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Archie Harris Digium D40 phone fails at dialling any SIP extension configured in Asterisk
I have a Digium D40 phone and a Zoiper soft phone application running on my workstation.
What should my next troubleshooting step be?
Thanks.
- I can see both are registered to Asterisk.
- From the soft phone, I can dial the D40 extension and the D40 rings. I can pick the D40 phone up and the call is in session.
- From the soft phone, I can dial the soft phone's own extension and it rings itself.
- From the D40, dialling any combination of the above immediately says "Failed" on the phone display.
What should my next troubleshooting step be?
Thanks.
Best Answer chosen by Archie HarrisArchie HarrisThis was apparently fixed by upgrading from Asterisk 1.8 to Certified Asterisk 13. Thanks.
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Best Answer chosen by Patrick SteiningerMalcolm DavenportTip positive.
Best Answer chosen by Brian (Digium) David (Sangoma) Digium phone configuration guides with Asterisk can be found at http://docs.digium.com/phones/digium-phone-user-guide-a.pdf
Greg Gentz moh
Is it possible to have separate files selected for music on hold.
Example:
Phone Ext Group 1 = Play Rock Music
Phone Ext Group 2 = Play Country Music
PHone Ext Group 3 = Play recorded advertisements
Example:
Phone Ext Group 1 = Play Rock Music
Phone Ext Group 2 = Play Country Music
PHone Ext Group 3 = Play recorded advertisements
Best Answer chosen by Brian (Digium) Charlotte (Digium) MOH is organized in groups of songs. Most MOH situations use the default group, which you can customize by adding and deleting songs. MOH in a call queue can use any group. For example, you can create a MOH group with announcements about special offers, and assign that group to your sales call queue.
Paolo Rossi IAX Softphone for Symbian and iPhone
Hi, I'm searching for a IAX compliant softphone for Symbian OS (on Nokia N8) to connect to an Asterisk Pbx. I have too much truoubles with SIP protocol. For the same reason I'm searching a good app for iPhone4.
Best Answer chosen by Brian (Digium) Denis (Sangoma) Paolo,
Unfortunately, Digium doesn't have plans to develop IAX compliant soft-phone and we are not aware of any that have been developed by a third party vendor for mobile devices.
As a workaround you can enable NAT on your Asterisk system or implement a VPN (Virtual Private Network) on your server. It's my understanding that most mobile devices support VPNs and there are several guides on the internet that show you how to implement a VPN on your Linux server.
Unfortunately, Digium doesn't have plans to develop IAX compliant soft-phone and we are not aware of any that have been developed by a third party vendor for mobile devices.
As a workaround you can enable NAT on your Asterisk system or implement a VPN (Virtual Private Network) on your server. It's my understanding that most mobile devices support VPNs and there are several guides on the internet that show you how to implement a VPN on your Linux server.
Best Answer chosen by Denis (Sangoma) Denis (Sangoma) Elvyn,
Unfortunately, we are not familiarized with Cisco IP 6921 Family phones since this isn't a product develop by Digium. I suggest to read the manual or administration guide of this phone, so you could determine which settings you need to change.
For more information about how to create a SIP peer in Asterisk, please read the sip.conf on your Asterisk's Directory or visit the following link
http://svn.digium.com/svn/asterisk/trunk/configs/sip.conf.sample
In this file, you should be able to see some examples of how to create sip peers for various types of phones, including Cisco.
Regards
~Denis Martinez
Unfortunately, we are not familiarized with Cisco IP 6921 Family phones since this isn't a product develop by Digium. I suggest to read the manual or administration guide of this phone, so you could determine which settings you need to change.
For more information about how to create a SIP peer in Asterisk, please read the sip.conf on your Asterisk's Directory or visit the following link
http://svn.digium.com/svn/asterisk/trunk/configs/sip.conf.sample
In this file, you should be able to see some examples of how to create sip peers for various types of phones, including Cisco.
Regards
~Denis Martinez
Aaron Doyle send a voicemail to another vm box
need to know how to forward a voicemail to another mailbox.
Best Answer chosen by Brian (Digium) Nathan (Digium) Hello Aaron,
Login to the /main panel for the extension that has the email you would like to transfer. Then navigate to the Voicemail/Fax drop down and select Mailbox. Then select the folder that the voicemail is in. Once you see the voicemail click the check box next to it. Then select the "Forward checked to extension" then select the extension you would like to forward it to.
Thank you,
Nathan Donovan
Login to the /main panel for the extension that has the email you would like to transfer. Then navigate to the Voicemail/Fax drop down and select Mailbox. Then select the folder that the voicemail is in. Once you see the voicemail click the check box next to it. Then select the "Forward checked to extension" then select the extension you would like to forward it to.
Thank you,
Nathan Donovan