@jmillican wrote:

Not sure I fthis is thr correct place but I am wondering what is the process for requesting a code update to a PHP module for FreePBX. In the bulkhandler module there is Bulkimport.class.php which currently works for importing DIDs and Extensions. I was looking for a way to import contacts and found that this can be done using fwconsole bulkimport if I added the following to Bulkimport.class.php inside the switch statement under “protected function execute(InputInterface $input, OutputInterface $output)”

case 'contacts':
            $output->writeln('Importing contacts');
            $ret = \FreePBX::Bulkhandler()->import('contacts', $data, $replace);
break;

Than I can call: fwconsole bulkimport --type=contacts contacts.csv --replace

I create the contacts.csv from a script running from crontab which pulls data from Active Directory and I use to populate the phonebook on the enduser phones.
The problem is this changes the signature and I get the red security warnig bar, I would like to avoid this.
Thanks,
John

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@JDEC01 wrote:

Hi all,

I have a script that pushes every night new entries to a Phonebook in the DPMA module. But the phonebook is not actually refreshed to the contacts-1.xml file.

How is it possible to force the file refresh in ssh ?

Thanks.

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@wpns wrote:

The description for Encryption for Amazon Chime Voice Connector trunks says:

/*
Encryption configures your Voice Connector to use TLS transport for SIP signaling and Secure RTP (SRTP) for media. Enabling encryption causes inbound calls to use TLS transport, and blocks unencrypted outbound calls.
*/

It tells me I must:

/*
Import the wildcard root certificate into your SIP infrastructure. Download here.
*/

I’ve tried a couple of things:

1. Put the above .pem file into /etc/asterisk/keys and click Admin->Certificate Management->Import Locally, but that gets me “No certificates to import”. Rebooting doesn’t help.

2. Copy the contents of the file into Admin->Certificate Management->New Certificate->Upload Certificate-> but that throws other errors.

I’ve tried to follow the instructions at https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial but that seems to be more about how to make my own certificate and configure extensions to use encryption, and I want to use someone else’s certificate and enable trunk encryption…

I must be missing something simple, can someone enlighten me?

Thanks!

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@wpns wrote:

When I ssh in and tell the system to reboot, or I use Admin->System Admin->Power Options->Reboot, the system takes just over 120 seconds to stop responding to ping and shut down before restarting.

It’s probably some kind of housekeeping timeout, but is there a way to shorten the time?

Brand new install and fully upgraded to 12.7.6-1904-1.sng7 if that helps.

Thanks!

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@EdPaine wrote:

I’m in the UK (+44) so the standard context (from-pstn-e164-us) doesn’t work for me. Thanks to several other posts on this forum I came up with the following, which I have added to /etc/asterisk/extensions_custom.conf. It first substitutes 0 for +44, if found, and then looks for a single + (assumed a non-UK country) which is substituted with 00 (UK international exit code).
Then I set the context in our trunk from from-pstn to from-pstn-remove-plus

[from-pstn-remove-plus]
exten => _X!,1,GotoIf($["${CALLERID(num):0:3}" != “+44”]?noukplusatstart)
exten => _X!,n,Set(CALLERID(num)=0${CALLERID(num):3})
exten => _X!,n(noukplusatstart),GotoIf($["${CALLERID(num):0:1}" != “+”]?noplusatstart)
exten => _X!,n,Set(CALLERID(num)=00${CALLERID(num):1})
exten => _X!,n(noplusatstart),Goto(from-pstn,${EXTEN},1)

I’m sure there are more elegant ways but this worked for me and I hope is useful for others.
Ed

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@bpatterson117 wrote:

Hello,

We are looking into integrating an old Simplex 5100 PA with our Current FreePBX system, I know this is not 100% FreePBX related but I figured I would consult some more experienced people here

So Our 5100 PA uses and admin station phone that you can Dial #10 in order to do an All-Page. Do any of you know the best way I could tie this into a SIP phone? It uses a proprietary amp so integrating directly with the amp with a SIP paging adapter is a no go for now.
I didn’t know if I could configure an ATA to auto answer the call and then send a DTMF tone over to initiate the page better yet if that’s possible is there way to confine that all to one button on a SIP phone. Almost like script the call? If that’s a thing.

We are looking to RIP this entire system out in the future however the speakers use a nonstandard voltage so that bumped it out of the budget range for now. So I am just looking for the best method to accomplish this.

Thanks!

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@qupfer wrote:

Hi, I want to use two inbound Routes. One default and one just for a single Number.
Let assume the single number is +49123456789.

I have two inbound routes.
DID/CID: ANY/ANY --> IVR
DID/CID ANY/+49123456789 --> DISA
but also the call from the single number is routed to IVR.

So, I thought ANY may bound all - regardless any other one configured. So I tried:
DID/CID: ANY/_X! --> IVR
DID/CID ANY/+49123456789 --> DISA

Now, my single Number is routed correctly, but all other does not work anymore. Probably because the + in the CID.
I used this wiki page https://wiki.freepbx.org/display/FPG/Inbound+Route+User+Guide and the linked https://wiki.freepbx.org/display/FPG/DIAL+PATTERN+INFO)

Thanks a lot
(Asterisk 16.4.1 & FreePBX 14.0.13.4)

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@TheFrenchFrog wrote:

Hello !

Since two weeks i’m trying to call with webrtc.

In this situation i wan’t to call my extentions “111” (3CX softphone) from the webRTC of “333”

problem.png1917×972 232 KB

_ _ _ ADVANCED SETTINGS _ _ _

mini-HTTP:

Advanced mini-HTTP.png931×344 20.6 KB

nodejs:

Advanced Nodejs.png903×224 14.7 KB

sip:

Advanced Sip.png922×343 18.4 KB

_ _ _ SIP SETTINGS _ _ _

general:

Sip general.png903×734 38 KB

pjsip:

Sip pjsip.png896×550 29.3 KB

sip:

Sip Sip.png897×920 52.5 KB

_ _ _ SYSTEM ADMIN _ _ _

https:

ports:

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@drandolph19 wrote:

I am new to the Freepbx environment but have found some really good help in this forum, I am currently working on upgrading our phone system from 2.11 on Cent OS 6.7 to Freepbx 14 Distro SNG7 Official, I utilized the Distro Conversion Tool (Elastix to PBXinaFlash to FreePBX Distro) have new server imaged with freepbx 14 and transferred everything over from the donor machine everything looks good. But as far as the cutover to the new system not sure what else is required, I know that I will have to swap IP Addresses for the Registar Server or should I just run them side by side and start transitioning people over to the new machine with new IP Address. Our environment isn’t that complicated we have about 50 Extensions with about 20 inbound routes and some daily schedules for call routing. Please help.

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@AndresAlvear wrote:

I am having an issue with my configuration. I am not able to call to external numbers.
I am using asterisk 13 with freepbx 14. I am using pjsip configuration.
I did some digging with “pjsip show history” and I found that my server sends the invitation to the sip server but after “trying” there is a message that says “502 bad gateway”. The message says:

SIP/2.0 502 Bad Gateway
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;rport=5060;received=XXX.XXX.XXX.XXX;branch=
From: “XXXXXX” sip:XXXXXXXX@XXX.XXX.XXX.XXX;tag=
To: sip:XXX.XXX.XXX.XXX@XXX.XXX.XXX.XXX;tag=gyodtxgc
Call-ID:
CSeq: 5465 INVITE
Warning: 399 37959.2432.P.261.5.105.0.4.32839.0.1075445762.ims.MY.PROVIDER.COM “Service lost”
Content-Length: 0
Content-Length: 0

The thing is that I did not change anything in my network configuration. Now it is working in a virtual machine and I have a snapshot with everything working, but I can not reload the configuration to make any change.

Thanks for your help

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@badmin wrote:

Hey guys, I hope you can help me out.
We have two FreePBX systems (A and B) that are connected over an IAX trunk.

• The main phone number arrives at System A (Let’s call it 555100100-1xx)
• External calls to System B should arrive at System A and routed over the IAX trunk
• System A uses extensions 100 to 149
• System B uses extensions 150 to 189
• Both systems can talk to each other on the internal extensions
• Both systems can do outbound calls
• External calls to System A is working
• External calls to System B is NOT working

I found a macro that should route all external calls to extension 150 to 189 over the iax trunk but i am not sure if this is acutally used by the system. This macro is a relic from the former asterisk system.

[from-did-direct] exten => _5551001001[5-8]X,1,Set(CALLERID(all)=00049${CALLERID(num):1}) exten => _5551001001[5-8]X,n,NoOP(+++++ Call to SystemB ++++++) exten => _5551001001[5-8]X,n,Wait(1) exten => _5551001001[5-8]X,n,Dial(IAX2/SystemB@IP/${EXTEN:9},30,tTw) exten => _5551001001[5-8]X,n,Busy(1) exten => _5551001001[5-8]X,n,Hangup()

Currently I believe the problem is the Inbound Route. What destination do I need to set there? At the moment it is set to the IAX Trunk.

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@mrjoe wrote:

I have FreePBX 15. If I try to delete a row inside an outbound rule, it looks like it’s deleted but if I go back in it still exists.

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@bonifacefj wrote:

I have recently noticed that I am unable to make outbound calls. I have had a flick through the logs and here are the warning and errors produced:

[2019-07-31 14:56:36] ERROR[8085] res_pjsip.c: Endpoint 'AAisp': Could not create dialog to invalid URI 'AAisp'. Is endpoint registered and reachable?
[2019-07-31 14:56:36] ERROR[8085] chan_pjsip.c: Failed to create outgoing session to endpoint 'AAisp'
[2019-07-31 14:56:36] WARNING[9859][C-00000011] app_dial.c: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)

I have been able to dial out intermittently between these faults so I’m not sure what this suggests.

From this I presume that PJSIP is unable to contact the provider. I have checked for correct resolution of DNS and avalibility but am not sure where to look next - am I working along the right lines or have I misunderstood the log entry?

I am using FreePBX 14.0.13.4 with Asterisk 13.22.0

Thanks
Fred.

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@WayneWenthin wrote:

I’ve been trying to get this system upgraded for a couple of years but could never get the customer to move until now. When I try to do any upgrade from the gui I get GPG errors. Even when trying from the console using the scripts on https://wiki.freepbx.org/display/PPS/FreePBX-Distro-5.211.65 I am seeing more gpg errors. After some research it looks like there is a mention of the GPG keys being poisoned. Is there no longer a path forward from this version to the current?

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@j3ven7 wrote:

I want to be able to check if my primary endpoint is up and running from DialPlan. If my primary is down I want to be able to find out so I can route traffic to a backup endpoint.

For testing, I have used an ip address I know won’t work, 1.1.1.1, I will be referring to this endpoint as the ‘broken primary’. The ‘backup’ endpoint, is an asterisk PBX I have verified works.

I setup OPTIONS pings for both the broken primary and backup, and as expected, I don’t receive a response from the broken primary, but I do receive a response from the working backup. So far so good.

I get the following message from the asterisk CLI to indicate that the broken primary is unreachable after the qualify timeout is reached, as expected.

– Contact X.X.X.X/sip:1.1.1.1:5060 is now Unreachable. RTT: 0.000 msec

The problem is, that I haven’t found a good way to check the qualify status of the ‘broken primary’ endpoint in my dial plan.

I did some research and saw that there was a SIPPEER function, so I tried to register the working backup endpoint as a peer by doing the following in my sip.conf:

[test-peer]
type=friend
context=phone
allow=ulaw,alaw
secret=1234
host=X.X.X.X ; in my conf this is a valid IP

But I get the following error when I run the asterisk CLI, indicating that the peer is unreachable.

[2019-07-31 14:24:48] NOTICE[5691]: chan_sip.c:30181 sip_poke_noanswer: Peer ‘test-peer’ is now UNREACHABLE! Last qualify: 0

So essentially I am asking a few things:

1. Is there an analogous function to SIPPEER for endpoints in dial plan? I didn’t find one after extensive search in the documentation, but I could be wrong. Note that I specifically need something that would tell me if an endpoint is up i.e. the “status” item of SIPPEER as I understand it.

2. Where is the following message logged:
   – Contact X.X.X.X/sip:1.1.1.1:5060 is now Unreachable. RTT: 0.000 msec
   Perhaps I can check this log and do some parsing to leverage it for my purposes, but I was unable to locate its source.

3. Is it even possible in asterisk to determine the status of a remote endpoint using options pings?

*** This is my first post and I am fairly new to asterisk so excuse any missteps in jargon.

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@jchuchla wrote:

I recently went in to add an extension for the first time in a long time and found that the new user wasn’t available to add to the extension. I checked the Directories tab in User Management and under the Active Directory entry, I see it’s throwing an error at the bottom that says “Invalid Credentials” I haven’t made any changes to the settings in here, nor with the user in AD since I first set it up a few years ago. I double checked the username and password and it is indeed valid in my AD.

I also noticed that the Active Directory module is now marked “(legacy)” Is that perhaps part of the issue? is the old one no longer functional?

I tried using the command: “fwconsole userman --syncall --force” and get the same “Invalid Credentials” error in a red bar.

I also tried changing the user/pass to my own account credentials which I know are good and that also gives the same errors.

I have no idea when this broke. I haven’t made any user changes in my system for the past 2 years, so it could have been this way for a while. But it did work when I first configured it.

Any suggestions where to go with it next?

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@sorvani wrote:

So the carrier Skyetel is adding a tenant functionality that I would like to make use of on one PBX (not actually multi-tenant).

I want to use this function to make use of the built in cost tools in Skyetel’s portal to get departmental usage stats.

To enable outbound calls to be linked correctly they want a SIP header added.

X-Tenant: somename.to.match

A search here led me to a few older posts that led me to this external post

The working bit of that post is Set(HASH(__SIPHEADERS,X-DID)=${CDR(did)})) to update the DID.

1. Would it be the same process to ADD a header? Set(HASH(__SIPHEADERS,X-Tenant)=somename.to.match))
2. Would I also need to hook into the trunk predial macro?

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@madmacka wrote:

Hi , I’m having issues with incoming call ring tone to the caller , when i dial in from the outside world (PTSN) the call is very slow to start the ringing tone to the external phone signifying receiving hand sets are ringing. however the phones with in the pbx are very quick to respond and are actually ringing whilst the phone on the ptsn side just sounds like a dead line this goes on for at very least 5 seconds or so. to the point where the incoming caller just thinks its not working and gives up.

does any one have any idea what would be causing this. is there a setting or something in freepbx that could be resposible? my gut feeling is that free pbx isn’t sending some ACK to my provider in an a acceptable time to siginfy that the end points are actualying ringing. thus not relaying the ringing tone to the incoming caller on the PTSN side?

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@gettrashed wrote:

I have Newest FreePbx and i have call center. so my question is if my call center all extensions are busy and inbound call came and status is no answear after this number(previus inbound call) called again i want to give him priority that it will go as first inbound call and extension will answear it can you help me? it is kinda priority calls

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@jenser wrote:

Hello everybody,

we are using FreePBX 14.0.11 in a virtual environment. Endpoints are 1 gigaset Pro N720 DECT Manager with 10 base stations and 60 handsets as well as around 10 gigaset Maxwell basic and Maxwell 3 phones. Last night, all endpoints have logged off again. What could be the cause?

[2019-08-01 01:56:23] VERBOSE[30320] pbx_variables.c: Setting global variable 'SIPDOMAIN' to 
'192.77.6.208'
[2019-08-01 01:56:23] VERBOSE[30320] netsock2.c: Using SIP RTP Audio TOS bits 184
[2019-08-01 01:56:23] VERBOSE[30320] netsock2.c: Using SIP RTP Audio TOS bits 184 in 
TCLASS field.
[2019-08-01 01:56:23] VERBOSE[30320] netsock2.c: Using SIP RTP Audio CoS mark 5
[2019-08-01 01:56:57] VERBOSE[30321] pbx_variables.c: Setting global variable 'SIPDOMAIN' to 
'192.77.6.208'
[2019-08-01 01:56:57] VERBOSE[30321] netsock2.c: Using SIP RTP Audio TOS bits 184
[2019-08-01 01:56:57] VERBOSE[30321] netsock2.c: Using SIP RTP Audio TOS bits 184 in 
TCLASS field.
[2019-08-01 01:56:57] VERBOSE[30321] netsock2.c: Using SIP RTP Audio CoS mark 5
[2019-08-01 01:57:19] VERBOSE[31758] pbx_variables.c: Setting global variable 'SIPDOMAIN' to 
'192.77.6.208'
[2019-08-01 01:57:19] VERBOSE[31758] netsock2.c: Using SIP RTP Audio TOS bits 184
[2019-08-01 01:57:19] VERBOSE[31758] netsock2.c: Using SIP RTP Audio TOS bits 184 in 
TCLASS field.
[2019-08-01 01:57:19] VERBOSE[31758] netsock2.c: Using SIP RTP Audio CoS mark 5
[2019-08-01 01:57:29] VERBOSE[17587] res_pjsip_registrar.c: Removed contact 
'sip:84@192.77.4.101:5060' from AOR '84' due to expiration
[2019-08-01 01:57:29] VERBOSE[5139] res_pjsip/pjsip_options.c: Contact 
84/sip:84@192.77.4.101:5060 has been deleted
[2019-08-01 01:57:29] VERBOSE[5139] res_pjsip/pjsip_configuration.c: Endpoint 84 is now 
Unreachable
[2019-08-01 01:57:59] VERBOSE[17587] res_pjsip_registrar.c: Removed contact 
'sip:36@192.77.4.101:5060' from AOR '36' due to expiration
[2019-08-01 01:57:59] VERBOSE[5139] res_pjsip/pjsip_options.c: Contact 
36/sip:36@192.77.4.101:5060 has been deleted
[2019-08-01 01:57:59] VERBOSE[5139] res_pjsip/pjsip_configuration.c: Endpoint 36 is now 
Unreachable
[2019-08-01 01:58:03] VERBOSE[4967] res_pjsip_registrar.c: Added contact 
'sip:84@192.77.4.101:5060' to AOR '84' with expiration of 180 seconds
[2019-08-01 01:58:03] VERBOSE[4967] res_pjsip/pjsip_configuration.c: Endpoint 84 is now 
Reachable
[2019-08-01 01:58:03] VERBOSE[4967] res_pjsip/pjsip_options.c: Contact 
84/sip:84@192.77.4.101:5060 is now Reachable.  RTT: 30.322 msec
[2019-08-01 01:58:23] VERBOSE[5139] res_pjsip/pjsip_options.c: Contact 
71/sip:71@192.77.4.101:5060 has been deleted
[2019-08-01 01:58:23] VERBOSE[5139] res_pjsip/pjsip_configuration.c: Endpoint 71 is now 
Unreachable
[2019-08-01 01:58:23] VERBOSE[5139] res_pjsip/pjsip_options.c: Contact 
82/sip:82@192.77.4.101:5060 has been deleted
[2019-08-01 01:58:23] VERBOSE[5139] res_pjsip/pjsip_configuration.c: Endpoint 82 is now 
Unreachable
[2019-08-01 01:58:23] VERBOSE[5139] res_pjsip/pjsip_options.c: Contact 
91/sip:91@192.77.4.101:5060 has been deleted
[2019-08-01 01:58:23] VERBOSE[5139] res_pjsip/pjsip_configuration.c: Endpoint 91 is now 
Unreachable
[2019-08-01 01:58:29] VERBOSE[17587] res_pjsip_registrar.c: Removed contact 
'sip:39@192.77.4.101:5060' from AOR '39' due to expiration
[2019-08-01 01:58:29] VERBOSE[5139] res_pjsip/pjsip_options.c: Contact 
39/sip:39@192.77.4.101:5060 has been deleted
[2019-08-01 01:58:29] VERBOSE[5139] res_pjsip/pjsip_configuration.c: Endpoint 39 is now 
Unreachable
[2019-08-01 01:58:30] VERBOSE[4967] res_pjsip_registrar.c: Added contact 
'sip:36@192.77.4.101:5060' to AOR '36' with expiration of 180 seconds
[2019-08-01 01:58:30] VERBOSE[4967] res_pjsip/pjsip_configuration.c: Endpoint 36 is now 
Reachable
[2019-08-01 01:58:30] VERBOSE[4967] res_pjsip/pjsip_options.c: Contact 
36/sip:36@192.77.4.101:5060 is now Reachable.  RTT: 30.407 msec
[2019-08-01 01:58:40] VERBOSE[4967] res_pjsip_registrar.c: Added contact 
'sip:71@192.77.4.101:5060' to AOR '71' with expiration of 180 seconds
[2019-08-01 01:58:40] VERBOSE[4967] res_pjsip/pjsip_configuration.c: Endpoint 71 is now 
Reachable
[2019-08-01 01:58:40] VERBOSE[4967] res_pjsip/pjsip_options.c: Contact 
71/sip:71@192.77.4.101:5060 is now Reachable.  RTT: 30.645 msec
[2019-08-01 01:58:46] VERBOSE[4967] res_pjsip_registrar.c: Added contact 
'sip:91@192.77.4.101:5060' to AOR '91' with expiration of 180 seconds
[2019-08-01 01:58:46] VERBOSE[4967] res_pjsip/pjsip_configuration.c: Endpoint 91 is now 
Reachable
[2019-08-01 01:58:46] VERBOSE[4967] res_pjsip/pjsip_options.c: Contact 
91/sip:91@192.77.4.101:5060 is now Reachable.  RTT: 30.493 msec
[2019-08-01 01:58:56] VERBOSE[4967] res_pjsip_registrar.c: Added contact 
'sip:82@192.77.4.101:5060' to AOR '82' with expiration of 180 seconds
[2019-08-01 01:58:56] VERBOSE[4967] res_pjsip/pjsip_configuration.c: Endpoint 82 is now 
Reachable
[2019-08-01 01:58:56] VERBOSE[4967] res_pjsip/pjsip_options.c: Contact 
82/sip:82@192.77.4.101:5060 is now Reachable.  RTT: 42.324 msec
[2019-08-01 01:58:59] VERBOSE[17587] res_pjsip_registrar.c: Removed contact 
'sip:21@192.77.4.101:5060' from AOR '21' due to expiration
[2019-08-01 01:58:59] VERBOSE[17587] res_pjsip_registrar.c: Removed contact 
'sip:95@192.77.4.101:5060' from AOR '95' due to expiration
[2019-08-01 01:58:59] VERBOSE[5139] res_pjsip/pjsip_options.c: Contact 
21/sip:21@192.77.4.101:5060 has been deleted
[2019-08-01 01:58:59] VERBOSE[5139] res_pjsip/pjsip_configuration.c: Endpoint 21 is now 
Unreachable
[2019-08-01 01:58:59] VERBOSE[5139] res_pjsip/pjsip_options.c: Contact 
95/sip:95@192.77.4.101:5060 has been deleted
[2019-08-01 01:58:59] VERBOSE[5139] res_pjsip/pjsip_configuration.c: Endpoint 95 is now 
Unreachable
[2019-08-01 01:59:02] VERBOSE[4967] res_pjsip_registrar.c: Added contact 
'sip:39@192.77.4.101:5060' to AOR '39' with expiration of 180 seconds
[2019-08-01 01:59:03] VERBOSE[4967] res_pjsip/pjsip_configuration.c: Endpoint 39 is now 
Reachable
[2019-08-01 01:59:03] VERBOSE[4967] res_pjsip/pjsip_options.c: Contact 
39/sip:39@192.77.4.101:5060 is now Reachable.  RTT: 44.106 msec
[2019-08-01 01:59:14] VERBOSE[5139] res_pjsip/pjsip_options.c: Contact 
63/sip:63@192.77.4.101:5060 has been deleted
[2019-08-01 01:59:14] VERBOSE[5139] res_pjsip/pjsip_configuration.c: Endpoint 63 is now 
Unreachable
[2019-08-01 01:59:26] VERBOSE[4967] res_pjsip_registrar.c: Added contact             
'sip:21@192.77.4.101:5060' to AOR '21' with expiration of 180 seconds
[2019-08-01 01:59:26] VERBOSE[4967] res_pjsip/pjsip_configuration.c: Endpoint 21 is now     
Reachable
[2019-08-01 01:59:26] VERBOSE[4967] res_pjsip/pjsip_options.c: Contact 
21/sip:21@192.77.4.101:5060 is now Reachable.  RTT: 32.153 msec
[2019-08-01 01:59:36] VERBOSE[4967] res_pjsip_registrar.c: Added contact 
'sip:63@192.77.4.101:5060' to AOR '63' with expiration of 180 seconds
[2019-08-01 01:59:36] VERBOSE[4967] res_pjsip/pjsip_configuration.c: Endpoint 63 is now 
Reachable
[2019-08-01 01:59:36] VERBOSE[4967] res_pjsip/pjsip_options.c: Contact 
63/sip:63@192.77.4.101:5060 is now Reachable.  RTT: 30.526 msec
[2019-08-01 01:59:49] VERBOSE[5139] res_pjsip/pjsip_options.c: Contact 
14/sip:14@192.77.4.101:5060 has been deleted
[2019-08-01 01:59:49] VERBOSE[5139] res_pjsip/pjsip_configuration.c: Endpoint 14 is now 
Unreachable
[2019-08-01 01:59:49] VERBOSE[5139] res_pjsip/pjsip_options.c: Contact 
96/sip:96@192.77.4.101:5060 has been deleted
[2019-08-01 01:59:49] VERBOSE[5139] res_pjsip/pjsip_configuration.c: Endpoint 96 is now 
Unreachable
[2019-08-01 02:00:23] VERBOSE[4967] res_pjsip_registrar.c: Added contact 
'sip:96@192.77.4.101:5060' to AOR '96' with expiration of 180 seconds
[2019-08-01 02:00:23] VERBOSE[4967] res_pjsip/pjsip_configuration.c: Endpoint 96 is now 
Reachable
[2019-08-01 02:00:23] VERBOSE[4967] res_pjsip/pjsip_options.c: Contact 
96/sip:96@192.77.4.101:5060 is now Reachable.  RTT: 35.980 msec
[2019-08-01 02:00:24] VERBOSE[4967] res_pjsip_registrar.c: Added contact 
'sip:14@192.77.4.101:5060' to AOR '14' with expiration of 180 seconds
[2019-08-01 02:00:24] VERBOSE[4967] res_pjsip/pjsip_configuration.c: Endpoint 14 is now 
Reachable
[2019-08-01 02:00:24] VERBOSE[4967] res_pjsip/pjsip_options.c: Contact 
14/sip:14@192.77.4.101:5060 is now Reachable.  RTT: 48.871 msec
[2019-08-01 02:00:25] VERBOSE[5139] res_pjsip/pjsip_options.c: Contact 
48/sip:48@192.77.4.101:5060 has been deleted
[2019-08-01 02:00:25] VERBOSE[5139] res_pjsip/pjsip_configuration.c: Endpoint 48 is now 
Unreachable
[2019-08-01 02:00:29] VERBOSE[17587] res_pjsip_registrar.c: Removed contact 
'sip:11@192.77.4.126:5060' from AOR '11' due to expiration
[2019-08-01 02:00:29] VERBOSE[5139] res_pjsip/pjsip_options.c: Contact 
11/sip:11@192.77.4.126:5060 has been deleted
[2019-08-01 02:00:29] VERBOSE[5139] res_pjsip/pjsip_configuration.c: Endpoint 11 is now 
Unreachable
[2019-08-01 02:00:47] VERBOSE[4967] res_pjsip_registrar.c: Added contact 
'sip:48@192.77.4.101:5060' to AOR '48' with expiration of 180 seconds
[2019-08-01 02:00:47] VERBOSE[4967] res_pjsip/pjsip_configuration.c: Endpoint 48 is now 
Reachable
[2019-08-01 02:00:47] VERBOSE[4967] res_pjsip/pjsip_options.c: Contact 
48/sip:48@192.77.4.101:5060 is now Reachable.  RTT: 30.408 msec
[2019-08-01 02:00:57] VERBOSE[4967] res_pjsip_registrar.c: Added contact 
'sip:95@192.77.4.101:5060' to AOR '95' with expiration of 180 seconds
[2019-08-01 02:00:57] VERBOSE[4967] res_pjsip/pjsip_configuration.c: Endpoint 95 is now 
Reachable
[2019-08-01 02:00:57] VERBOSE[4967] res_pjsip/pjsip_options.c: Contact 
95/sip:95@192.77.4.101:5060 is now Reachable.  RTT: 30.583 msec
[2019-08-01 02:00:59] VERBOSE[17587] res_pjsip_registrar.c: Removed contact 
'sip:98@192.77.4.130:5060' from AOR '98' due to expiration
[2019-08-01 02:00:59] VERBOSE[5139] res_pjsip/pjsip_options.c: Contact 
98/sip:98@192.77.4.130:5060 has been deleted
[2019-08-01 02:00:59] VERBOSE[5139] res_pjsip/pjsip_configuration.c: Endpoint 98 is now 
Unreachable
[2019-08-01 02:02:29] VERBOSE[17587] res_pjsip_registrar.c: Removed contact 
'sip:40@192.77.4.101:5060' from AOR '40' due to expiration
[2019-08-01 02:02:29] VERBOSE[5139] res_pjsip/pjsip_options.c: Contact 
40/sip:40@192.77.4.101:5060 has been deleted
[2019-08-01 02:02:29] VERBOSE[5139] res_pjsip/pjsip_configuration.c: Endpoint 40 is now 
Unreachable
[2019-08-01 02:02:47] VERBOSE[5139] res_pjsip/pjsip_options.c: Contact 
47/sip:47@192.77.4.101:5060 has been deleted
[2019-08-01 02:02:47] VERBOSE[5139] res_pjsip/pjsip_configuration.c: Endpoint 47 is now 
Unreachable
[2019-08-01 02:02:47] VERBOSE[5139] res_pjsip/pjsip_options.c: Contact 
63/sip:63@192.77.4.101:5060 has been deleted
[2019-08-01 02:02:47] VERBOSE[5139] res_pjsip/pjsip_configuration.c: Endpoint 63 is now 
Unreachable
[2019-08-01 02:03:02] VERBOSE[4967] res_pjsip_registrar.c: Added contact 
'sip:40@192.77.4.101:5060' to AOR '40' with expiration of 180 seconds
[2019-08-01 02:03:02] VERBOSE[4967] res_pjsip/pjsip_configuration.c: Endpoint 40 is now 
Reachable
[2019-08-01 02:03:02] VERBOSE[4967] res_pjsip/pjsip_options.c: Contact 
40/sip:40@192.77.4.101:5060 is now Reachable.  RTT: 31.921 msec
[2019-08-01 02:03:12] VERBOSE[4967] res_pjsip_registrar.c: Added contact 
'sip:63@192.77.4.101:5060' to AOR '63' with expiration of 180 seconds
[2019-08-01 02:03:12] VERBOSE[4967] res_pjsip/pjsip_configuration.c: Endpoint 63 is now 
Reachable
[2019-08-01 02:03:12] VERBOSE[4967] res_pjsip/pjsip_options.c: Contact 
63/sip:63@192.77.4.101:5060 is now Reachable.  RTT: 32.296 msec
[2019-08-01 02:03:13] VERBOSE[4967] res_pjsip_registrar.c: Added contact 
'sip:47@192.77.4.101:5060' to AOR '47' with expiration of 180 seconds
[2019-08-01 02:03:13] VERBOSE[4967] res_pjsip/pjsip_configuration.c: Endpoint 47 is now 
Reachable
[2019-08-01 02:03:13] VERBOSE[4967] res_pjsip/pjsip_options.c: Contact 
47/sip:47@192.77.4.101:5060 is now Reachable.  RTT: 31.903 msec
[2019-08-01 02:03:22] VERBOSE[5139] res_pjsip/pjsip_options.c: Contact 
70/sip:70@192.77.4.101:5060 has been deleted
[2019-08-01 02:03:22] VERBOSE[5139] res_pjsip/pjsip_configuration.c: Endpoint 70 is now 
Unreachable
[2019-08-01 02:03:29] VERBOSE[17587] res_pjsip_registrar.c: Removed contact 
'sip:34@192.77.4.101:5060' from AOR '34' due to expiration
[2019-08-01 02:03:29] VERBOSE[5139] res_pjsip/pjsip_options.c: Contact 
34/sip:34@192.77.4.101:5060 has been deleted
[2019-08-01 02:03:41] VERBOSE[4967] res_pjsip_registrar.c: Added contact 
'sip:70@192.77.4.101:5060' to AOR '70' with expiration of 180 seconds
[2019-08-01 02:03:41] VERBOSE[4967] res_pjsip/pjsip_configuration.c: Endpoint 70 is now 
Reachable
[2019-08-01 02:03:41] VERBOSE[4967] res_pjsip/pjsip_options.c: Contact 
70/sip:70@192.77.4.101:5060 is now Reachable.  RTT: 29.907 msec
[2019-08-01 02:03:58] VERBOSE[5139] res_pjsip/pjsip_options.c: Contact 
77/sip:77@192.77.4.101:5060 has been deleted
[2019-08-01 02:03:58] VERBOSE[5139] res_pjsip/pjsip_configuration.c: Endpoint 77 is now 
Unreachable
[2019-08-01 02:03:58] VERBOSE[5139] res_pjsip/pjsip_options.c: Contact 
20/sip:20@192.77.4.101:5060 has been deleted
[2019-08-01 02:03:58] VERBOSE[5139] res_pjsip/pjsip_configuration.c: Endpoint 20 is now 
Unreachable
[2019-08-01 02:04:00] VERBOSE[4967] res_pjsip_registrar.c: Added contact 
'sip:34@192.77.4.101:5060' to AOR '34' with expiration of 180 seconds
[2019-08-01 02:04:01] VERBOSE[4967] res_pjsip/pjsip_options.c: Contact 
34/sip:34@192.77.4.101:5060 is now Reachable.  RTT: 21.017 msec
[2019-08-01 02:04:06] VERBOSE[4967] res_pjsip_registrar.c: Added contact 
'sip:77@192.77.4.101:5060' to AOR '77' with expiration of 180 seconds
[2019-08-01 02:04:06] VERBOSE[4967] res_pjsip/pjsip_configuration.c: Endpoint 77 is now 
Reachable
[2019-08-01 02:04:06] VERBOSE[4967] res_pjsip/pjsip_options.c: Contact 
77/sip:77@192.77.4.101:5060 is now Reachable.  RTT: 42.057 msec
[2019-08-01 02:04:28] VERBOSE[4967] res_pjsip_registrar.c: Added contact 
'sip:11@192.77.4.126:5060' to AOR '11' with expiration of 180 seconds
[2019-08-01 02:04:28] VERBOSE[4967] res_pjsip/pjsip_configuration.c: Endpoint 11 is now 
Reachable
[2019-08-01 02:04:28] VERBOSE[4967] res_pjsip/pjsip_options.c: Contact 
11/sip:11@192.77.4.126:5060 is now Reachable.  RTT: 512.184 msec
[2019-08-01 02:04:29] VERBOSE[5139] res_pjsip/pjsip_options.c: Contact 
97/sip:97@192.77.4.101:5060 has been deleted
[2019-08-01 02:04:29] VERBOSE[5139] res_pjsip/pjsip_configuration.c: Endpoint 97 is now 
Unreachable
[2019-08-01 02:04:30] VERBOSE[4967] res_pjsip_registrar.c: Added contact 
'sip:20@192.77.4.101:5060' to AOR '20' with expiration of 180 seconds
[2019-08-01 02:04:30] VERBOSE[4967] res_pjsip/pjsip_configuration.c: Endpoint 20 is now 
Reachable
[2019-08-01 02:04:30] VERBOSE[4967] res_pjsip/pjsip_options.c: Contact 
20/sip:20@192.77.4.101:5060 is now Reachable.  RTT: 30.403 msec
[2019-08-01 02:04:59] VERBOSE[17587] res_pjsip_registrar.c: Removed contact 
'sip:18@192.77.4.101:5060' from AOR '18' due to expiration
[2019-08-01 02:04:59] VERBOSE[5139] res_pjsip/pjsip_options.c: Contact 
18/sip:18@192.77.4.101:5060 has been deleted
[2019-08-01 02:04:59] VERBOSE[5139] res_pjsip/pjsip_configuration.c: Endpoint 18 is now 
Unreachable
[2019-08-01 02:05:03] WARNING[17452] taskprocessor.c: The 'pjsip/distributor-00000086' task 
processor queue reached 500 scheduled tasks.
[2019-08-01 02:05:29] VERBOSE[17587] res_pjsip_registrar.c: Removed contact 
'sip:28@192.77.4.101:5060' from AOR '28' due to expiration
[2019-08-01 02:05:29] VERBOSE[5139] res_pjsip/pjsip_options.c: Contact 
28/sip:28@192.77.4.101:5060 has been deleted
[2019-08-01 02:05:29] VERBOSE[5139] res_pjsip/pjsip_configuration.c: Endpoint 28 is now 
Unreachable
[2019-08-01 02:05:44] VERBOSE[5139] res_pjsip/pjsip_options.c: Contact 
42/sip:42@192.77.4.101:5060 has been deleted
[2019-08-01 02:05:44] VERBOSE[5139] res_pjsip/pjsip_configuration.c: Endpoint 42 is now 
Unreachable
[2019-08-01 02:05:59] VERBOSE[17587] res_pjsip_registrar.c: Removed contact 
'sip:12@192.77.4.101:5060' from AOR '12' due to expiration
[2019-08-01 02:05:59] VERBOSE[17587] res_pjsip_registrar.c: Removed contact 
'sip:32@192.77.4.101:5060' from AOR '32' due to expiration
[2019-08-01 02:05:59] VERBOSE[17587] res_pjsip_registrar.c: Removed contact 
'sip:56@192.77.4.101:5060' from AOR '56' due to expiration
[2019-08-01 02:05:59] VERBOSE[17587] res_pjsip_registrar.c: Removed contact 
'sip:75@192.77.4.101:5060' from AOR '75' due to expiration
[2019-08-01 02:05:59] VERBOSE[17587] res_pjsip_registrar.c: Removed contact 
'sip:92@192.77.4.101:5060' from AOR '92' due to expiration
[2019-08-01 02:05:59] VERBOSE[5139] res_pjsip/pjsip_options.c: Contact 
12/sip:12@192.77.4.101:5060 has been deleted
[2019-08-01 02:05:59] VERBOSE[5139] res_pjsip/pjsip_configuration.c: Endpoint 12 is now 
Unreachable
[2019-08-01 02:05:59] VERBOSE[5139] res_pjsip/pjsip_options.c: Contact 
32/sip:32@192.77.4.101:5060 has been deleted
[2019-08-01 02:05:59] VERBOSE[5139] res_pjsip/pjsip_configuration.c: Endpoint 32 is now 
Unreachable
[2019-08-01 02:05:59] VERBOSE[5139] res_pjsip/pjsip_options.c: Contact 
56/sip:56@192.77.4.101:5060 has been deleted
[2019-08-01 02:05:59] VERBOSE[5139] res_pjsip/pjsip_configuration.c: Endpoint 56 is now 
Unreachable
[2019-08-01 02:05:59] VERBOSE[5139] res_pjsip/pjsip_options.c: Contact 
75/sip:75@192.77.4.101:5060 has been deleted
[2019-08-01 02:05:59] VERBOSE[5139] res_pjsip/pjsip_configuration.c: Endpoint 75 is now 
Unreachable
[2019-08-01 02:05:59] VERBOSE[5139] res_pjsip/pjsip_options.c: Contact 
92/sip:92@192.77.4.101:5060 has been deleted
[2019-08-01 02:05:59] VERBOSE[5139] res_pjsip/pjsip_configuration.c: Endpoint 92 is now 
Unreachable
[2019-08-01 02:06:20] VERBOSE[5139] res_pjsip/pjsip_options.c: Contact 
40/sip:40@192.77.4.101:5060 has been deleted
[2019-08-01 02:06:20] VERBOSE[5139] res_pjsip/pjsip_configuration.c: Endpoint 40 is now 
Unreachable
[2019-08-01 02:06:20] VERBOSE[5139] res_pjsip/pjsip_options.c: Contact 
36/sip:36@192.77.4.101:5060 has been deleted
[2019-08-01 02:06:20] VERBOSE[5139] res_pjsip/pjsip_configuration.c: Endpoint 36 is now 
Unreachable
[2019-08-01 02:06:20] VERBOSE[5139] res_pjsip/pjsip_options.c: Contact 
47/sip:47@192.77.4.101:5060 has been deleted
[2019-08-01 02:06:20] VERBOSE[5139] res_pjsip/pjsip_configuration.c: Endpoint 47 is now 
Unreachable
[2019-08-01 02:06:20] VERBOSE[5139] res_pjsip/pjsip_options.c: Contact 
48/sip:48@192.77.4.101:5060 has been deleted
[2019-08-01 02:06:20] VERBOSE[5139] res_pjsip/pjsip_configuration.c: Endpoint 48 is now 
Unreachable
[2019-08-01 02:06:20] VERBOSE[5139] res_pjsip/pjsip_options.c: Contact 
811/sip:811@192.77.4.101:5060 has been deleted
[2019-08-01 02:06:20] VERBOSE[5139] res_pjsip/pjsip_configuration.c: Endpoint 811 is now 
Unreachable
[2019-08-01 02:06:20] VERBOSE[5139] res_pjsip/pjsip_options.c: Contact 
88/sip:88@192.77.4.101:5060 has been deleted
[2019-08-01 02:06:20] VERBOSE[5139] res_pjsip/pjsip_configuration.c: Endpoint 88 is now 
Unreachable
[2019-08-01 02:06:20] VERBOSE[5139] res_pjsip/pjsip_options.c: Contact 
71/sip:71@192.77.4.101:5060 has been deleted
[2019-08-01 02:06:20] VERBOSE[5139] res_pjsip/pjsip_configuration.c: Endpoint 71 is now 
Unreachable
[2019-08-01 02:06:20] VERBOSE[5139] res_pjsip/pjsip_options.c: Contact 
57/sip:57@192.77.4.101:5060 has been deleted
[2019-08-01 02:06:20] VERBOSE[5139] res_pjsip/pjsip_options.c: Contact 
95/sip:95@192.77.4.101:5060 has been deleted
[2019-08-01 02:06:20] VERBOSE[5139] res_pjsip/pjsip_configuration.c: Endpoint 95 is now 
Unreachable
[2019-08-01 02:06:20] VERBOSE[5139] res_pjsip/pjsip_options.c: Contact 
63/sip:63@192.77.4.101:5060 has been deleted
[2019-08-01 02:06:20] VERBOSE[5139] res_pjsip/pjsip_configuration.c: Endpoint 63 is now 
Unreachable
[2019-08-01 02:06:20] VERBOSE[5139] res_pjsip/pjsip_options.c: Contact 
91/sip:91@192.77.4.101:5060 has been deleted
[2019-08-01 02:06:20] VERBOSE[5139] res_pjsip/pjsip_configuration.c: Endpoint 91 is now 
Unreachable

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