ive got calls coming into an 11.21.0 box. The internal phones are analogue off a TDM400 board, and SIP extensions.Using an analogue internal phone, the remote party always hears an echo on its side. We do not hear an echo. Doesnt matter who is the call..
I have an asterisk server at home. Im looking to replace my internal phones with sip cordless (DECT) phones. Im now looking at the Siemens A510IP base ($90 ) and A510H handset ($40) and the Grandview DP715 base ($80) and DP710 handset ($45).The Siem..
Our Asterisk system (126.96.36.199-1digium1~squeeze) has been very stable and generally doing a good job — except that one day, voicemail recordings started being garbled. It only manifests when the VM comes from our telco gateway service — OnSIP/Junct..
Ive got the following use case where I want to simultaneously dial 2 endpoints that both need different CallerID presentation. How can I do that, from the dialplan preferably ? For instance, let say phone A needs to both dial B, an internal SIP ph..
Hallo, I have a production asterisk server running on Ubuntu however all my configs where done using the CLI. I would like to implement a reporting element into the server so I can know the number of calls made, for what duration, on what dates. W..
Hi I want to change my old answering phone machine and two wireless phones with asterisk box + degium TDM400p (3 fxs+1FXO)+ one LAN phone(Aastra Nortel 9133i) + Wifi/SIP phone I am wondering if I´ll lost actual functionalities that are present in..
I thought one possible solution might be creating an [applicationmap] that essentially handles the assisted transfer manually. Ive done a great deal of reading on this matter and aside from the fact that Im still a bit fogy as to how i would e..
I have a 100MB internal lan. aastras are wired. asterisk box is wired next to the switch. But look: sip show peers …….. 142/14188.8.131.52D A5060 OK (137 ms) 144/14410.10.10.44D A5060 OK (136 ms) 145/14510.10.10.45D A5060 OK (168 ms) 150/15010.10.10…