@gatozgz wrote:

Hello good morning,

On this moment I have this configuration:

Asterisk <----> Cisco Call Manager <--MGCP--> Gateway PRI Cisco 2801
|
|<---------MGCP----------> Gateway FXS Cisco VG224

On my Asterisk PBX I have 650 IP-phones registered and I want to migrate the VG224 to Asterisk.
And when I migrate the VG224 I want to migrate 2801 and swicth off the CCM

I have had several problems with VG224:

1) When I tried to register 1 X FXS port of my VG224 on my Asterisk the port didnt register with this config:
sip.conf:
[77001]
type = friend
host =dynamic
secret=pepito
dtmfmode=rfc2833
canreinvite=no
context=from-internal
trustrpid=yes
sendrpid=no
nat=no
port=5060
qualify=yes
qualifyfreq=60
transport=udp
callcounter=yes
faxdetect=no

If I do the same with a Grandstream 4224 (for example) I can register the extension without any problems.

In the VG224 I created a dial peer associated to extension 77001 with pass pepito and if I executed CLI> sip show peers ,the exension appears with UNKNOWN status

If I change host=dynamic to host = IP of VG224 then the status change to OK

Any idea about I needed to change the host from dynamic to IP of VG224 to can register the VG224 port on my Asterisk IPPBX?

2) When I tried to do calls from an IP phone to an analog phone which was connected on VG224 (both phones registered on my Asterisk-IPPBX) it appeared an error which said: 400 Bad Request 'Malformed/Missing...'

I did can observe that when I called to 77001 ext., on the INVITE message the field To was created on a bad manner:
To: @DireccionIPCentralita;transport=UDP>
instead of
To:

If I add username = 77001 on sip.conf this problem disaapeared.

¿Any idea about why I need to use the username field? With grandstream 4224 I didnt have that kind of poblems

When I do calls between VOIP Phones I dont need to add the username parameter and the INVITE send the To:<... field good

3) And the last problem is related with faxes:

If I connect an analog phone to the FXS port I can make internal/outbound calls and receive incoming calls without problems.

If I connect a FAX on the same FXS port:
I can call to the FAX from a phone and the FAX offhook.
I can call to the FAX from another FAX, then the FAX offhook and finally gives an error message.
If I try to call from the FAX the call is not cursed, even doesnt appears on the asterisk console ¿It does any sense? I think that at least it should appear a triying of call on the CLI (although the call finally can give an error)

Maybe the VG224 config that Im using is not correct. On vg224 I have created a dial peer with:
- modem passthrough nse codec g711alaw
- vad deactivated
- echo canc. deactivate
- dtmf-relay voip codec all mode nte-ca
I have not limited the fax speed.

Best regards and thanks by your help

Miguel Sanz

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@bksales wrote:

the 13.0.23 backup module completes successfully but reports an error. in this case it is backing up to a windows ftp server with no local back up. there is a directory on the server called atlas that the backup is successfully ftp'ed to. it looks like it is trying to create this directory locally.

Saving Backup 2...done!
Initializing Backup 2
Backup Lock acquired!
Running pre-backup hooks...
Adding items...
Building manifest...
Creating backup...
Storing backup...
Creating directory '/atlas'
Directory '/atlas' did not exist and we could not create it
Saving file to remote ftp
Running post-backup hooks...
Backup completed with errors!

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@redhat wrote:

Hello All,

We have a bunch of Cisco SPA 504G and 509G Phones. For some reason when a call comes in, aside from the ring itself there is this loud beep that comes in as well. It is very annoying and happens both when the call comes in from an outside source or from an internal extension.

Does anyone know how in the world to stop this thing?

Thank You!

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@redhat wrote:

Hello All,

I would like to set up distinctive ring for 3 scenarios, external calls, internal calls from another extension and then the door bell (its a sip door phone).

How can I accomplish this? We are using Cisco SPA 504g and 509g phones.

Thank You

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@redhat wrote:

Can someone please help with specific steps to enable auto answer for intercom calls?

I would like to have it so that when you intercom someone's phone it automatically picks up and enables it so that the receiving part can speak as well.

Thanks

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@rcraig114 wrote:

I've gone through a barrage of different web articles on setting up an external cdr database (using MS SQL Server). I tried using the settings in "Advanced Settings" to set this up, but I'm getting a weird error
[FATAL] SQLSTATE[HY000] [2013] Lost connection to MySQL server at 'reading initial communication packet', system error: 104 SQLSTATE[HY000] [2013] Lost connection to MySQL server at 'reading initial communication packet', system error: 104

Some articles have suggested recompiling Asterisk and setting up ODBC support. I guess I'm a little afraid to do that on a FreePBX distro. Anyone have any ideas?

Robert

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@alreilly wrote:

Any Cisco Experts out there? I have a Cisco 7975G Phone that I am trying to get working with FreePBX and have not had any success, phone has been factory reset by hitting # and waiting the lights then 123456789*0# and phone attempts to look for for IP but can't find any TFTP and I can not SSH into the phone. Any ideas? I have successfully used a SPA2102 with FreePBX and have had no issues with that.

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@redhat wrote:

Hello,

I am having an issue with BLFs and a pjsip extension. I have extension 101 which is a pjsip extension and has 2 phones registered to it. I just set up BLFs using the commercial endpoint manager, but all the BLF fields show up in orange.

Can someone please help me figure out how to get this working?

Thanks

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@kbocek wrote:

After spending some time with wifi SIP phones on my cell phone I've decided that call quality and stability is just not there. So I'm looking at the wired-base DECT phones.

Price is good, $60 for the base plus phone and add-on phones are $30.

But the reviews are all over the map. Worrisome.

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@daniel379ba wrote:

I am interested in creating an application that utlizes our FreePBX/VoIP system.

The company I intern for is starting to use a CRM software called ZenDesk. We would like it to generate a ticket and auto populate some information every time a support tech gets a phone call. I've become familiar with their API and feel comfortable with it, but was wondering if FreePBX has some sort of API equivalent utility.

In other words,

How can I get the computer to tell when a VoIP call is coming in?
How do I sniff it for its phone number?

Many thanks,
Daniel

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@NightShift wrote:

Hi all.

I have installed FreePBX 10.13.66-11 from the latest distro with asterisk 13. I need to access the asterisk "menuselect" function so I can enable MySQL
I have installed the Chan_SCCP-b module for use on cisco 7960G phones and so far, so good. But I need to enable or use a realtime database and from what I can find I need to access the menuselect to activate / select res_mysql_conf?

Sorry if this doesn't make sense but I am completely new to linux and freePBX. Or if there is another way to do it and someone could walk me through it step by step I'd really appreciate it, we've all got to start learning somewhere.

Thanks

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@cbsystems wrote:

Im getting this error showing up in the CLI when we make an outbound call:

features_config.c:1349 ast_get_chan_applicationmap: Unknown DYNAMIC_FEATURES item 'apprecord' on channel DAHDI/i1/123-84.

extensions_additional.conf has this set up in the globals section:
DYNAMIC_FEATURES = apprecord

features_applicationmap_additional.conf has this:
apprecord=>*1,caller,Macro,one-touch-record

extensions_additional.conf has this:
[macro-one-touch-record]
include => macro-one-touch-record-custom
exten => s,1,Set(ONETOUCH_REC_SCRIPT_STATUS=)
exten => s,n,System(/var/lib/asterisk/bin/one_touch_record.php "${CHANNEL(name)}")
exten => s,n,Noop(ONETOUCH_REC_SCRIPT_STATUS: [${ONETOUCH_REC_SCRIPT_STATUS}])
exten => s,n,ExecIf($["${REC_STATUS}"="RECORDING"]?Playback(beep))
exten => s,n,ExecIf($["${REC_STATUS}"="STOPPED"]?Playback(beep&beep))
exten => s,n,ExecIf($["${ONETOUCH_REC_SCRIPT_STATUS:0:6}"="DENIED"]?Playback(access-denied))
exten => s,n,MacroExit()

What am I missing? Why is the feature unknown??
Im running FPBX 13.0.123 on Centos 6 with asterisk 13.9.1

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@Sablapet wrote:

Hi there,

I've installed the FreePBX distro running Asterisk version 13.7.1. I've configured several extensions and a sip trunk. All extensions can call each other further more its possible to make outgoing calls.
Unfortunatelly I'm not able to receive incoming calls. The extensions are configured as chan_sip on port 5060 as well as the trunk.
I've read several threads regarding call issues but unfortunatelly I haven't found a solution for my issues. Could someone explain how to debug such a issue correctly?

What I've already done. Double checked the extension and trunk configuration. Trunk configuration is from a blog related to my provider shich should work. At Generals SIP settings I've set NAT to yes and IP Configuration to static IP since I obtain a static one from my provider.

To determine the issue I gone to the Asterisk CLI and set debug on. Unfortunatelly I'm not able to finde the issues within the output.
Is it correct that the call from my cell phone is registrated as "Unknown" and the FreePBX server tries to forward it to the extension and for some reaosn the extension rejects the call?

`
Asterisk 13.7.1, Copyright (C) 1999 - 2014, Digium, Inc. and others.
Created by Mark Spencer markster@digium.com
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 13.7.1 currently running on VoiP-Server (pid = 1634)
Reliably Transmitting (no NAT) to 172.16.0.10:5060:
OPTIONS sip:108@172.16.0.10;line=6852 SIP/2.0
Via: SIP/2.0/UDP 172.16.0.2:5060;branch=z9hG4bK36a9dd40
Max-Forwards: 70
From: "Unknown" ;tag=as50a73c6b
To:
Contact:
Call-ID: 5c063d805660a811214da2fb26359a96@172.16.0.2:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-13.0.124(13.7.1)
Date: Tue, 07 Jun 2016 08:49:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

---

<--- SIP read from UDP:172.16.0.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.0.2:5060;branch=z9hG4bK36a9dd40
Max-Forwards: 70
From: "Unknown" ;tag=as50a73c6b
To:
Call-ID: 5c063d805660a811214da2fb26359a96@172.16.0.2:5060
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Encoding: identity
Accept-Language:
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Date: Tue, 07 Jun 2016 08:49:29 GMT
Supported: replaces, timer
User-Agent: Phone/03.24.0012 (MAC=000000000000; SER= 00000; HW=255)
Content-Length: 0

<------------->
--- (15 headers 0 lines) ---
Really destroying SIP dialog '5c063d805660a811214da2fb26359a96@172.16.0.2:5060' Method: OPTIONS
Reliably Transmitting (no NAT) to 172.16.0.10:5060:
OPTIONS sip:109@172.16.0.10;line=9032 SIP/2.0
Via: SIP/2.0/UDP 172.16.0.2:5060;branch=z9hG4bK60fe3c02
Max-Forwards: 70
From: "Unknown" ;tag=as223ebbfe
To:
Contact:
Call-ID: 4d7bf43c1fa80380064ee9cc7996ea04@172.16.0.2:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-13.0.124(13.7.1)
Date: Tue, 07 Jun 2016 08:49:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

---

<--- SIP read from UDP:172.16.0.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.0.2:5060;branch=z9hG4bK60fe3c02
Max-Forwards: 70
From: "Unknown" ;tag=as223ebbfe
To:
Call-ID: 4d7bf43c1fa80380064ee9cc7996ea04@172.16.0.2:5060
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Encoding: identity
Accept-Language:
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Date: Tue, 07 Jun 2016 08:49:29 GMT
Supported: replaces, timer
User-Agent: Phone/03.24.0012 (MAC=000000000000; SER= 00000; HW=255)
Content-Length: 0

<------------->
--- (15 headers 0 lines) ---
Really destroying SIP dialog '4d7bf43c1fa80380064ee9cc7996ea04@172.16.0.2:5060' Method: OPTIONS
Reliably Transmitting (NAT) to 217.0.23.100:5060:
OPTIONS sip:tel.myprovider.com SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK27143365;rport
Max-Forwards: 70
From: "Unknown" sip:0123987456@xxx.xxx.xxx.xxx;tag=as25bbe6f5
To:
Contact:
Call-ID: 4733926e25e863b74119846378888628@xxx.xxx.xxx.xxx:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-13.0.124(13.7.1)
Date: Tue, 07 Jun 2016 08:49:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

---

<--- SIP read from UDP:217.0.23.100:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;received=xxx.xxx.xxx.xxx;rport=1365;branch=z9hG4bK27143365
To: ;tag=h7g4Esbg_cy8v8vz72eib14dk8o4v9bmcpg4iu9fz
From: "Unknown" sip:0123987456@xxx.xxx.xxx.xxx;tag=as25bbe6f5
Call-ID: 4733926e25e863b74119846378888628@xxx.xxx.xxx.xxx:5060
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '4733926e25e863b74119846378888628@xxx.xxx.xxx.xxx:5060' Method: OPTIONS

<--- SIP read from UDP:172.16.0.10:5060 --->
REGISTER sip:172.16.0.2 SIP/2.0
Via: SIP/2.0/UDP 172.16.0.10:5060;rport;branch=z9hG4bKj8aori78d2z7cnjkudcbrnyen9uhm
Max-Forwards: 70
From: ;tag=k5v0yr6qg84cjo
To:
Call-ID: c8thhtm2d1f8yl0r.mfozx2rs
CSeq: 22768 REGISTER
Contact:
Allow: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, INFO, PRACK, UPDATE
Allow-Events: talk,hold
Expires: 3600
User-Agent: Phone/03.24.0012 (MAC=000000000000; SER= 00000; HW=255)
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Sending to 172.16.0.10:5060 (NAT)
Sending to 172.16.0.10:5060 (NAT)

<--- Transmitting (no NAT) to 172.16.0.10:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.16.0.10:5060;branch=z9hG4bKj8aori78d2z7cnjkudcbrnyen9uhm;received=172.16.0.10;rport=5060
From: ;tag=k5v0yr6qg84cjo
To: ;tag=as60738306
Call-ID: c8thhtm2d1f8yl0r.mfozx2rs
CSeq: 22768 REGISTER
Server: FPBX-13.0.124(13.7.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="39025d73"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog 'c8thhtm2d1f8yl0r.mfozx2rs' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:172.16.0.10:5060 --->
REGISTER sip:172.16.0.2 SIP/2.0
Via: SIP/2.0/UDP 172.16.0.10:5060;rport;branch=z9hG4bKqfzlnv1mow5v
Max-Forwards: 70
From: ;tag=k5v0yr6qg84cjo
To:
Call-ID: c8thhtm2d1f8yl0r.mfozx2rs
CSeq: 22769 REGISTER
Contact:
Allow: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, INFO, PRACK, UPDATE
Allow-Events: talk,hold
Authorization: Digest username="109", realm="asterisk", nonce="39025d73", uri="sip:172.16.0.2", response="a9221b1e287ed56be193c8ed9f246d30", algorithm=MD5
Expires: 3600
User-Agent: Phone/03.24.0012 (MAC=000000000000; SER= 00000; HW=255)
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Sending to 172.16.0.10:5060 (no NAT)
Reliably Transmitting (no NAT) to 172.16.0.10:5060:
OPTIONS sip:109@172.16.0.10;line=9032 SIP/2.0
Via: SIP/2.0/UDP 172.16.0.2:5060;branch=z9hG4bK31edbfb5
Max-Forwards: 70
From: "Unknown" ;tag=as4e426b1f
To:
Contact:
Call-ID: 15d95d0f352f41d055a02fe7495b9b18@172.16.0.2:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-13.0.124(13.7.1)
Date: Tue, 07 Jun 2016 08:49:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

---

<--- Transmitting (no NAT) to 172.16.0.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.0.10:5060;branch=z9hG4bKqfzlnv1mow5v;received=172.16.0.10;rport=5060
From: ;tag=k5v0yr6qg84cjo
To: ;tag=as60738306
Call-ID: c8thhtm2d1f8yl0r.mfozx2rs
CSeq: 22769 REGISTER
Server: FPBX-13.0.124(13.7.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 3600
Contact: ;expires=3600
Date: Tue, 07 Jun 2016 08:49:43 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog 'c8thhtm2d1f8yl0r.mfozx2rs' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:172.16.0.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.0.2:5060;branch=z9hG4bK31edbfb5
Max-Forwards: 70
From: "Unknown" ;tag=as4e426b1f
To:
Call-ID: 15d95d0f352f41d055a02fe7495b9b18@172.16.0.2:5060
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Encoding: identity
Accept-Language:
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Date: Tue, 07 Jun 2016 08:49:43 GMT
Supported: replaces, timer
User-Agent: Phone/03.24.0012 (MAC=000000000000; SER= 00000; HW=255)
Content-Length: 0

<------------->
--- (15 headers 0 lines) ---
Really destroying SIP dialog '15d95d0f352f41d055a02fe7495b9b18@172.16.0.2:5060' Method: OPTIONS

<--- SIP read from UDP:172.16.0.10:5060 --->
REGISTER sip:172.16.0.2 SIP/2.0
Via: SIP/2.0/UDP 172.16.0.10:5060;rport;branch=z9hG4bKespxeft5vb7duiv86ifpftrst.yzs3
Max-Forwards: 70
From: ;tag=a0r2ag
To:
Call-ID: h67dc3yt158hmmxrir8e3.gg.is9y0
CSeq: 9063 REGISTER
Contact:
Allow: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, INFO, PRACK, UPDATE
Allow-Events: talk,hold
Expires: 3600
User-Agent: Phone/03.24.0012 (MAC=000000000000; SER= 00000; HW=255)
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Sending to 172.16.0.10:5060 (NAT)
Sending to 172.16.0.10:5060 (NAT)

<--- Transmitting (no NAT) to 172.16.0.10:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.16.0.10:5060;branch=z9hG4bKespxeft5vb7duiv86ifpftrst.yzs3;received=172.16.0.10;rport=5060
From: ;tag=a0r2ag
To: ;tag=as65858164
Call-ID: h67dc3yt158hmmxrir8e3.gg.is9y0
CSeq: 9063 REGISTER
Server: FPBX-13.0.124(13.7.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7e728470"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog 'h67dc3yt158hmmxrir8e3.gg.is9y0' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:172.16.0.10:5060 --->
REGISTER sip:172.16.0.2 SIP/2.0
Via: SIP/2.0/UDP 172.16.0.10:5060;rport;branch=z9hG4bKr2z5.tadivu.gn
Max-Forwards: 70
From: ;tag=a0r2ag
To:
Call-ID: h67dc3yt158hmmxrir8e3.gg.is9y0
CSeq: 9064 REGISTER
Contact:
Allow: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, INFO, PRACK, UPDATE
Allow-Events: talk,hold
Authorization: Digest username="108", realm="asterisk", nonce="7e728470", uri="sip:172.16.0.2", response="74b05b30001795f408ee6ba1899a59f9", algorithm=MD5
Expires: 3600
User-Agent: Phone/03.24.0012 (MAC=000000000000; SER= 00000; HW=255)
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Sending to 172.16.0.10:5060 (no NAT)
Reliably Transmitting (no NAT) to 172.16.0.10:5060:
OPTIONS sip:108@172.16.0.10;line=6852 SIP/2.0
Via: SIP/2.0/UDP 172.16.0.2:5060;branch=z9hG4bK19091641
Max-Forwards: 70
From: "Unknown" ;tag=as4891b887
To:
Contact:
Call-ID: 7f2d6bf868e5922d65cbd9c54d2cbb1c@172.16.0.2:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-13.0.124(13.7.1)
Date: Tue, 07 Jun 2016 08:49:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

---

<--- Transmitting (no NAT) to 172.16.0.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.0.10:5060;branch=z9hG4bKr2z5.tadivu.gn;received=172.16.0.10;rport=5060
From: ;tag=a0r2ag
To: ;tag=as65858164
Call-ID: h67dc3yt158hmmxrir8e3.gg.is9y0
CSeq: 9064 REGISTER
Server: FPBX-13.0.124(13.7.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 3600
Contact: ;expires=3600
Date: Tue, 07 Jun 2016 08:49:45 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog 'h67dc3yt158hmmxrir8e3.gg.is9y0' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:172.16.0.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.0.2:5060;branch=z9hG4bK19091641
Max-Forwards: 70
From: "Unknown" ;tag=as4891b887
To:
Call-ID: 7f2d6bf868e5922d65cbd9c54d2cbb1c@172.16.0.2:5060
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Encoding: identity
Accept-Language:
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Date: Tue, 07 Jun 2016 08:49:45 GMT
Supported: replaces, timer
User-Agent: Phone/03.24.0012 (MAC=000000000000; SER= 00000; HW=255)
Content-Length: 0

<------------->
--- (15 headers 0 lines) ---
Really destroying SIP dialog '7f2d6bf868e5922d65cbd9c54d2cbb1c@172.16.0.2:5060' Method: OPTIONS`

Can some one point in the right direct or provide a way to debug this correctly?

Best regards,
Sablapet

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@mdioguardi wrote:

All,

One of our clients has a FreePBX appliance running v 6.12.65-32, that's been installed for about a year and change. They have about 12 Polycom VVX 410 phones that all work perfectly and one Polycom IP 6000 that from day 1 has had low volume on both the sending and receiving sides. All phones were configured thru EPM.

Last week I decided to try to fix the low volume issue, the client hadn't mentioned it being an issue in 6 months or so, so I has hoping that it was resolved with one of the FreePBX updates. So I factory reset the phone, deleted the extension in EPM and Extensions, used EPM to find and reprogram the IP6000, but after all of that it had the same exact issue.

I looked high and low and dont see any specific settings related to audio volume, so I decided to take the phone with me and bench it at my office on another phone system.

Needless to say I formatted and then provisioned the IP6000 on a hosted system and it works perfectly, great volume from the speaker phone, great pickup on the microphone, etc.

I have to bring the phone back to the customer and get it working properly but dont know where to look at this point. Has anyone had this type of issue before?

The phone is now running ROM 4.0.8.1608, it was on 4.0.6.0711 on the FreePBX.

Any direction would be greatly appreciated..Thanks Mike

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@jsilva wrote:

hello good afternoon everyone,

I need help with a failure that only occurs with GXW4008 equipment. When I call in the second line (call waiting) makes the first call silent for a few seconds. I did the tests with other CISCO LINKSYS SPA8000 equipment, with this I did not have this problem. The second line usually played without leaving my first line changes.

I use the FreePBX version 12.0.33 with the asterisk 11.

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@jdmage_mx5 wrote:

Let me start out by saying that I am a beginner with FreePBX. I am coming from an older Freeswitch system that everyone in my office hates due to many bugs and issues. We are currently working on moving everything over to FreePBX box.

I am familiar with using Bulk Handler but I am wanting to set the voicemail option of "Require From Same Extension" and "Disable (*) in Voicemail Menu" via Bulk Handler. Does anyone have any insight on the header name I need to label a cell in the CSV to accomplish these task?

Also, is there an easy way to assign AD users to extensions via Bulk handler method?

Thank you everyone for any and all help you may be able to provide.

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@mspuhler wrote:

Does anyone else have issues with Feature Code Admin and the lack of a submit button to save any changes?

Freepbx 13, Feature Code Admin 13.0.5. I have the same issue on multiple servers and have tried both Chrome and IE to rule out any browser type of issues.

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@jiaqi1993 wrote:

Managed to solve the previous registrar problem,

I'm trying to setup spa3000

I can now call from spa to pstn by dialing a prefix, but no sound is coming out.

"[2016-06-08 10:12:37] NOTICE[5496] chan_sip.c: Disconnecting call 'SIP/pstn-0000000f' for lack of RTP activity in 31 seconds" I believed this is the error message.

May I know if anyone has the experience in setting up this device?

FreePBX 13.0.119
SPA3000 SW 3.1.20(GW)

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@CSerpent wrote:

Hi

We have an issue on a distro deployment 10.13.66-12 with phonebook module 13.0.5.2. Originally phonebook was working however now on accessing you get:

Undefined index: name
File:/var/www/html/admin/modules/phonebook/Phonebook.class.php:129

firing up at the top right.

I've seen the same issue but reported as fixed on bugtracker ID FREEPBX-10514.

Any ideas anyone or does this need a new bugtracker ticket?

Thanks

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@newbefreepbxuser wrote:

Hello,

I will have to configure a new sip trunk to my freepbx. My sip provider will install a Cisco Cube in my LAN.

There is no problem tu configure a new sip trunk, but I need to confirm some parameters in my freepbx.

Where can I confirm these parameters in my pbx?:·
Min-SE header = 7200·
DTMF in-band

Is it possible to confirm this values from my
graphic interface?, or do I have to use commands?

Thanks in advance

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