@jtomelevage wrote:

Hello,

We want to force the Outbound Route CID instead of the Trunk CID. How can we do this?

Thanks,
John

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@nebulatic wrote:

Hi,
I’m fairly new to FreePBX, but I have a need to remove four digits from a dialed extension in order for it to be a valid extension and although I can do this simply in outbound routes, I’m confused on how to do that with internal extensions.

First, the reason I had phones adding a prefix is in order to determine the physical location of the phone and therefor the outbound DID and the correct physical address (for E911). We have a public login system where users can log in to their phone with their extension and passcode, at any phone, at any location, and so the phone itself, not the extension, needs to determine location using the prefix.

I’ve just enacted this setup in order to deal with the new E911 rules, which as I say works wonderfully for outbound calls, but unexpectedly broke the internal calls. I should also add that the four digital prefix is obviously going to change depending on the location, so the prefix removal needs to be wildcard for at least one of the digits.

Help anyone?

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@MDTechTeam wrote:

So i just installed Asterisk and FreePBX. I am also using Zoiper and trying to get 2 computers to dial each other.

Only thing apart from installing the system i have done is open up ports 5060 and 5061 to point to the Asterisk server.

My Asterisk server is on a VM on the main server.

In zoiper it looks like i register. My login has a green check and i have an unregister option in the settings. This is for both users i have set up. However, i cannot call either computer.

I also have been unable to use the extension to create an extension or user that zoiper can connect with. The only way i was able to get Zoiper to connect to the system was mofiying the config files inside Admin->ConfigEdit

is there some simple step 1 or 2 i have missed in all the how-to guides i have read for quick setups? below are my mods to the conf files. i also did the fwconsole restart a couple times

inside extensions.conf
[from-internal]
exten=1000,1,Dial(SIP/user1,10)
exten=1000,1,Dial(SIP/user2,10)

inside pjsip.conf
[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0

;Templates for the necessary config sections

[endpoint_internal](!)
type=endpoint
context=from-internal
disallow=all
allow=ulaw

[auth_userpass](!)
type=auth
auth_type=userpass

[aor_dynamic](!)
type=aor
max_contacts=1

;Definitions for our phones, using the templates above
[user1](endpoint_internal)
auth=user1
aors=user1
[user1](auth_userpass)
password=badpassword
username=user1
[user1](aor_dynamic)

[user2](endpoint_internal)
auth=user2
aors=user2
[user2](auth_userpass)
password=badpassword
username=user2
[user2](aor_dynamic)

and finally inside sip.conf
[general]
transport=udp

[friends_internal](!)
type=friend
host=dynamic
context=from-internal
disallow=allow
allow=ulaw

[user1](friends_internal)
secret=badpassword

[user2](friends_internal)
secret=badpassword

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@victorcasale wrote:

Dear friends, I’m running FreePBX 15.0.16.20 in a local machine.
I have created 4 PJSIP extensions under setup page and i’m using a Grandstrem HT814 where all extensions are running. I can call any extension with no problems.

I’m just concerned with this behavor:
My extensions numbers are 201, 202, 203, 204

If extension number 201 (Robert) calls extension 202(Jhon) , both calling parties can listen and talk perfeclty. Robert was wishing to call jhon, but he wasn’t in the room and alex placed the call. Alex know that jhon is in the next room where the extension is 204. In my opinion, in a normal situation, when Alex press flash key, Robert would listen to hold music, when Alex would dialing to extension 204 and sayng to jhon that Robert wishes to talk to him, so when Alex puts his phone on hook, the connection between robert and jhoh, now on extension 204 would be stabilished. It’s not happening to me. When Alex presses flash key, both Robert and Alex begin listenin a dial tone, so when alex dial 204, both listen calling tone and when jhon picks up the extension 204 both listen the conversation, like in a conference.
Can someone please help me on how solving it ?

Many thanks

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@pbsolutions wrote:

I am currently rebuilding my FreePBX (that has the Cisco patch) that has been running for several years. I have FreePBX setup, phones registered (7800 and 8800 series), trunks configured - all is well there!

But, I’m having trouble getting the patch to install. I’ve taken the patching documentation from the FreePBX wiki and updated the versions and dead links and I’m left with this:

yum groupinstall "Development Tools"
yum install sangoma-devel
mkdir -p ~/rpmbuild/{BUILD,RPMS,SOURCES,SPECS,SRPMS}
echo '%_topdir %(echo $HOME)/rpmbuild' > ~/.rpmmacros

asterisk-version-switch

cd ~/rpmbuild/SOURCES
yumdownloader --source asterisk13
wget http://download.opensuse.org/source/tumbleweed/repo/oss/src/mISDNuser-2.1.0+2.0.22+git6-1.1.src.rpm
wget http://usecallmanager.nz/includes/cisco-usecallmanager-13.30.0.patch

rpm --nomd5 -ivh ~/rpmbuild/SOURCES/asterisk13-13.29.2-1.sng7.src.rpm
rpm --nomd5 -ivh ~/rpmbuild/SOURCES/mISDNuser-2.1.0+2.0.22+git6-1.1.src.rpm
yum-builddep ~/rpmbuild/SPECS/mISDNuser.spec 
rpmbuild -bp ~/rpmbuild/SPECS/mISDNuser.spec
rpmbuild -ba ~/rpmbuild/SPECS/mISDNuser.spec
rpm -Uvh ~/rpmbuild/RPMS/x86_64/libmisdn1-*.rpm
rpm -Uvh ~/rpmbuild/RPMS/x86_64/mISDNuser*.rpm
yum-builddep ~/rpmbuild/SPECS/asterisk13.spec
rpmbuild -bp ~/rpmbuild/SPECS/asterisk13.spec
nano ~/rpmbuild/SPECS/asterisk13.spec
	[Add this line below the Patch9 entry at line 27]
	Patch10: cisco-usecallmanager-13.30.0.patch
	[Add this below the patch8 entry at line 612]
	%patch10 -p1
	[save the file and close the editor]
rpmbuild -bp ~/rpmbuild/SPECS/asterisk13.spec
rpmbuild -ba ~/rpmbuild/SPECS/asterisk13.spec
rpm -Uvh ~/rpmbuild/RPMS/x86_64/asterisk13-28*.rpm --force
rpm -Uvh ~/rpmbuild/RPMS/x86_64/asterisk13-[n-r]*.rpm --force
rpm -Uvh ~/rpmbuild/RPMS/x86_64/asterisk13-t*.rpm --force
rpm -Uvh ~/rpmbuild/RPMS/x86_64/asterisk13-voicemail-13*.rpm --force

I’m specifically having trouble when running “rpmbuild -bp ~/rpmbuild/SPECS/asterisk13.spec” after editing the asterisk13.spec file. When I run that command again, I am left with this:

error: Bad exit status from /var/tmp/rpm-tmp.wqiTov (%prep)

RPM build errors:
    bogus date in %changelog: Thu Nov 11 2019 Matteo Bignotti <mbignotti@sangoma.com> - 13.29.2-1
    Bad exit status from /var/tmp/rpm-tmp.wqiTov (%prep)

Same result when running “rpmbuild -ba ~/rpmbuild/SPECS/asterisk13.spec”

Then, when I try to run the last four “rpm -Uvh” commands, the file it’s looking for is not found. When I ls the ~/rpmbuild/RPMS/x86_64 directory, there are no asterisk13 RPMs, so I’m assuming that has something to do RPM build errors.

Has anyone done this recently? If so, do you have any insight?

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@mercer2 wrote:

Hi

I have a freepbx and a 2N Ip verso on a gate, I currently have the IP verso listening for dtmf tones to open the gate.

However once you talk the 2N Ip verso picks up voice and registers it as a dtmf tone and opens the gate.

I’m using as freepbx extensions the sangoma db20. Is there a better way to integrate a 2N intercom to freepbx without using dtmf

Thank you

P.s. the dtmf code is 1. I do not want to use a long code just one button to open the gate.

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@dux wrote:

Hello, everyone.
In macro-user-callerid extension, file extensions_additional.conf names of callers (callerid(name)) is limited to 40 characters (the line exten => s,n,Set(CALLERID(name)=${CALLERID(name):0:40})
), which does not work with non-ascii characters. If I set the limit to 80 characters, it is overwritten on the next reload. I have tried to override the extension in the extensions_override.conf file, but it does not work. How to correctly override the extension?

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@claloano wrote:

Beep before recording the message

I never use mailboxes but now I have a blasting message box
Everything seems ok but I don’t hear the beeeepp … before you start recording, is there perhaps a way to enable and disable the acoustic warning before you start recording?

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@LPV wrote:

Hi all.
When i’m make call to trunk, the macro [macro-dialout-trunk] from extensions_additional.conf does strange thing.
Called from local extension : 6103
Called to extension in another telephony station through H323 trunk, number : 1307
In log file:
– Executing [s@macro-dialout-trunk:21] ExecIf(“PJSIP/6103-00000041”, “1?Set(CONNECTEDLINE(num,i)=1307)”) in new stack
– Executing [s@macro-dialout-trunk:22] ExecIf(“PJSIP/6103-00000041”, “1?Set(CONNECTEDLINE(name,i)=CID:6103)”) in new stack
– Executing [s@macro-dialout-trunk:23] ExecIf(“PJSIP/6103-00000041”, “0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)6103)”) in new stack

So in history on phone i see entry CID:6103 without dialed number.
In history:
Dialed name : CID:6103
Dialed number : 1307

It may be error in dialplan:
exten => s,n,ExecIf($["${DB(AMPUSER/${AMPUSER}/cidname)}" != “”]?Set(CONNECTEDLINE(num,i)=${DIAL_NUMBER}))
exten => s,n,ExecIf($[$["${DB(AMPUSER/${AMPUSER}/cidname)}" != “”] & $["${CALLERID(name)}"!=“hidden”]]?Set(CONNECTEDLINE(name,i)=CID:${CALLERID(number)}))
exten => s,n,ExecIf($[$["${DB(AMPUSER/${AMPUSER}/cidname)}" != “”] & $["${CALLERID(name)}"=“hidden”]]?Set(CONNECTEDLINE(name,i)=CID:(Hidden)${CALLERID(number)}))

???

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@Croissantisokay wrote:

Hi,

I need to know the following:

1)Does the Originate action support multiple Application keys? If so,how does it handle the order in which they’re added to the Originate action?

2)If it does not support multiple Application keys, I’ll have to instruct the Originate action to enter a context in the dialplan, and pass the sequence of applications in its Variable key. How would I configure the dialplan context to dynamically handle the sequence of applications to execute? Are there alternatives for doing what I require?

Hope someone know about this.
Good monday y’all

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@ttquattroman wrote:

Hi, Is there a voicemail timer value that can be changed. I want to extend the time an extension rings before diverting to voiclemail.

Thanks

Peter

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@AmidouFlorian92 wrote:

When trying to send SMS via my freepbx server using a Linphone sofphone I got several errors when typing sip set debug on on asterisk console and I got on my linphone “[Mon Feb 17 12:43:10 2020] Your message to 53584523 has failed. Retry later.” :
I want to send SMS to my smarthphone which number is 53584523 (calls work perfectly) .This is set debug on results,I have for example SIP/2.0 401 Unauthorized , SIP/2.0 401 Unauthorized, SIP/2.0 404 Not Found and unsupported media type error.Thanks for your help

:

<— SIP read from UDP:192.168.20.10:5060 —>
MESSAGE sip:53584523@192.168.20.3:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.20.10:5060;branch=z9hG4bK.CwnP5b6Wp;rport
From: sip:1002@192.168.20.3;tag=hdpZn9Fx-
To: sip:53584523@192.168.20.3
CSeq: 20 MESSAGE
Call-ID: elwH6dHZcu
Max-Forwards: 70
Supported: replaces, outbound
Content-Type: application/im-iscomposing+xml
Content-Length: 282
Date: Mon, 17 Feb 2020 12:43:09 GMT
User-Agent: Linphone Desktop/4.1.1 (belle-sip/1.6.3)

<?xml version="1.0" encoding="UTF-8"?>

active60
<------------->
— (12 headers 2 lines) —
Sending to 192.168.20.10:5060 (NAT)
Receiving message!

<— Transmitting (NAT) to 192.168.20.10:5060 —>
SIP/2.0 415 Unsupported Media Type
Via: SIP/2.0/UDP 192.168.20.10:5060;branch=z9hG4bK.CwnP5b6Wp;received=192.168.20.10;rport=5060
From: sip:1002@192.168.20.3;tag=hdpZn9Fx-
To: sip:53584523@192.168.20.3;tag=as605f30b4
Call-ID: elwH6dHZcu
CSeq: 20 MESSAGE
Server: FPBX-14.0.13.24(13.29.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘elwH6dHZcu’ in 32000 ms (Method: MESSAGE)

<— SIP read from UDP:192.168.20.10:5060 —>
MESSAGE sip:53584523@192.168.20.3:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.20.10:5060;branch=z9hG4bK.usBQMkqBB;rport
From: sip:1002@192.168.20.3;tag=45kI9FZK0
To: sip:53584523@192.168.20.3
CSeq: 20 MESSAGE
Call-ID: 9qg9M0HYZK
Max-Forwards: 70
Supported: replaces, outbound
Content-Type: text/plain
Content-Length: 4
Date: Mon, 17 Feb 2020 12:43:10 GMT
User-Agent: Linphone Desktop/4.1.1 (belle-sip/1.6.3)

allo
<------------->
— (12 headers 1 lines) —
Sending to 192.168.20.10:5060 (NAT)
Receiving message!
Looking for 53584523 in dpma_message_context (domain 192.168.20.3)
– Executing [53584523@dpma_message_context:1] NoOp(“Message/ast_msg_queue”, “SMS receiving dialplan invoked”) in new stack

<— Transmitting (NAT) to 192.168.20.10:5060 —>
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP 192.168.20.10:5060;branch=z9hG4bK.usBQMkqBB;received=192.168.20.10;rport=5060
From: sip:1002@192.168.20.3;tag=45kI9FZK0
To: sip:53584523@192.168.20.3;tag=as5d61d737
Call-ID: 9qg9M0HYZK
CSeq: 20 MESSAGE
Server: FPBX-14.0.13.24(13.29.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘9qg9M0HYZK’ in 32000 ms (Method: MESSAGE)
– Executing [53584523@dpma_message_context:2] NoOp(“Message/ast_msg_queue”, “To sip:53584523@192.168.20.3:5060”) in new stack
– Executing [53584523@dpma_message_context:3] NoOp(“Message/ast_msg_queue”, “From sip:1002@192.168.20.3”) in new stack
– Executing [53584523@dpma_message_context:4] NoOp(“Message/ast_msg_queue”, “Body allo”) in new stack
– Executing [53584523@dpma_message_context:5] Set(“Message/ast_msg_queue”, “ACTUALTO=sip:53584523”) in new stack
– Executing [53584523@dpma_message_context:6] MessageSend(“Message/ast_msg_queue”, “sip:53584523,sip:1002@192.168.20.3”) in new stack
[2020-02-17 12:43:10] WARNING[8042][C-00000007]: chan_sip.c:6276 create_addr: Purely numeric hostname (53584523), and not a peer–rejecting!
– Executing [53584523@dpma_message_context:7] NoOp(“Message/ast_msg_queue”, “Send status is FAILURE”) in new stack
– Executing [53584523@dpma_message_context:8] GotoIf(“Message/ast_msg_queue”, “1?sendfailedmsg”) in new stack
– Goto (dpma_message_context,53584523,10)
– Executing [53584523@dpma_message_context:10] Set(“Message/ast_msg_queue”, “MESSAGE(body)=”[Mon Feb 17 12:43:10 2020] Your message to 53584523 has failed. Retry later."") in new stack
– Executing [53584523@dpma_message_context:11] Set(“Message/ast_msg_queue”, "ME_1=sip:1002@192.168.20.3>") in new stack
– Executing [53584523@dpma_message_context:12] Set(“Message/ast_msg_queue”, “ACTUALFROM=sip:1002”) in new stack
– Executing [53584523@dpma_message_context:13] MessageSend(“Message/ast_msg_queue”, “sip:1002,ServiceCenter”) in new stack
Reliably Transmitting (NAT) to 192.168.20.10:5060:
MESSAGE sip:1002@192.168.20.10;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.20.3:5060;branch=z9hG4bK1573d1a6;rport
Max-Forwards: 70
From: “ServiceCenter” sip:Unknown@192.168.20.3;tag=as6e047433
To: sip:1002@192.168.20.10;transport=udp
Contact: sip:Unknown@192.168.20.3:5060
Call-ID: 0650f324652bd2be5b1d926e4d1c324a@[::1]:5060
CSeq: 102 MESSAGE
User-Agent: FPBX-14.0.13.24(13.29.2)
Content-Type: text/plain;charset=UTF-8
Content-Length: 78

“[Mon Feb 17 12:43:10 2020] Your message to 53584523 has failed. Retry later.”

Scheduling destruction of SIP dialog ‘0650f324652bd2be5b1d926e4d1c324a@[::1]:5060’ in 6400 ms (Method: MESSAGE)
Really destroying SIP dialog ‘029316ad65683795599254832e64f400@[::1]:5060’ Method: MESSAGE
– Executing [53584523@dpma_message_context:14] Hangup(“Message/ast_msg_queue”, “”) in new stack
== Spawn extension (dpma_message_context, 53584523, 14) exited non-zero on ‘Message/ast_msg_queue’

<— SIP read from UDP:192.168.20.10:5060 —>
MESSAGE sip:Unknown@192.168.20.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.20.10:5060;branch=z9hG4bK.O5Rombk8E;rport
From: sip:1002@192.168.20.3;tag=wdJF3jYze
To: “ServiceCenter” sip:Unknown@192.168.20.3
CSeq: 20 MESSAGE
Call-ID: ZUE1hx8Xvd
Max-Forwards: 70
Supported: replaces, outbound
Content-Type: message/imdn+xml
Content-Length: 274
Date: Mon, 17 Feb 2020 12:43:10 GMT
User-Agent: Linphone Desktop/4.1.1 (belle-sip/1.6.3)

<?xml version="1.0" encoding="UTF-8"?>

0650f324652bd2be5b1d926e4d1c324a@[::1]:50602020-02-17T12:43:10Z
<------------->
— (12 headers 2 lines) —
Sending to 192.168.20.10:5060 (NAT)
Receiving message!

<— Transmitting (NAT) to 192.168.20.10:5060 —>
SIP/2.0 415 Unsupported Media Type
Via: SIP/2.0/UDP 192.168.20.10:5060;branch=z9hG4bK.O5Rombk8E;received=192.168.20.10;rport=5060
From: sip:1002@192.168.20.3;tag=wdJF3jYze
To: “ServiceCenter” sip:Unknown@192.168.20.3;tag=as6d54fbad
Call-ID: ZUE1hx8Xvd
CSeq: 20 MESSAGE
Server: FPBX-14.0.13.24(13.29.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘ZUE1hx8Xvd’ in 32000 ms (Method: MESSAGE)

<— SIP read from UDP:192.168.20.10:5060 —>
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.20.3:5060;branch=z9hG4bK1573d1a6;rport
From: “ServiceCenter” sip:Unknown@192.168.20.3;tag=as6e047433
To: sip:1002@192.168.20.10;transport=udp;tag=at8w4DM
Call-ID: 0650f324652bd2be5b1d926e4d1c324a@[::1]:5060
CSeq: 102 MESSAGE

<------------->
— (6 headers 0 lines) —
Really destroying SIP dialog ‘0650f324652bd2be5b1d926e4d1c324a@[::1]:5060’ Method: MESSAGE

<— SIP read from UDP:192.168.20.10:41818 —>
REGISTER sip:192.168.20.3:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.20.10:41818;branch=z9hG4bK-524287-1—e85f659cf032c4a8;rport
Max-Forwards: 70
Contact: sip:1992@192.168.20.10:41818;rinstance=067e33109b32057f;transport=UDP
To: sip:1992@192.168.20.3:5060;transport=UDP
From: sip:1992@192.168.20.3:5060;transport=UDP;tag=ac66bb41
Call-ID: IzxDds9PtvjHgDOf8UJ5uw…
CSeq: 137 REGISTER
Expires: 60
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
User-Agent: Z 5.3.7 rv2.9.30
Authorization: Digest username=“1992”,realm=“asterisk”,nonce=“0ef31d4e”,uri=“sip:192.168.20.3:5060;transport=UDP”,response=“b2d0b8744ce678e313e7d70777d8d5d0”,algorithm=MD5
Allow-Events: presence, kpml, talk
Content-Length: 0

<------------->
— (14 headers 0 lines) —
Sending to 192.168.20.10:41818 (NAT)
Sending to 192.168.20.10:41818 (NAT)

<— Transmitting (NAT) to 192.168.20.10:41818 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.20.10:41818;branch=z9hG4bK-524287-1—e85f659cf032c4a8;received=192.168.20.10;rport=41818
From: sip:1992@192.168.20.3:5060;transport=UDP;tag=ac66bb41
To: sip:1992@192.168.20.3:5060;transport=UDP;tag=as1075ea75
Call-ID: IzxDds9PtvjHgDOf8UJ5uw…
CSeq: 137 REGISTER
Server: FPBX-14.0.13.24(13.29.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“6929e230”
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘IzxDds9PtvjHgDOf8UJ5uw…’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:192.168.20.10:41818 —>
REGISTER sip:192.168.20.3:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.20.10:41818;branch=z9hG4bK-524287-1—15ee7145ae04880d;rport
Max-Forwards: 70
Contact: sip:1992@192.168.20.10:41818;rinstance=067e33109b32057f;transport=UDP
To: sip:1992@192.168.20.3:5060;transport=UDP
From: sip:1992@192.168.20.3:5060;transport=UDP;tag=ac66bb41
Call-ID: IzxDds9PtvjHgDOf8UJ5uw…
CSeq: 138 REGISTER
Expires: 60
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
User-Agent: Z 5.3.7 rv2.9.30
Authorization: Digest username=“1992”,realm=“asterisk”,nonce=“6929e230”,uri=“sip:192.168.20.3:5060;transport=UDP”,response=“dbe5fdb08cb8b67c6ab66d0007715616”,algorithm=MD5
Allow-Events: presence, kpml, talk
Content-Length: 0

<------------->
— (14 headers 0 lines) —
Sending to 192.168.20.10:41818 (NAT)
Reliably Transmitting (NAT) to 192.168.20.10:41818:
OPTIONS sip:1992@192.168.20.10:41818;rinstance=067e33109b32057f;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.20.3:5060;branch=z9hG4bK32acbc87;rport
Max-Forwards: 70
From: “Unknown” sip:Unknown@192.168.20.3;tag=as0c63a747
To: sip:1992@192.168.20.10:41818;rinstance=067e33109b32057f;transport=UDP
Contact: sip:Unknown@192.168.20.3:5060
Call-ID: 0b9e320818ecec6c574b9f99167d2351@192.168.20.3:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-14.0.13.24(13.29.2)
Date: Mon, 17 Feb 2020 12:43:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

---

<— Transmitting (NAT) to 192.168.20.10:41818 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.20.10:41818;branch=z9hG4bK-524287-1—15ee7145ae04880d;received=192.168.20.10;rport=41818
From: sip:1992@192.168.20.3:5060;transport=UDP;tag=ac66bb41
To: sip:1992@192.168.20.3:5060;transport=UDP;tag=as1075ea75
Call-ID: IzxDds9PtvjHgDOf8UJ5uw…
CSeq: 138 REGISTER
Server: FPBX-14.0.13.24(13.29.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 60
Contact: sip:1992@192.168.20.10:41818;rinstance=067e33109b32057f;transport=UDP;expires=60
Date: Mon, 17 Feb 2020 12:43:13 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘IzxDds9PtvjHgDOf8UJ5uw…’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:192.168.20.10:41818 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.20.3:5060;branch=z9hG4bK32acbc87;rport=5060
Contact: sip:192.168.20.10:41818
To: sip:1992@192.168.20.10:41818;rinstance=067e33109b32057f;transport=UDP;tag=6b05b34d
From: “Unknown” sip:Unknown@192.168.20.3;tag=as0c63a747
Call-ID: 0b9e320818ecec6c574b9f99167d2351@192.168.20.3:5060
CSeq: 102 OPTIONS
Accept: application/sdp, application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 5.3.7 rv2.9.30
Allow-Events: presence, kpml, talk
Content-Length: 0

<------------->
— (14 headers 0 lines) —
Really destroying SIP dialog ‘0b9e320818ecec6c574b9f99167d2351@192.168.20.3:5060’ Method: OPTIONS

<— SIP read from UDP:192.168.20.10:5060 —>

<------------->

<— SIP read from UDP:192.168.20.4:5060 —>
REGISTER sip:192.168.20.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.20.4:5060;branch=z9hG4bK0706db3b;rport
Max-Forwards: 70
From: sip:1900@192.168.20.3;tag=as74c4e565
To: sip:1900@192.168.20.3
Call-ID: 6f0b049f350046771e3a75a926f3bab6@0.0.0.0
CSeq: 231 REGISTER
User-Agent: VoxStack Wireless Gateway
Authorization: Digest username=“1900”, realm=“asterisk”, algorithm=MD5, uri=“sip:192.168.20.3”, nonce=“285b31e1”, response=“ae68941458b5216a70f147d07ee1ce13”
Expires: 120
Contact: sip:1900@192.168.20.4:5060
Content-Length: 0

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@fleex2017 wrote:

I have an inquiry or someone can help me
I have a Freepbx communications server and GSM Gateway 32 Chanels Goip

When the call is made through Goip, the call is counted as answered, and the call actually does not answer

as explain

An agent calling my personal number and my mobile number rings and i’m no answer and busy
login in cdr show same calls answered
how to fix this
any try to call outbound (all answered)

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@vorobyov_ms wrote:

Hi ! Please help
I have two asterisk servers.
On first server ip addresses: 193.178.146.224(external) and 193.178.146.224 (internal)
On second server ip addr: 172.21.21.4(internal)
On both asterisk servers i have sip trunks who point out each other

Calls go from first asterisk to second but i have error
NoOp("SIP/193.178.146.224-00000003", "Received incoming SIP connection from unknown peer to 670101281") in new stack
Firewall turn off on both servers

Please help ) Where is my mistakes ?

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@Balticfinance wrote:

Is there any way to fix the CDR reports in FreePBX to show special characters such as the German Umlaute äüö?

We have a lot of names with those characters and everywhere else they seem to work. They are reported properly to the phones as CDRnames, the console shows them " – Executing [s@macro-user-callerid:44] Set(“SIP/sipconnect-00000071”, “CALLERID(name)=C:Krützfeldt, Christian”) in new stack". All other parts of the FreePBX GUI are fine with them.

Just the reports under https://myserver/admin/config.php?display=cdr don’t like the characters and show garbage. Same is true if I export the reports as CSV File.
It probably has something to do with the character encoding but I don’t know how to fix it.

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@Hawkeye wrote:

HI, just finished upgrading a FreePBX 13 server to FreePBX 14 using the upgrade-distro found on https://wiki.freepbx.org/display/PPS/Upgrading+from+FreePBX+10.13.66+to+SNG7

After upgrade is completed. # fwconsole start doesn’t start asterisk

fwconsole start

Running FreePBX startup…
Taking too long? Customize the chown command, See http://wiki.freepbx.org/display/FOP/FreePBX+Chown+Conf
Setting Permissions…
Setting base permissions…Done
Setting specific permissions…
59637 [============================]
Finished setting permissions
Starting Asterisk…
[--------------->------------] 32 secs
In Start.class.php line 187:

Unable to connect to Asterisk. Did it start?

start [–pre] [–post] [–skipchown] [–] []…

Tried fwconsole chown and then fwconsole start but it does same thing as above.

Can login to the GUI but since it cannot connect to asterisk cannot get to advanced settings to see if there is a difference in admin password vs /etc/asterisk/manager.conf

Can anyone provide any info to get this server working?
Thanks.

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@coderXtreme wrote:

I’m trying to figure out the best setup for a failover destination in the queue, based on the DID. For example, let’s say I have two DIDs 555-555-5555 and 555-555-1234. Let’s say all operators are busy and both of those DIDs receive a call, I want 5555 to forward to 904-222-2222, and the number ending in 1234 to forward to 904-888-2555. Is this possible?

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@keffa wrote:

I have an odd problem that I can’t seem to get my head around.

I have two different trunks.One is a trunk that uses IP authentication, the other one registers with the SIP provider using a username and password. We have just added the trunk that uses IP authentication as our provider has moved away from traditional copper lines.

When a call comes in on either trunk, the inbound route for them directs each of them to the same place and they work exactly the same except it’s two different numbers from two different providers.

The calls route to the phone handsets like this:

Call comes in on trunk > Goes to override condition (Force phones to ring on/off) > Goes to time conditions (Office open/closed) > Goes to Queue (300: Incoming sales calls).

Everything works fine at the phone end of things, irrespective of which trunk the call comes in on, calls come in. Caller ID is passed at all stages of the call correctly (Using PAI), the phones see the callers ID along with the CID prefixes we add. The phones work as expected.

However, in the case of the trunk that uses IP authentication (Or IP trust if thats the correct term), the call logs are being recorded very strangely.

What SHOULD happen is the call type is reflected as “In”, the source is the caller ID of the incoming caller, and the destination is either the DID of the trunk or the SIP providers internal extension.

Instead, what we are seeing is the call type is appearing as “External”, and the source of the call as being the number of the queue (300) instead of the incoming callers ID, and the destination as the agents extension number that actually handled the call. This is definitely not a desirable situation.

Additionally, it is generating call record logs and empty, 0 second call recordings to all the other agents phones in the queue that follow the same logged format. Only the agent who answered the call actually gets the call audio recorded.

Yet oddly, on calls that come in through the trunk that uses username/password registration, the calls are correctly classified as “In” on the call type, and the caller ID is passed correctly as the source and the destination is shown correctly as that sip providers trunk extension (Which is expected as they do not pass the DID but instead the trunks extension number on their system). In other words its working exactly as I expect it to.

Here is the relevant trunk sip settings for the trunk that is using IP authentication and not logging the calls correctly. For privacy reasons I have obscured identifiable details…

[OUTGOING TRUNK NAME]
SIPPROVIDER-MAINNUMBER

[OUTGOING PEER DETAILS]
type=peer
fromuser=12345678901
fromdomain=123.123.123.123
host=321.321.321.321
dtmfmode=rfc2833
canreinvite=no
insecure=invite
qualify=no
disallow=all
allow=ulaw,alaw

[INCOMING USER CONTEXT]
12345678901

[INCOMING USER DETAILS]
type=peer
fromuser=12345678901
fromdomain=123.123.123.123
host=321.321.321.321
dtmfmode=rfc2833
canreinvite=no
insecure=invite
qualify=no
disallow=all
allow=ulaw,alaw
context=from-trunk

[REGISTER STRING]
***empty***

Call recording is forced at the inbound route stage.

I am at a total loss to understand how this might be occurring if the phones are getting the correct incoming caller ID information? Any pointers would be of huge help if anyone can see where it might be going wrong.

Many thanks for your help.

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@SBaalman wrote:

Our FreePBX is a VM and the host needed install updates, so shut down the FreePBX 14 VM and installed updates on the VM host. When powered the FreePBX VM back up it banned all IP including all our phones and our trunk.

In the emails it is sending out it is stating "The IP xxx.xxx.xxx.xxx has just been banned by Fail2Ban after attempts agains SIP on FreePBX

I checked the firewall in FreePBX to make sure that the settings were correct for the network interfaces and also zones for internal network were set to trusted.

Ahhh! No idea what changed to make it ban everything!

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@kb9mfd wrote:

I am trying to setup a development environment. I downloaded the latest distro, installed it default, and followed the instructions on https://wiki.freepbx.org/display/FOP/Setting+up+a+Development+environment+from+the+FreePBX+Distro When I get to
./install --dev-links -n
It errors out with -
Unable to locate the FreePBX BMO Class 'Callerid’A required module might be disabled or uninstalled. Recommended steps (run from the CLI): 1) fwconsole ma install callerid
2) fwconsole ma enable callerid

if I try to do what it says I get -
Unable to install module callerid:
- Cannot find module

What do I do now… I have the complete log of what I have done so far and can post it, its just long so I did not want to post it right away unless I need to but here is the output of that command -

[root@freepbx framework]# ./install --dev-links -n
Assuming you are Database Root
Checking if SELinux is enabled…Its not (good)!
Reading /etc/asterisk/asterisk.conf…Done
Checking if Asterisk is running and we can talk to it as the ‘asterisk’ user…Y es. Determined Asterisk version to be: 16.6.2
Checking if NodeJS is installed and we can get a version from it…Yes. Determin ed NodeJS version to be: 8.11.3
Preliminary checks done. Starting FreePBX Installation
Checking if this is a new install…Partial
Database Root installation checking credentials and permissions…Connected!
Initializing FreePBX Settings
Finished initalizing settings
Linking files (this may take a bit)…
20280/20280 [============================] 100%
Done
bin is: /var/lib/asterisk/bin
sbin is: /usr/sbin
Symlinking /var/lib/asterisk/bin/amportal to /usr/sbin/amportal …Done
Finishing up directory processes…Done!
Running variable replacement…Done
Creating missing #include files…Done
Setting up Asterisk Manager Connection…Done
Running through upgrades…
Checking for upgrades…
No further upgrades necessary
Finished upgrades
Setting FreePBX version to 16.0.1…Done
Writing out /etc/amportal.conf…Done
Writing out /etc/freepbx.conf…Done
Chowning directories…
Taking too long? Customize the chown command, See http://wiki.freepbx.org/display/FOP/FreePBX+Chown+Conf
Setting Permissions…

In Self_Helper.class.php line 212:

Unable to locate the FreePBX BMO Class 'Callerid’A required module might be disabled or uninstalled. Recommended steps (run from the CLI): 1) fwconsole ma install callerid
2) fwconsole ma enable callerid

chown [-f|–file FILE] [-m|–module MODULE]

In Process.php line 239:

The command “/usr/sbin/fwconsole chown” failed.

Exit Code: 255(Unknown error)

Working directory: /usr/src/freepbx/framework

Output:

Error Output:

install [–dbengine DBENGINE] [–dbname DBNAME] [–dbhost DBHOST] [–cdrdbname CDRDBNAME] [–dbuser DBUSER] [–dbpass DBPASS] [–user USER] [–group GROUP] [–dev-links] [–skip-install] [–webroot WEBROOT] [–astetcdir ASTETCDIR] [–astmoddir ASTMODDIR] [–astvarlibdir ASTVARLIBDIR] [–astagidir ASTAGIDIR] [–astspooldir ASTSPOOLDIR] [–astrundir ASTRUNDIR] [–astlogdir ASTLOGDIR] [–ampbin AMPBIN] [–ampsbin AMPSBIN] [–ampcgibin AMPCGIBIN] [–ampplayback AMPPLAYBACK] [-r|–rootdb] [-f|–force]

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