how, is there any documentation related on how to setup TE110p with cisco router or how to setup a TE110p card, what are the parameters needed. I am using asterisk 11.
Thanks in Advance
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Setting up TE110p Card with Cisco Router
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Can't update firmware
FreePBX 2.11.0.10
Digium Phones Config 2.11.0.2
Hi,
I DIGIUM PHONES > FIRMWARE
It is impossible for me to do anything, namely I can't get check for updates
See:
http://cl.ly/image/0i3f0L3s2N0M
Any idea how to fix this?
THANK YOU !
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Digium Phones edit page doesn't save changes
When I go to Connectivity > Digium Phones and click Edit on my phone, and then change any settings and click Save, none of my changes are reflected on the reloaded page. I've tried several different fields, including Phone PIN, Phone MAC Address, Timezone, Name Format, and on an on, and none of them change when I click Save.
Also, no firmwares appear in the drop-down menu for Select Firmware. I have gone to the firmware page and clicked Check for Updates, and have one firmware listed there (firmware_1_3_3_0_package). Is that supposed to appear in the firmware drop-down on the phone's Edit page?
I'm on FreePBX distro with FreePBX 2.11.0.10 and Asterisk 11.4.0. The Module Administration page shows Digium Addons 2.11.0.3 and Digium Phones Config 2.11.0.2.
Any help would be greatly appreciated.
Cheers,
Jim
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DPMA module not working
I have PBX-In-A-Flash with FreePBX 2.10.0.11 and the DPMA module. Initially a 2.10. version of the DPMA module was installed but within a few minutes it upgraded to a 2.11.0.2 version. I can not save anything having to do with the phones in that module. I can change things in the General or Network menus, but the phones are a no go. When trying to add a phone, I fill in all the fields and click submit/save. The screen refreshes and goes right back to where it was. The only way I found to add phones is to put the module in "Easy" mode, go the the FreePBX extensions menu, select an extension and click Save on it. Going back to the DPMA module now has all of the phones Digium or not, in the list. I can delete out the non-Digium phones, but when I select the D70 and make changes and click submit, again, the screen refreshes and all of the original values are still there. No changes where saved.
When I take the DPMA module out of "Easy" mode. I see a ton of additional settings but can not save any changes, add phones or anything.
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Digium Free Fax for Asterisk Add-on (Digium Add-ons 2.11.03)
I was recently notified that we are no longer receiving faxes through our AsteriskNOW 3.0 (Asterisk 11.5.1) PBX. The call comes through, and the fax machine indicates 'receiving', but the call is dropped part way through 100% of the time.
We use a SIP trunk into our Asterisk PBX, and the analog fax machine is connected to the PBX via a Linksys SPA8000 SIP ATA.
Faxes previously worked fine, and I suspect the issue is due to the recent upgrade of the Digium Add-ons module for FreePBX. Has anyone else experienced these issues?
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New TE133 (T1 Card ) not auto-detecting
I just got a new TE133 card to attach my PRI to my FreePBX system but when i click on connectivity and DAHDi nothing is showing up under digital hardware. I can't figure out what I'm doing wrong.
I just reinstalled freepbx off the latest ISO just to be sure I hadn't broken anything. Here is the details from my install.
[root@localhost ~]# uname -r
2.6.32-358.2.1.el6.i686
[root@localhost ~]# asterisk -V
Asterisk 11.2.1
[root@localhost ~]# lspci -n
02:00.0 0280: d161:800a (rev 01)
[root@localhost ~]# lspci
02:00.0 Network controller: Digium, Inc. Device 800a (rev 01)
[root@localhost ~]# rpm -qa | grep dahd
dahdi-firmware-oct6114-256-1.05.01-1_centos6.noarch
dahdi-firmware-vpmoct032-1.12.0-1_centos6.noarch
dahdi-linux-2.6.2-1_centos6.i686
dahdi-firmware-te820-1.76-1_centos6.noarch
dahdi-firmware-hx8-2.06-1_centos6.noarch
dahdi-firmware-oct6114-064-1.05.01-1_centos6.noarch
kmod-dahdi-linux-2.6.2-1_centos6.2.6.32_358.0.1.el6.i686.i686
dahdi-firmware-2.0.4-1_centos6.noarch
dahdi-firmware-tc400m-MR6.12-1_centos6.noarch
dahdi-firmware-oct6114-128-1.05.01-1_centos6.noarch
kmod-dahdi-linux-fwload-vpmadt032-2.6.2-1_centos6.2.6.32_358.0.1.el6.i686.i686
asterisk-dahdi-11.2.1-2_centos6.i686
dahdi-tools-2.6.2-1_centos6.i686
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Updating DPMA module wipes firmware folder
I am using FreePBX (Distro) 3.211.63-5 with Asterisk 11.3.0 and Digium Phones Config FreePBX module version 2.11.0.5 (From the FreePBX Repo and manually downloaded from Digium doesnt matter). Didnt try AsteriskNOW with FreePBX to see if it wipes the firmware directory there too.
Even before the FreePBX module appeared for DPMA, any time I uploaded either a new module, or downloaded an upgrade module (Both local and from the repo in FreePBX), it wipes out any firmware located in /var/www/html/admin/modules/digium_phones/firmware_package
And displays an error to the effect at the top of the FreePBX web page.
The simple solution is to manually download the missing version and untar/zip it in the same folder to get rid of the error and continue to be able to upgrade firmware on Digium phones from that point on.
Also, until you download/replace the missing version(s), the "check for updates" button does nothing.
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Digium phones, DPMA and UDP/TCP settings NAT Keepalives
I am not sure if this is a good place to discuss this, but I tried Digium support and got back answers that dont fix the problem.
If you host FreePBX with rentpbx or other cloud company like the Amazon AMI image Ward Mundy published for IncrediblePBX, theres a good chance that Digium phones will be all considered "remote" and behind a firewall.
For the life of me, I cannot find any docs anywhere that talk about enabling and/or adjusting a keepalive value/timer when UDP is used.
Also, when I asked the Digium support (and submitted a feature request), I was lectured about how (in his opinion) no other phones support this feature, and is therefore unneeded.
His suggestions were:
Adjust your firewall to a higher UDP timeout.
30 seconds for a UDP timeout is unusual. (Not so unusual in Sonicwall, PIX/ASA, Mikrotik RouterOS, etc)
Set Asterisk extension to expire session less than 30 seconds
Set qualifyfreq less than 30 seconds in Asterisk.
As far as I know:
All versions of SPA5xa/SPA9xx/Sapura/Linksys/Cisco Phones and ATA's all have keepalive available and default to 20 seconds.. Not sure why, perhaps that company expects udp timeouts to be 30 seconds by default on firewalls?
Also, Grandstream phones/ATA's have this option.
Also, Polycom phones have this option.
Also, Snom phones and cordlesses have this option....
I think I even recall seeing it in my Aastra 5xxx phones...
Allworx proprietary SIP phones have it....
Shoretel proprietary SIP phones have it....
Cant figure for the life of me why Digium does not?
So, being a crafty devil, I switched them to TCP.
Oooops.
Solves my lack of keepalive problem without any other funky workarounds but causes a new problem that even Digium admits.
In TCP mode, the "Reconfigure" option in DPMA doesnt work since it only sends out its reconfigure message by UDP.
Unless Digium has or provides a NAT keepalive option, or Digium does "reconfigures" by TCP when the extension is configured to use TCP there are about 2 options I can see for remote Digium phones.
1) Use Digium Phones/DPMA/FreePBX in UDP mode and one of the workarounds to fake a keepalive or extend the firewall's udp timeouts. (Which in a large enough network, can quickly run out of UDP translations... Doh)
or
2) Use TCP mode and live with the fact that the reconfigure option will never work with remote Digium phones behind firewalls.
Also, using the TCP option with a Digium remote phone causes the visual voicemail to not show any messages in your inbox when in fact the light is blinking and there is voicemail. For this issue, you can use the *97 FreePBX feature code, or reboot your phone and the visual voicemail works for a period of time before it fails again.
Lastly, I requested (using the feature request email) for Digium phones/DPMA to support provisioning STUN/TURN/ICE and their response was "This wont help your NAT issue".
Which of course I was already aware... I simply was requesting the feature.....
Of course, with the NAT Comedia mode of Asterisk now, there is almost no need for STUN/TURN unless its the only way for the endpoint to automatically know it needs to send keepalives on UDP sip sessions....
Am I missing any other workarounds?
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1.4 phone firmware package whoops
Howdy,
So you may have seen the 1.4 firmware package available for download beginning the Friday before Astricon. We whoops'd it. Fixed package coming later today, but it'll require you to go down to the database and clean up some stuff. Details here - http://forums.asterisk.org/viewtopic.php?f=14&t=88055&p=192667#p192667
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Best practice for logging in and out please?
FreePBX 2.11.0.10
Digium Phones Config 2.11.0.2
Hi,
We use several D40 Digium Phones at our office.
We place them in "open space" workstations where people will work when they're at the office.
Meaning that it's not always the same person using a specific phone.
So right now what we do is:
Everytime somone leaves the space they will put their extension in "DND" or "away"
When someone new arrives, they will go into settings > Advanced > Reset to factory default then choose their extension.
I can't believe this is the right way to do it?
Is there an easier way / best practice for logging on and off a D40 phone please?
Thanks !!!
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Line key problem POST firmware upgrade
Just updated my D40 for the very first time today to firmware package 1_4_0_0
Everything works perfect
EXCEPT
that now I only have 1 working line-key
In FreePBX, in the PHONE config in DIGIUM tab.
For one of my phones I have assigned TWO extensions.
Before, what this would do is that it would put one extension on each one of the line keys of the D40.
Now, only one line key is active, the other one has no lab, and nothing happens when I punch it.
I double checked, reconfigured, restarted, etc... nothing works.
Please help.
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Digium Phones - BLF and Customization
Hi,
After reading plenty of good reviews about these phones i bought a few (six) to use with a new setup.
I am using the lastest Distro on asterisk 11. I have installed and licensed the DPMA and got it working (appart from the mDNS, no idea why). I have also upgraded the asterisk module to 1.7. I have upgraded each phone also to the latest firmware.
I need to be able to:
Call Flow:
Change one of the BLF's on the phone to show RED or GREEN for the call flow toggle (hint *280) i have created a phonebook entry and so far i have made it show GREEN for "ON" (red on all other phone makes) and OFF for "GREEN" (green on all other phone makes).
I understand i may need to use a Smart BLF. However i'm stumpped as to where to configure this. I have been through the below however it fails to tell me where the location of the XML config should go. Unless i've missed it.
https://wiki.asterisk.org/wiki/display/DIGIUM/Smart+BLF#SmartBLF-SmartBLF
If possible i would like to create a BLF config and push it out to all the phones.
Phone Name:
On the top left of the phones screen it shows: "Digium Asterisk" is there a way to change this?
Bare in mind that I have been using YeaLink and Cisco SPA942 and SPA504Gs in the past and have had them configured through the OSS endpoint manager for a number of years. never before have i had so mush hassle with customising phones!
Hope someone can shed some light?
Thanks.
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status "Congestion" with Digium Free Fax for Asterisk Add-on
Hello there,
I'm using FreepBX for my VOIPsystem. I'm very happy with it, it's working like a charm. I've even made a telephone system with queue music en choice menu's. LOL.
BUT.. I have some troubles with fax. Receiving was a piece of cake, i'm getting my faxes nicely in my email with a PDF attached. GREAT! Sending is however still a problem.
i followed the manual for implementing the fax module for FreePBX. I got everything working actually, exept that when i send a fax using the GUI in FreePBX nothing is end.
When i look at the CDR status monitor, you can see that a fax has been send over VOIP (there is somekind of VOIP call with a duration of 20 0r 22 seconds), but the status is "CONGESTION".
maybe someone has a idea what is going wrong?
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licensing g729
Helo everybody i am installing the codec g729 for a client, we got the licence key for 30 channels and when i got the email y followed the steps of the readme got the file at /var/lib/asterisk/licenses that matches the information of the license purchased, however when i install the codec if I do load the module without restart i doesn't show the codec on translations and in general asterisk acts bugy if i do 'g729 show licenses' wich yield no result or error or anything at all, when i do restart asterisk it loads but blocks all calls and refuses to work.
i am runing elastix 2.4 x64 distro,
the codec i load is codec_g729a-1.8.0_3.1.5-barcelona_64 wich is sugested by the benchg729 utility however i tried to load a couple of other versions as generic_64 with the same result.
this is the licence file i get
Key-ID: G729-CXXXXXXXXXXX
Product: G.729 Codec
Channels: 30
ExpDate: 2033-11-20
Host-ID: ae:xx:xx:xx:xx:xx:xx:xx:xx:xx:xx:Xx:xx:XX:XX:XX:XX:XX:XX:XX
Signature: Yxxxxxxxxxxxxxxxxxxxxx
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How to insert an extension in AsteriskNOW dpma_message_context to use with IM for X-Lite 4.5.5
Hi all, I'm a real rookie and I'm just trying to get things work.
All it's working quite nice, calls go as a charm, but when I try to send an instant message to another x-lite (same version) extension, Asterisk logs report:
[2013-12-04 11:29:24] WARNING[6018][C-00000003] pbx.c: Channel 'Message/ast_msg_queue' sent to invalid extension but no invalid handler: context,exten,priority=dpma_message_context,s,1
No error on the x lite client, the message seems sent...
Any tip on how to correct/troubleshoot this case ?
Best regards,
Gef.
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Long Dial Time
I have a free PBX setup with digium FXS and FXO ports.
The problem I am having is that when we dial a number on the phone it takes a really long time to start ringing on outbound calls.
Has anyone had this problem before?
Thank you for all the help!
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Prohibit dialing out on some channels of digium Card
Good Day All,
I have installed a Digium TDM400P running asterisk 11.7.0. I have successfully managed to install the card and I am making and receiving calls just fine. However, I would like to be able to restrict incoming calls on one channel of the dial card and only enable outgoing calls on said channel. Reason for this, we have 2 analogue lines one is for voice and one for fax, I would like to have the fax line double as a voice line when we need it, but I would still to have the fax machine's setup to remain as it is. Basically what I plan on doing is splitting the fax line and have one drop connected directly to the fax and another to the pbx, but I do not want the PBX to answer any incoming calls on the fax line. I have done some research and I see where asterisk has the ability to handle fax, but, I would rather not go this route right now. Any assistance you guys can provide would be much appreciated.
Thanks.
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Unused line button programming?
I am using the latest AsteriskNow with Digium phones and was wondering if I can program an unused line button to something other than a quick dial. In particular, a quick way to set/clear DND.
I've read a bit on BLF, but never seen reference to that particular function and given the other forum topic, is it possible with FreePBX?
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IVR not answering, goes to voicemail
Hi,
I'm going to admit that I'm no pro at this. I'm hoping I've overlooked something simple.
I have the Centos 6.5 64bit AsteriskNOW with FreePBX installed. Everything works well, except I cannot get the IVR to answer, meaning, it just keeps ringing and goes to voicemail.
-The inbound route is set to the IVR destination
-I can dial the IVR by extension and it picks up and works properly (responds properly to dialed numbers).
-It is recorded in the correct format (tested with 2 recordings).
The real problem here is that it just does not answer. Everything else works as it should.
Any help or things I could check are appreciated.
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Multiple custom ringtones on Digium phone?
Is it possible to upload multiple custom ringtones in the Digium DPMA area of FreePBX and have them show up on Digium D40 phones?
It seems that only one ringtone can be chosen to be available for selection on the Digium phones, although the documentation states otherwise: https://wiki.asterisk.org/wiki/display/DIGIUM/DPMA+Configuration#DPMAConfiguration-PhoneConfigurationOptions
"ringtone -- Loads ringtones onto a phone. More than one ringtone may be loaded onto a phone."
How can you load more than one ringtone on the Digium phones through the FreePBX UI or in the res_digium_phone_devices.conf file? I've tried adding multiple "ringtone=" lines to the phone config in the conf file, but they aren't available on the phone after reconfiguring the phone.
Thank you
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