I've upgraded some of the phones to 1.4, and it looks like the contacts app buttons broke.
Here is how a contact looks:
Same contact on 1.3.3 phone has Dial VM as a button.
Happens also to intercom and transfer 'buttons', they just get added to the list in all caps.
So far I've tried connecting to a different freepbx and creating a new contact in easy mode and connecting with a factory reset phone.
We've had 2 or 3 PBX's and an FXS PCI board that have been unable to detect the reorder tone on our remote rural PSTN lines. We've tried the auto-detect that some of these devices have had, and we've manually set the frequencies, cadence, and dB. Maybe the problem is with the degraded signal we're getting. Now we're looking at getting an FXO gateway. Has anyone been up against this problem and been able to solve it with a Digium FXO gateway? Our reorder tone really does sound like a reorder tone, and it seems like a device should be able to detect it.
Since a couple of weeks ago, when I create a new extension and a new phone, the digium phone module does not create a new contacts-interenal-ext-xml file.
Freepbx 2.11 distro, Digium DPMA version 2.11.0.8
Any help will be greatly appreciated.
Thank you.
Playing around trying to setup some Digium Phones with DPMA and the plan was to use 10 digit extensions in Asterisk but when you create the 10 digit extensions, only one extension shows up under Connectivity > Digium Phones. If you look in the actual conf file at /etc/askterisk/res_digium_phone_devices.conf you will see 1 phone type entry that with the heading that doesn't match the extension number.
I further tested 8 and 9 digit numbers and they seem to work properly and add an entry for every extension.
Is this an oversite in coding, a bug, or intentional design? Anyone?
Thanks in advance.
Hey all,
I am running FreePBX 3.211.63-10 with all current module updates up to date.
I have recently installed the g729 codec to the system and all works really well. I went to use a page group i have setup and noticed that audio only came out of the Yealink Phone not either of the SNOMs I have setup here.
I can intercom by dialing *80 no problem but when i try and use the paging group 400 i get the below errors.[2014-03-13 12:19:44] WARNING[1224][C-000009bc]: translate.c:443 ast_translator_build_path: No translator path from g729 to slin
This error just keeps scrolling through the CLI.
When i run core show codecs, i get the below
Disclaimer: this command is for informational purposes only.
It does not indicate anything about your configuration.
ID TYPE NAME DESCRIPTION
-----------------------------------------------------------------------------------
100001 audio g723 (G.723.1)
100002 audio gsm (GSM)
100003 audio ulaw (G.711 u-law)
100004 audio alaw (G.711 A-law)
100011 audio g726 (G.726 RFC3551)
100006 audio adpcm (ADPCM)
100019 audio slin (16 bit Signed Linear PCM)
100007 audio lpc10 (LPC10)
100008 audio g729 (G.729A)
100009 audio speex (SpeeX)
100016 audio speex16 (SpeeX 16khz)
100010 audio ilbc (iLBC)
100005 audio g726aal2 (G.726 AAL2)
100012 audio g722 (G722)
100021 audio slin16 (16 bit Signed Linear PCM (16kHz))
300001 image jpeg (JPEG image)
300002 image png (PNG image)
200001 video h261 (H.261 Video)
200002 video h263 (H.263 Video)
200003 video h263p (H.263+ Video)
200004 video h264 (H.264 Video)
200005 video mpeg4 (MPEG4 Video)
400001 text red (T.140 Realtime Text with redundancy)
400002 text t140 (Passthrough T.140 Realtime Text)
100013 audio siren7 (ITU G.722.1 (Siren7, licensed from Polycom))
100014 audio siren14 (ITU G.722.1 Annex C, (Siren14, licensed from Polycom))
100017 audio testlaw (G.711 test-law)
100015 audio g719 (ITU G.719)
100028 audio speex32 (SpeeX 32khz)
100020 audio slin12 (16 bit Signed Linear PCM (12kHz))
100022 audio slin24 (16 bit Signed Linear PCM (24kHz))
100023 audio slin32 (16 bit Signed Linear PCM (32kHz))
100024 audio slin44 (16 bit Signed Linear PCM (44kHz))
100025 audio slin48 (16 bit Signed Linear PCM (48kHz))
100026 audio slin96 (16 bit Signed Linear PCM (96kHz))
100027 audio slin192 (16 bit Signed Linear PCM (192kHz))Translation times between formats (in microseconds) for one second of data
Source Format (Rows) Destination Format (Columns)
gsm ulaw alaw g726 adpcm slin lpc10 g729 ilbc g726aal2 g722 slin16 testlaw
gsm - 15000 15000 15000 15000 9000 15000 15000 15000 15000 17250 26250 15000
ulaw 15000 - 9150 15000 15000 9000 15000 15000 15000 15000 17250 26250 15000
alaw 15000 9150 - 15000 15000 9000 15000 15000 15000 15000 17250 26250 15000
g726 15000 15000 15000 - 15000 9000 15000 15000 15000 15000 17250 26250 15000
adpcm 15000 15000 15000 15000 - 9000 15000 15000 15000 15000 17250 26250 15000
slin 6000 6000 6000 6000 6000 - 6000 6000 6000 6000 8250 17250 6000
lpc10 15000 15000 15000 15000 15000 9000 - 15000 15000 15000 17250 26250 15000
g729 15000 15000 15000 15000 15000 9000 15000 - 15000 15000 17250 26250 15000
ilbc 15000 15000 15000 15000 15000 9000 15000 15000 - 15000 17250 26250 15000
g726aal2 15000 15000 15000 15000 15000 9000 15000 15000 15000 - 17250 26250 15000
g722 15600 15600 15600 15600 15600 9600 15600 15600 15600 15600 - 9000 15600
slin16 21600 21600 21600 21600 21600 15600 21600 21600 21600 21600 6000 - 21600
testlaw 15000 15000 15000 15000 15000 9000 15000 15000 15000 15000 17250 26250 -Any advice would be really appreciated. Let me know if you need further info.
Cheers.
Greetings,
I am trying to get to get the new digium module installed(through the module admin); however, I am receiving an error:
This Module Requires The Digium RPM to be installed (php-digium_register-3.0.5-1_centos6.i686.rpm). Please see this page for more information: http://wiki.freepbx.org/display/F2/Digium+AddonsError(s) installing digiumaddoninstaller:
Failed to run installation scripts
I have followed the path and read the instructions and nothing.
When I try and find the package through CLI I get no package available...Nothing to do
Any help would be great.
As of right now I have uninstalled it through the module admin and tried to re install it, and I still get the same error.
Thanks,
Mac
I have a rapid dial/BLF for call flow control. Call flow control requires authentication. I'd like to have the PIN programmed into the rapid dial button but I can't find any documentation for this. Is there a method for this? A comma, P, w?
Thanks!
Hello,
Do Digium phones (1_4_0_0) support CANCEL Sip Header 'Call completed elsewhere'? Am I missing an obvious setting somewhere?
I have a Queue with a couple of phones, "Mark calls answered elsewhere" in the queue settings is checked.
Queue starts ringing everyone, one phone picks up,
PBX sends CANCEL to the other:
---
Reliably Transmitting (NAT) to xx.xx.xx.xx:1028:
CANCEL sip:111@xx.xx.xx.xx:1039;ob SIP/2.0
Via: SIP/2.0/UDP YY.YY.YY.YY:5060;branch=z9hG4bK679e6866;rport
Max-Forwards: 70
From: "EQ: Queue" <sip:5556667777@YY.YY.YY.YY>;tag=as5511de44
To: <sip:111@xx.xx.xx.xx:1039;ob>
Call-ID:
:5060
CSeq: 102 CANCEL
User-Agent: FPBX-2.11.0(11.6.0)
Reason: SIP;cause=200;text="Call completed elsewhere"
Content-Length: 0<--- SIP read from UDP:xx.xx.xx.xx:1028 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP YY.YY.YY.YY:5060;rport=5060;received=YY.YY.YY.YY;branch=z9hG4bK679e6866
Call-ID:
:5060
From: "EQ: Queue" <sip:5556667777@YY.YY.YY.YY>;tag=as5511de44
To: <sip:111@xx.xx.xx.xx:1039;ob>;tag=p4qc-x0Pr4nYBnXng-JH7MIa.W8HpFAI
CSeq: 102 CANCEL
Content-Length: 0
<--- SIP read from UDP:xx.xx.xx.xx:1028 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP YY.YY.YY.YY:5060;rport=5060;received=YY.YY.YY.YY;branch=z9hG4bK679e6866
Call-ID:
:5060
From: "EQ: Queue" <sip:5556667777@YY.YY.YY.YY>;tag=as5511de44
To: <sip:111@xx.xx.xx.xx:1039;ob>;tag=p4qc-x0Pr4nYBnXng-JH7MIa.W8HpFAI
CSeq: 102 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
User-Agent: Digium D70 1_4_0_0_57389
Content-Length: 0But I still see a missed call notification on that phone.
Shouldn't 200/"Call completed elsewhere" not show a missed call?
I started a thread in the General support area regarding Intercom and ringing, but I am talking to myself over there and I found that it is not at all related to intercom anyway. It's related to the phones, I believe.
Changing the ALERT_INFO from ring-answer to answer, the phones no longer auto answer. They ring normally. I have tried Answer and answer and I have viewed the ring types wiki.
Am I missing something, or do I need to make additional edits for this to work properly?
D70 with Firmware 1.4.1.0
Asterisk 1.8-certified
FreePBX 2.11.0.27
Thank you.
I'm using AsteriskNOW. Today I got a digium card. It's easy to attach it and get it worked. But, after making some test calls, I checked CDR list and found a problem: all Caller IDs were set to 'Unknown'. It looks very strange because I also have a SIP trunk (Skype) and it shows Caller ID very well.
I need to know Caller ID. Please help me!
I'm using Digium's AEX2400E analog card with FreePBX 2.11.0.30 and have a question about what connections on the 66 block are actually being used. In FreePBX I have 4 FXOs on ports 33-36 and 8 FXS' on ports 25-32. Does anyone know how this maps out on a 66 block?
Is there a way to get Digium phones to update caller id after an attended transfer?
I tried changing SIP sendrpid to yes and pai, and directmedia to update. That doesn't seem to work.
Blind Transfer works fine.
Hi,
Trying to connect a FreePBX with my old Panasonic TDA600 through Digium TE110P E1 Card.
The E1 Card is installed and recognized. Alarms OK and everything seems fine.
I have configured a DAHDI trunk and also created an Outbound Route
To ilustrate:
FreePBX (1XXX) => E1 => Panasonic (2XXX) => E1 => Cisco CME (4XXX)
Dialing from FreePBX to Cisco CME or vice-versa works fine (using from-pstn context.
Also Outbound calls from FreePBX to Panasonic work fine.
The only thing that I'm having trouble to figure out is Inbound to FreePBX from Panasonic.
What happens is the FreePBX is getting only the first digit dialed from Panasonic:
asterisk*CLI>
-- Accepting call from '2130' to '1' on channel 0/31, span 1
> [INSERT INTO cel (eventtype,eventtime,cid_name,cid_num,cid_ani,cid_rdnis,cid_dnid,exten,context,channame,appname,appdata,amaflags,accountcode,uniqueid,linkedid,peer,userdeftype,userfield) VALUES ('CHAN_START',{ts '2014-04-21 15:02:19'},'','2130','','','','1','from-pstn','DAHDI/i1/2130-5','','',3,'','1398085339.776','1398085339.776','','','')]
-- Executing [1@from-pstn:1] Set("DAHDI/i1/2130-5", "__FROM_DID=1") in new stack
-- Executing [1@from-pstn:2] NoOp("DAHDI/i1/2130-5", "Received an unknown call with DID set to 1") in new stack
-- Executing [1@from-pstn:3] Goto("DAHDI/i1/2130-5", "s,a2") in new stack
-- Goto (from-pstn,s,2)
-- Executing [s@from-pstn:2] Answer("DAHDI/i1/2130-5", "") in new stack
-- Executing [s@from-pstn:3] Wait("DAHDI/i1/2130-5", "2") in new stack
> [INSERT INTO cel (eventtype,eventtime,cid_name,cid_num,cid_ani,cid_rdnis,cid_dnid,exten,context,channame,appname,appdata,amaflags,accountcode,uniqueid,linkedid,peer,userdeftype,userfield) VALUES ('ANSWER',{ts '2014-04-21 15:02:19'},'','2130','2130','','1','s','from-pstn','DAHDI/i1/2130-5','Answer','',3,'','1398085339.776','1398085339.776','','','')]
-- Executing [s@from-pstn:4] Playback("DAHDI/i1/2130-5", "ss-noservice") in new stack
-- Playing 'ss-noservice.ulaw' (language 'en')
-- Span 1: Channel 0/31 got hangup request, cause 16
== Spawn extension (from-pstn, s, 4) exited non-zero on 'DAHDI/i1/2130-5'
-- Executing [h@from-pstn:1] Macro("DAHDI/i1/2130-5", "hangupcall,") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("DAHDI/i1/2130-5", "1?theend") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] ExecIf("DAHDI/i1/2130-5", "0?Set(CDR(recordingfile)=)") in new stack
-- Executing [s@macro-hangupcall:4] Hangup("DAHDI/i1/2130-5", "") in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'DAHDI/i1/2130-5' in macro 'hangupcall'
== Spawn extension (from-pstn, h, 1) exited non-zero on 'DAHDI/i1/2130-5'
-- Hungup 'DAHDI/i1/2130-5'
> [INSERT INTO cel (eventtype,eventtime,cid_name,cid_num,cid_ani,cid_rdnis,cid_dnid,exten,context,channame,appname,appdata,amaflags,accountcode,uniqueid,linkedid,peer,userdeftype,userfield) VALUES ('HANGUP',{ts '2014-04-21 15:02:21'},'','2130','2130','','1','h','from-pstn','DAHDI/i1/2130-5','','',3,'','1398085339.776','1398085339.776','','','')]
> [INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,lastapp,lastdata,duration,billsec,disposition,amaflags,uniqueid) VALUES ({ ts '2014-04-21 15:02:19' },'2130','2130','s','from-pstn','DAHDI/i1/2130-5','Playback','ss-noservice',2,2,'ANSWERED',3,'1398085339.776')]
> [INSERT INTO cel (eventtype,eventtime,cid_name,cid_num,cid_ani,cid_rdnis,cid_dnid,exten,context,channame,appname,appdata,amaflags,accountcode,uniqueid,linkedid,peer,userdeftype,userfield) VALUES ('CHAN_END',{ts '2014-04-21 15:02:21'},'','2130','2130','','1','h','from-pstn','DAHDI/i1/2130-5','','',3,'','1398085339.776','1398085339.776','','','')]
> [INSERT INTO cel (eventtype,eventtime,cid_name,cid_num,cid_ani,cid_rdnis,cid_dnid,exten,context,channame,appname,appdata,amaflags,accountcode,uniqueid,linkedid,peer,userdeftype,userfield) VALUES ('LINKEDID_END',{ts '2014-04-21 15:02:21'},'','2130','2130','','1','h','from-pstn','DAHDI/i1/2130-5','','',3,'','1398085339.776','1398085339.776','','','')]
Any help would be greatly appreciated
Niti
Hi Everyone
I've just installed a fresh copy of FreePBX Stable-5.211.65-11 64-bit into a VMWare ESX 5.1 VM. I've done minimal configuration in that I've created a trunk + inbound + outbound route plus one extension. I've registered my DPMA key via the FreePBX GUI as well and 'digium_phones license status' shows that all is good there.
Now, I have the FreePBX server in our database on the local subnet 10.253.249.0/26. The IP phone in question is on another network with local subnet 10.4.2.0/23. There is an MPLS private IP network between the two locations, so those two networks can route between each other with no NAT what so ever. The phone can ping FreePBX and vice-versa without an issue.
On the phone, I'm manually specifying the FreePBX server IP of 10.253.249.34:5060 and the requests are timing out. Running tcpdump on the server side, I see UDP packets coming in like this:
15:14:15.911594 IP (tos 0x0, ttl 62, id 27830, offset 1560, flags [none], proto UDP (17), length 39)
10.4.3.62 > 10.253.249.34: udp
0x0000: 4500 0027 6cb6 00c3 3e11 fdeb 0a04 033e E..'l...>......>
0x0010: 0afd f922 4420 4345 5254 4946 4943 4154 ..."D.CERTIFICAT
0x0020: 452d 2d2d 2d2d 0a00 0000 0000 0000 E-----........
but there doesn't appear to be a response from Asterisk at all, hence the request fails and times out. I've turned on core verbosity and debug and nothing appears to come through. Sip debugging doesn't yield anything either.
Can anyone suggest any further debugging steps?
Thanks,
Dave
I'm trying to create BLF buttons that will intercom an extension instead of trying to call it. I tried creating external contacts in the contact directory but then BLF does not work. All internal contacts will automatically tell it to just call. I edited the actual contacts configuration file and got it working perfectly using that but on the next restart that file got overwritten by the freepbx gui. How can I make this work?
I've been searching for information on utilizing multicast paging with Digium phones. My test phone is a D40, the test system is FreePBX 5.211.65-11.
Has anyone set up a paging system utilizing Digium and multicasting? If so, any advice is most welcome. For the Yealinks it was fairly straightforward but I haven't found very much on this type of application for Digium.
I have read that Digiums are capable of using multicast DNS discovery, but have not found mention yet of being able to set a multicast listen or send address on the phones.
Thank you for your thoughts.
Has anyone had any success configuring a custom logo on the new Digium D45 phones? I have been able to set this up using the D70. but am running into a wall on the D45's.
I'm setting up some Digium D40 phones with Asterisk/FreePBX and have encountered some issues with the web GUI module. The screenshots of the Digium Phone FreePBX module on the Asterisk Wiki show a tab-based navigation on the top with ten tabs. On my FreePBX install, I have right-side navigation with only five options. Missing options are Phonebooks, Alerts, Ringtones, Phone Applications, and Logos. Running `digium_phones license status` in the Asterisk CLI returns 'OK, Valid product license found', and the D40's have access to the visual voicemail and other DMPA features. Are the screenshots on the Asterisk Wiki outdated or is my install messed up? If the screenshots are outdated, how would I go about accessing the features shown there now? Thanks in advance for any help!
Hi guys,
Using the the 64 bit version of FreePBX, Stable-5.211.65-13 (FreePBX 2.11, Linux 6.5 and Asterisk 11) with Digium Phones (D70s, D50s and D40s)
I'd like to display some specific words (names or labels) for the line key.
Right now it displays the internal extension.
We only use the line keys for internal extensions.
But some phones have several extensions, mainly because they want different CIDs.
So basically, for each required CID, the phone has an extension.
The extension numbers don't mean much, that's why I would rather have a label to identify the CID.
Any way to do that? It'd be awesome.
Hi guys,
Using the the 64 bit version of FreePBX, Stable-5.211.65-13 (FreePBX 2.11, Linux 6.5 and Asterisk 11) with Digium Phones (D70s, D50s and D40s)
Some phones have multiple extensions configured on different Line Keys.
Every time we want to change the STATUS to (for exemple) DND, it only changes the very first Line Key.
If we presse another Line Key, the STATUS option disappears.
It would be nice that the STATUS "application" or feature changes the STATUS for all extensions configured on the different line keys.
Thanks!
