@Margon wrote:

Good afternoon, I’m just learning asterisk.
I am using FreePBX Distro 14.0.5.25. All extention work on pjsip.
I was able to configure everything so that my 200 phones (snom D715) would work fine.
I also have 2 phones nortel 1200 with which there are problems.
These phones are registered on asterisk and work normally for some time.
after about an hour, asterisk stops seeing them.
It looks like this:

pjsip show endpoint 2237

Endpoint: <Endpoint/CID…> <State…> <Channels.>
I/OAuth: <AuthId/UserName…>
Aor: <Aor…>
Contact: <Aor/ContactUri…> <Hash…> <RTT(ms)…>
Transport: <TransportId…> <BindAddress…>
Identify: <Identify/Endpoint…>
Match: <criteria…>
Channel: <ChannelId…> <State…> <Time…>
Exten: <DialedExten…> CLCID: <ConnectedLineCID…>

Endpoint: 2237/2237 Unavailable 0 of inf
InAuth: 2237-auth/2237
Aor: 2237 1

ParameterName : ParameterValue

100rel : yes
accept_multiple_sdp_answers : false
accountcode :
acl :
aggregate_mwi : true
allow : (ulaw|alaw|g726|g723|g722|g729)
allow_overlap : true
allow_subscribe : true
allow_transfer : true
aors : 2237
asymmetric_rtp_codec : false
auth : 2237-auth
bind_rtp_to_media_address : false
bundle : false
call_group :
callerid : “Чистякова И.” <2237>
callerid_privacy : allowed_not_screened
callerid_tag :
connected_line_method : invite
contact_acl :
context : from-internal
cos_audio : 5
cos_video : 4
device_state_busy_at : 0
direct_media : true
direct_media_glare_mitigation : none
direct_media_method : invite
disable_direct_media_on_nat : false
dtls_auto_generate_cert : No
dtls_ca_file :
dtls_ca_path :
dtls_cert_file :
dtls_cipher :
dtls_fingerprint : SHA-256
dtls_private_key :
dtls_rekey : 0
dtls_setup : active
dtls_verify : No
dtmf_mode : rfc4733
fax_detect : false
fax_detect_timeout : 0
follow_early_media_fork : true
force_avp : false
force_rport : true
from_domain :
from_user :
g726_non_standard : false
ice_support : false
identify_by : username,ip
inband_progress : false
incoming_mwi_mailbox :
language : ru
mailboxes :
max_audio_streams : 1
max_video_streams : 1
media_address :
media_encryption : no
media_encryption_optimistic : false
media_use_received_transport : false
message_context :
moh_passthrough : false
moh_suggest : default
mwi_from_user :
mwi_subscribe_replaces_unsolicited : false
named_call_group :
named_pickup_group :
notify_early_inuse_ringing : false
one_touch_recording : true
outbound_auth :
outbound_proxy :
pickup_group :
preferred_codec_only : false
record_off_feature : apprecord
record_on_feature : apprecord
refer_blind_progress : true
rewrite_contact : true
rpid_immediate : false
rtcp_mux : false
rtp_engine : asterisk
rtp_ipv6 : false
rtp_keepalive : 0
rtp_symmetric : true
rtp_timeout : 0
rtp_timeout_hold : 0
sdp_owner : -
sdp_session : Asterisk
send_diversion : true
send_pai : true
send_rpid : false
set_var :
srtp_tag_32 : false
sub_min_expiry : 0
subscribe_context :
t38_udptl : false
t38_udptl_ec : none
t38_udptl_ipv6 : false
t38_udptl_maxdatagram : 0
t38_udptl_nat : false
timers : yes
timers_min_se : 90
timers_sess_expires : 1800
tone_zone :
tos_audio : 184
tos_video : 136
transport :
trust_id_inbound : true
trust_id_outbound : false
use_avpf : false
use_ptime : false
user_eq_phone : false
voicemail_extension :
webrtc : no

In this case, the call to this extetion passes, but it is impossible to call from it.

I will be immensely grateful for the tips in which direction I need to move.

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@moehammond wrote:

Dear All,

I have freepbx configured with Skype for business and another SIP provider. I am trying to create a dial plan where I can translate a full URI to an extension.

The Tel URI on Skype for Business is something like +12466261223;Ext=44125
I would like to be able to dial from FreePBX only via 44125 and translate this to the full URI or DID only.

I tried to do this by prepending +DID | 4XXXX but the translation comes as +1246626122344125.

Any help would be much appreciated.

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@Shaynek007 wrote:

I was hoping someone could point me in the right direction. First of all our FreePBX server is fully updated and running in our datacenter and all of our 26 sites connect directly to this datacenter. I’ve slowly been converting our sites over to FreePBX with centralized SIP trunking and so far all the sites I’ve converted can use E911 with our provider without any issues. For the sites I’ve converted the only hardware I have at these sites are the phones (PBX Server and Cisco 3845 CUBES are in the datacenters) .The problem is with the next site I need to convert - Our provider is not able to provide E911 service for this site and we are required to install an Analog line at this local facility and direct all 911 calls over this line. From the research I’ve been doing I see I can use the Extension routing module to restrict this site to utilize a different trunk but what I’m looking for is a Analog trunk unit I can install at that facility. Does anyone have any experience with this type of situation and if you do is there a specific model you’ve used for the outbound analog trunk support? Am I moving in the right direction or should I be doing something different?

Any suggestions you can provide would be very much appreciated!!

Thanks
Shayne

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@mvogel4949 wrote:

I’m assuming that if I have a backup of a 32 bit system that can only be restored to another 32 bit system? Or can I restore to a 64 bit system? Thanks

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@grekasa wrote:

Hello,
I am running a Freepbx distro v14 and configure it for chan_sip sms between extensions using the below setup:

Sip settings
accept_outofcall_message=yes
outofcall_message_context=astsms

And added to the extensions_custom.conf file

exten => _.,1,NoOp(SMS receiving dialplan invoked)
exten => _.,n,NoOp(To ${MESSAGE(to)})
exten => _.,n,NoOp(From ${MESSAGE(from)})
exten => _.,n,NoOp(Body ${MESSAGE(body)})
exten => _.,n,Set(ACTUALTO=${CUT(MESSAGE(to),@,1)})
exten => _.,n,MessageSend(${ACTUALTO},${MESSAGE(from)})
exten => _.,n,NoOp(Send status is ${MESSAGE_SEND_STATUS})
exten => _.,n,GotoIf($["${MESSAGE_SEND_STATUS}" != “SUCCESS”]?sendfailedmsg)
exten => _.,n,Hangup()
;
; Handle failed messaging
exten => _.,n(sendfailedmsg),Set(MESSAGE(body)="[${STRFTIME(${EPOCH},%d%m%Y-%H:%M:%S)}] Your message to ${EXTEN} has failed. Retry later.")
exten => _.,n,Set(ME_1=${CUT(MESSAGE(from),<,2)})
exten => _.,n,Set(ACTUALFROM=${CUT(ME_1,@,1)})
exten => _.,n,MessageSend(${ACTUALFROM},ServiceCenter)
exten => _.,n,Hangup()
exten => _.,n,Hangup()

It worked for years but stopped, (the latest sms was at 19/01/2019), the only change was system and module updates.
GS Wave SIP Massage log shows sip/2.0 404 not found as a reply when I send a message!
When it works the responsed was sip/2.0 202 accepted.

I would like to ask if there are any changes of how Freepbx handles custom.conf files or any change that may affect the above functionality?

Thank you very much in advance!

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@pbx3 wrote:

Hello,

We support a few instances of freePBX. We upgraded one instance to version FreePBX SNG7-PBX-64bit releases and I have decided not to upgrade some of the other instances.

I ran yum -y install http://package1.sangoma.net/distro-upgrade-1807-2.sng7.noarch.rpm on one box and now definitely do not want to upgrade another one with a reboot.

How can I prevent this from happening? I tried uninstalling [root@]# rpm -e distro-upgrade-1807-2.sng7.noarch.rpm

And recieved this:
error: package distro-upgrade-1807-2.sng7.noarch.rpm is not installed

How can I prevent the upgrade?
“After installing this, you can reboot the machine at any time and the upgrade will be automatically started.”

Thank you

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@oleggram wrote:

Hello. I have working configuration with Microsoft AD. FreePBX mode is Deviceandusers. LDAP Status = Connected. I can receive User attributes from LDAP w/o problem, but I can’t create a new User from FreePBX Application/Users/Add user. If I try to specify to use LDAP directory instead of Local (Select User Directory) I get an error “LDAP is in read only mode.Addition denied”. I tried to change “Manage group locally” in the Directory configuration - result is the same. What is wrong?

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@oleggram wrote:

Is it possible to configure Freepbx (Deviceandusers mode) to use external database Postgresql instead of internal MySQL. Maybe it is possible via /etc/amportal.conf, but what parameters I should use?

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@blakeryan126 wrote:

I would like to know if anyone can tell me if you can answer an incoming call from a phone that is not ringing. I have about 8 phones in our office however only 3 of them are part of a ring group. We do not need to have all of our phones ringing for every call. I just need to know if there is a way to answer an incoming call from one of the phones that is not part of the ring group.

Thanks.

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@faisalkhan wrote:

hi all,

CDR src field is showing the DID associated or Emergency Caller ID as a src instead of showing the local extension.

how can i set it to show local extension instead of the emergency callerid.

image.jpg829x245 117 KB

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@bhfisher wrote:

Is it safe to update to PHP Version to 5.4 currently on 5.3.3?
I’m using Incredible PBX 13-13.7 on Scientific Linux
Seems Mautic and RoundCube wants PHP 5.4

Thanks, Bart

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@gqwebb wrote:

PBX Firmware: 12.7.5-1902-2.sng7
FreePBX 14.0.5.25
System Admin: 14.0.31

On this new install I am all patched up and getting set up for deployment and the provisioning protocol is missing from the system admin sub menu. I have checked what I can see as the only difference is the pbx firmware on my last system is 12.7.5-1902-1.sng7 and it is working. I am trying to enable TFTP, and I don’t know how to manually enable it. Please help or comment, and if I am missing any key info just ask please.

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@AllianceDoug wrote:

In Twilio’s documentation (found here) for setting up SIP connections, they mention sending SIP Options messages from my PBX to the Twilio SIP Trunk. Here is what is says:

Optionally set-up your Communications Infrastructure to issue SIP OPTIONS messages as a ping mechanism to your Elastic SIP Trunk (Send the Message Request To: Termination URI you created ( example.pstn.twilio.com )); the Twilio platform will respond appropriately. Please maintain the Ping lower than 1 SIP OPTIONS every 10-15 seconds to avoid your requests from being banned by our Platform.

A bit further down in the same page in the “Deploying behing a NAT” (here), it mentions:

If you’re deploying behind a NAT without a Session Border Controller, it’s important to keep open the NAT translation binding.

For Signaling, when using UDP, this may be achieved by periodically sending SIP OPTIONS to Twilio, which will respond with a 200OK.

Where is this set up in FreePBX?

We have intermittent outages that end up resolving after a couple call in/out attempts, and I’m hoping this will help that issue.

Thanks!

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@jarchibald wrote:

I just did a new install of Freepbx distro.

Created a new trunk, inbound and outboud routes and an extension.

I putty’ed into the server and went into asterisk to see the following:

[2019-02-08 14:15:45] ERROR[27973]: pjproject:0 <?>: sip_inv.c .Error parsing/validating SDP body: Missing SDP rtpmap for dynamic payload type (PJMEDIA_SDP_EMISSINGRTPMAP)
[2019-02-08 14:19:11] ERROR[27973]: pjproject:0 <?>: sip_inv.c .Error parsing/validating SDP body: Missing SDP rtpmap for dynamic payload type (PJMEDIA_SDP_EMISSINGRTPMAP)
[2019-02-08 14:37:44] WARNING[27973]: res_pjsip_registrar.c:989 registrar_on_rx_request: Endpoint ‘anonymous’ has no configured AORs
[2019-02-08 14:39:44] ERROR[27973]: pjproject:0 <?>: sip_inv.c .Error parsing/validating SDP body: Missing SDP rtpmap for dynamic payload type (PJMEDIA_SDP_EMISSINGRTPMAP)
== Setting global variable ‘SIPDOMAIN’ to ‘96.47.191.109’
– Executing [9011442030340184@from-sip-external:1] NoOp(“PJSIP/anonymous-00000527”, “Received incoming SIP connection from unknown peer to 9011442030340184”) in new stack
– Executing [9011442030340184@from-sip-external:2] Set(“PJSIP/anonymous-00000527”, “DID=9011442030340184”) in new stack
– Executing [9011442030340184@from-sip-external:3] Goto(“PJSIP/anonymous-00000527”, “s,1”) in new stack
– Goto (from-sip-external,s,1)
– Executing [s@from-sip-external:1] GotoIf(“PJSIP/anonymous-00000527”, “1?setlanguage:checkanon”) in new stack
– Goto (from-sip-external,s,2)
– Executing [s@from-sip-external:2] Set(“PJSIP/anonymous-00000527”, “CHANNEL(language)=en”) in new stack
– Executing [s@from-sip-external:3] GotoIf(“PJSIP/anonymous-00000527”, “1?noanonymous”) in new stack
– Goto (from-sip-external,s,5)
– Executing [s@from-sip-external:5] Set(“PJSIP/anonymous-00000527”, “TIMEOUT(absolute)=15”) in new stack
– Channel will hangup at 2019-02-08 14:49:08.344 EST.
[2019-02-08 14:48:53] WARNING[16505][C-00000527]: func_channel.c:460 func_channel_read: Unknown or unavailable item requested: ‘recvip’
– Executing [s@from-sip-external:6] Log(“PJSIP/anonymous-00000527”, "WARNING,"Rejecting unknown SIP connection from “”) in new stack
[2019-02-08 14:48:53] WARNING[16505][C-00000527]: Ext. s:6 @ from-sip-external: "Rejecting unknown SIP connection from "
– Executing [s@from-sip-external:7] Answer(“PJSIP/anonymous-00000527”, “”) in new stack
> 0x7f7c94009cd0 – Strict RTP learning after remote address set to: 46.166.139.12:5072
– Executing [s@from-sip-external:8] Wait(“PJSIP/anonymous-00000527”, “2”) in new stack
– Executing [s@from-sip-external:9] Playback(“PJSIP/anonymous-00000527”, “ss-noservice”) in new stack
– <PJSIP/anonymous-00000527> Playing ‘ss-noservice.ulaw’ (language ‘en’)
– Executing [s@from-sip-external:10] PlayTones(“PJSIP/anonymous-00000527”, “congestion”) in new stack
– Executing [s@from-sip-external:11] Congestion(“PJSIP/anonymous-00000527”, “5”) in new stack
== Spawn extension (from-sip-external, s, 11) exited non-zero on ‘PJSIP/anonymous-00000527’
– Executing [h@from-sip-external:1] Hangup(“PJSIP/anonymous-00000527”, “”) in new stack
== Spawn extension (from-sip-external, h, 1) exited non-zero on ‘PJSIP/anonymous-00000527’

how can I stop the anonymous connections?

I was getting bombarded with these, like every 2 seconds. I did some changes on the FreePBX firewall and setup some GeoIP blocking rules on my Untangle firewall.
but these are still coming in.

I did just notice that the connection before was at 13:32 or 64 min prior.
Does fail2ban look for these entries?

Thanks

Joe

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@sentinelace wrote:

Has anyone been successful in auto provision this phone? I have one but there is no template in endpoint.

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@jposhea wrote:

Hi All:

Firstly, please forgive me is this is a stupid question but I am deploying 11 of Sangoma S500 handsets with hosted FreePBX as back end and using EndPoint Manager to configure the phones

I have sorted most queries except one that really has my head wrecked.
If I pick up an incoming call on an extension and discover that I need to transfer this call to another extension. Call is live and I hit the “Transfer” Horizontal Key.
I wish to be able to look up the PBX Contacts list to allow me complete the transfer but I don’t seem to be able to look up the Contacts App from within a live call / Talk mode.

When the phone is Idle I have Full Access to this Contact list as I have a horizontal key programmed as “REST-Contacts” but have no access to this once on live call / Talk Mode.

I reckon that the difference is something about “REST-Contacts” is an application running from the PBX and not a “Native Function” of the handset e.g. Directory?

If this is the case, is there any workaround? The old system did it and they seemed to have relied on it quite a bit - now I am quickly losing face . . .

Any help appreciated . . .

Thanks a Million

John

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@doanvulong94 wrote:

I trying generate 1 certificate follow tutorial, but can not generate, please help me why!

Screen Shot 2019-02-10 at 6.25.27 PM.png2812x162 21.3 KB

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@dsirota wrote:

Hi there,

I’m trying to configure two Mitel IP phones for use in our school’s press box/booth. I already have FreePBX up and running in the booth, and I’ve gone through the UI and configured a few things. However, my difficulty comes when I try to connect the phones. I know that Mitel isn’t necessarily the best brand to use with FreePBX, but we had some extras sitting in a closet, and I wanted to put them to use.

I’ve gotten as far as accessing the Mitel IP Phone Configuration tool through my web browser by typing in the phone’s IP address. However, I’m stuck there. The phone gives a dial tone and some preset line buttons, but once I hit “Dial”, nothing happens, and I can’t dial in to the phone either. Here are my questions:

• How can I find the IP address of my FreePBX TFTP server?
• How do I use the EndPoint Manager to configure the phone (which I already paid $150 for an upgraded version of ECM…)?

I would appreciate any help possible as soon as possible. Our spring sports season starts very soon, and I want to make sure that the system is ready to go before then!

Thanks.

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@Whacka wrote:

I really need some help with setting up a small network of cisco IP phones. I cannot get them to register. I have a SPA9000, a PC running FreeBPX and two cisco IP phones. A 7962 and a 8961. Both of them got their own IP addresses. I can access the 7962 online UI but there is nothing to configure. Just shows information. And I cannot access the 8961’s web UI at all, and yes I tried https://. The SPA9000 setup wizard does not see either phone. I’m a newbie, so this article (pbx DOT org/display/FOP/Cisco) didn’t help because I don’t understand how to accomplish almost all of the steps given in the article. I can see it wants me to update the phones firmware, but I don’t know what to download and then where to apply the changes with the firmware.

I just need these to be able to call each other. And I’ll hook up a trunk to make out going calls

So what am I missing, what do I need to do, and how do I do it?

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@LesD wrote:

I see that many people are using FreePBX with Hyper-V.

We have a new server and need to migrate the existing VirtualBox VM to Hyper-V. I have converted the file to VHD format and started it up under Hyper-V - and it fails to boot.

I have seen mention of using the Legacy NIC option and done that but no difference - as expected - as the boot fails before it gets anywhere near that.

I can’t get a clear idea of the issue as I find it impossible to get at the log messages. All I have is the final page
What seems to be relevant are the following warnings:

Could not boot
/dev/SangomaVG/root does not exist
/dev/SangomaVG/swaplv1 does not exist
/dev/mapper/SangomaVG-root does not exist

I’m new to Hyper-V so don’t really know what to change in the settings.

I would be grateful for any pointers.

Many thanks

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