Hola, estoy instalando un ATA HT503 de Grandstream conectado a Asterisk y todo funciona bastante bien (llamadas en entrada y salientes). El unico problema que tengo es con el colgado. Si la llamada entrante va al buzón de voz y la persona cuelga..
When you perform an attended transfer, the extension of the person transferring is displayed to the co-worker.
Can I override the caller ID to display the callers callerID during a blind transfer?
Core show channels verbose – if you do asterisk -rx cscv from bash From: firstname.lastname@example.org [mailto:email@example.com] On Behalf Of eherr Sent: Wednesday, November 16, 2011 12:06 PM To: Asterisk Users Mail..
Connecting Traditional Telephony Services with Asterisk Communications SystemsTE820 Offers Highest Single-Card Port Density Available for use with AsteriskDigium®, Inc., the Asterisk® Company, today announced the availability of the TE820 Octal-S..
group, I have this situation: I have several contexts with a few extensions each one. I need to give every context a limited quantity of minutes they can use. All the extensions in the context will share the same bag of minutes. Meaning ext 101 use 1..
Any has Skype For Asterisk (SFA) license. http://www.digium.com/en/products/software/skypeforasterisk.php PLEASE NOTE: Skype for Asterisk is no longer available for sale. Skype for Asterisk will be supported for two more years, until July 26, 2013..
all, Do you have an idea on the best way on how to implement a system with multiple Asterisk servers with BLF working in such a way that a peer on one server can subscribe to another peer on the other server in a seamless manner? Has anyone set-u..
I tried making a video SIP call using Asterisk …. But it didnt work….only voice call works?
Is there a way Asterisk can be used to send out SIP invites to external Network Gateways? I.E., I have an Asterisk with some softphones registered on it. I simply want to send out SIP invite, as simple as sip:firstname.lastname@example.org;transport=tcp, to an exter..