be lost…

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