no we are not using AD. these were once users on the system which i thought we deleted a long time ago. the upgrade seems to have brought them back.
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How to delete unused users in user manager
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PJSIP extension won't register when created via extension module
Here is the error from /var/log/asterisk/full
[2018-05-14 17:50:49] NOTICE[25927] res_pjsip/pjsip_distributor.c: Request ‘REGISTER’ from ‘“1104” <sip:1104@obfuscated IP address of server>’ failed for ‘obfuscated IP address of endpoint:49752’ (callid: BNRT2mxwQYMRwpFv679MxQ…) - No matching endpoint found
[2018-05-14 17:50:49] NOTICE[25927] res_pjsip/pjsip_distributor.c: Request ‘REGISTER’ from ‘“1104” <sip:1104@obfuscated IP address of server>’ failed for ‘obfuscated IP address of endpoint:49752’ (callid: BNRT2mxwQYMRwpFv679MxQ…) - No matching endpoint found
[2018-05-14 17:50:49] NOTICE[25927] res_pjsip/pjsip_distributor.c: Request ‘REGISTER’ from ‘“1104” <sip:1104@obfuscated IP address of server>’ failed for ‘obfuscated IP address of endpoint:49752’ (callid: BNRT2mxwQYMRwpFv679MxQ…) - Failed to authenticate
Here is the config generated by the extensions module:
[0.0.0.0-udp]
type=transport
protocol=udp
bind=0.0.0.0:6060
external_media_address=obfuscated IP address
external_signaling_address=obfuscated IP address
allow_reload=yes
local_net=192.168.100.0/24
local_net=192.168.5.0/24
[0.0.0.0-tcp]
type=transport
protocol=tcp
bind=0.0.0.0:6061
external_media_address=obfuscated IP address
external_signaling_address=obfuscated IP address
allow_reload=yes
local_net=192.168.100.0/24
local_net=192.168.5.0/24
[1104]
type=aor
max_contacts=1
remove_existing=yes
maximum_expiration=7200
minimum_expiration=60
qualify_frequency=60
[1104-auth]
type=auth
auth_type=userpass
password=password
username=1104
[1104]
type=endpoint
aors=1104
auth=1104-auth
allow=ulaw,alaw,gsm,g726,g723,speex,g722,silk
context=from-internal
callerid=device <1104>
dtmf_mode=rfc4733
aggregate_mwi=yes
use_avpf=no
rtcp_mux=no
ice_support=no
media_use_received_transport=no
trust_id_inbound=yes
media_encryption=no
timers=yes
media_encryption_optimistic=no
rtp_symmetric=yes
rewrite_contact=yes
force_rport=yes
language=en
[1104-identify]
type=identify
endpoint=1104
If i delete the ext from extensions module and build a config manually with the config shown below it will register:
[1104]
type=aor
max_contacts=1
remove_existing=yes
maximum_expiration=7200
minimum_expiration=60
qualify_frequency=60
[auth1104]
type=auth
auth_type=userpass
password=password
username=1104
[1104]
type=endpoint
aors=1104
auth=auth1104
allow=ulaw,alaw,gsm,g726,g723,speex,g722,silk
context=from-internal
callerid=device <1104>
dtmf_mode=rfc4733
aggregate_mwi=yes
use_avpf=no
rtcp_mux=no
ice_support=no
media_use_received_transport=no
trust_id_inbound=yes
media_encryption=no
timers=yes
media_encryption_optimistic=no
rtp_symmetric=yes
rewrite_contact=yes
force_rport=yes
language=en
[1104-identify]
type=identify
endpoint=1104
I do not have any custom modules or code at the moment for troubleshooting purposes.
Does it matter what version of asterisk I’m running? I’m running Asterisk 13.19.1
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PJSIP extension won't register when created via extension module
When you restart asterisk it will tell you in the logs what issues it’s having.
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Avaya 96x1 extended Features
Shawn, can you describe what the phone needs to see to start contacting the PPM server for more than just ‘getInitialEndpointConfiguration’? I’m trying to implement correct replies for my SIP proxy but I’m having trouble following your Asterisk patch to make sense of it all.
I see the phone sends SUBSCRIBE with Event: avaya-cm-feature-status but I don’t really know what to do from here. I can send 202 Accepted but then do I need to send NOTIFY? What headers/info do I need to send in NOTIFY? Do you happen to have traces you can upload of what the SIP dialogue should look like to enable each feature?
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How to delete unused users in user manager
In User Management, Users tab, select a directory. You can then add or delete users at will. You may have surplus directory(ies) as well.
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Polycom Color Expansion Module not populating
From the pictures I see that you still have empty buttons on your phone.
I remember having this issue with Polycom, if you have, say 10 buttons on your phone, and you assign 5 buttons and then you add 5 to the EXP, it will fill up the phone instead of going to the EXP.
So try adding a couple more extensions to your EXP, and see if it adds at least 1 extension to the EXP.
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How to delete unused users in user manager
that works for the directory called “imported freepbx directory”, but I select the one called “imported voicemail directory” it does show all these users I want to delete but there is simply no delete button or icon for these users.
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Polycom Color Expansion Module not populating
AFAIK, all Polycom phones are unusual that way, the buttons on the phone must be full before they will overflow to the sidecar.
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Critical Errors Found - There are 83 bad destinations
That sparked my memory, as I did have edge mode enabled for trying out a module, and the edge version of contact manager got installed. I rolled it back to 14.0.3.5, and no errors with destinations anymore.
Thanks.
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Inbound caller ID includes @IP address
Hi Dave (@cynjut)
I use X-Lite 3 and also Phoner Lite, and they both perform the same way. Interestingly, I use the same softphone configuration (except for the server address/password) with another FreePBX system ( FreePBX 13.0.194.10 ) that system only passes the caller ID to the phone.
With X-Lite, you can have an address book that will look up the Caller ID and display the person’s name (if in the book). To get X-Lite to display the name on the softphone with the the FreePBX 14 system, X-lite needs to include the @ IP address to match correctly. With the @'s in the number entry, the address book entry cannot be used to dial out.:confused.
I’m confused.
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Issue with GUI and Trusted Interfaces
First off - I feel dumb for failing to mention that they are virtual (ESXi 6.0). It still seems strange to me that this would be the case. Nevertheless - I would love to know what those extra-PBX settings are as well as recommended adapter for the vm.
Thank you for your previous response.
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FindmeFollow CID Prefix Not Displayed on Mobile Phones
I’m confused a bit. Assume prefix assigned in FMFM is “PREFIX”, CallerID(num) is 9995551212. When PBXact dials the local extension, the invite contains
P-Asserted-Identity: “PREFIX 9995551212” sip:9995551212@12.34.56.78
But when it dials the mobile, the P-Asserted-Identity is not sent.
So, when I tell FMFM to add a prefix, I assume it is inserting the P-Asserted-Identity, not changing CallerID(name). Apparently most carriers ignore P-Asserted-Identity. I added code to set the From: in invite to
From: “FAFS:17278001068” sip:17278001068@71.100.73.81;"
but Verizon still ignores it and Just says, “Unknown Name”. I know they pass CallerID(name) because I usually see it on my cell phone. I thought maybe it was the colon or the length, so I tried just sending
From: “FAFS” sip:17278001068@71.100.73.81
but still no joy. Does anyone know how to send invite to Verizon that results in CalllerID(name) being displayed?
(Those invite lines have “<” and “>” in proper place, but they are not displayed here.)
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FindmeFollow CID Prefix Not Displayed on Mobile Phones
Verizon will never pass the CID Name. They take the number you send and look it up in their own LIBDB that they maintain and use the name from that database.
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PJSIP extension won't register when created via extension module
Found this in /var/log/asterisk/full after reloading asterisk. Could not find anything else on that ext other than it being loaded to the dial plan. This is with 1104 built within extensions module and nothing built in the custom area of configuration file editor.
[2018-05-14 22:00:26] ERROR[17492] config_options.c: Error parsing allow=ulaw,alaw,gsm,g726,g723,speex,g722,silk at line 15 of /etc/asterisk/pjsip.endpoint.conf
[2018-05-14 22:00:26] ERROR[17492] res_sorcery_config.c: Could not create an object of type ‘endpoint’ with id ‘1104’ from configuration file ‘pjsip.conf’
[2018-05-14 22:00:26] ERROR[17492] res_pjsip_config_wizard.c: Unable to load config file ‘pjsip_wizard.conf’
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Can't find 'firewall' user in /etc/asterisk/manager_additional.conf
Hello All,
Getting the below error and can’t click “apply changes” nor does an fwconsole restart get past the error.
Console Error:
Can’t find ‘firewall’ user in /etc/asterisk/manager_additional.conf. Firewall m
onitoring broken! (Did you forget to click ‘Reload’?)
GUI Error:
exit: 255
Unable to continue. Call to undefined function FreePBX\modules\endpoint_apiApp() in /var/www/html/admin/modules/restapps/Restapps.class.php on line 1314
#0 /var/www/html/admin/libraries/Composer/vendor/filp/whoops/src/Whoops/Run.php(383): Whoops\Run->handleError(1, ‘Call to undefin…’, ‘/var/www/html/a…’, 1314)
#1 [internal function]: Whoops\Run->handleShutdown()
#2 {main}
Any assistance or direction would be appreciated.
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Class of Service - ChanSpy / Barge
Hello, that was the problem!
It was intentionally disabled so nobody would spy…
thanks!
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PJSIP extension won't register when created via extension module
Well I would like to thank everyone that helped. Its fixed and working now.
After seeing the error in the log as Andrew mentioned, it seemed like it was something to do with the codec part of the config.
I ended up going into the advanced tab for that extension and i set “Disallowed Codecs” = all
and “Allowed Codecs” to ulaw and the error went away and my extension now registers properly and everything works as it should.
Thank you once again!
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NetBorder SS7 VoIP Gateway Models for H248
hello:
My customer wants to ask for H248 with netborder appliance, does netborder SS7 still can support:
VoIP protocols: SIP V2/RFC3261, SIGTRAN M2UA RFC 3331, SCTP RFC 2960, Megaco/H248
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Freepbx on Azure or AWS?
We are trying to cut costs. We have over 30 VMs running with a provider about $25 a piece per month. I was curious if anyone has tired AWS or Azure and if that might be cheaper? We do not currently get charged for usage. Anyone have issues running freepbx with Azure?
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NetBorder SS7 VoIP Gateway Models for H248
Specs say it does so yeah it should.
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