I know there are people with much experience in asterisk, and Iwant to ask if anyone had experiance with this gw http://www.eurotech-communication.com/products/voip-gateways/VoIP-32-CHANNELS-2E1-PRI-1U/Im having trouble getting connect with asteriskany..
We have a FreePBX-12 / Asterisk-12 setup that supports about 24extensions, most internal Snom870s but six or so external (Jitsi-2.8). we use TLS and SRTP everywhere on our side of the fence.The server host is a dedicated atom(tm) box using the Free..
Im having an issue with CDR. Basically, I expect to have all legs of a call having the same linkedid and differing only by the sequence value. That does happen, but Im getting null dst values after doing an attended transfer.Im not sure if this i..
I found an issue with how PJSIP handles a typo in the Dial application. If the Channel is mistakenly typed with two slashes (i.e Dial(PJSIP//xxxx…), the Dial applications fails (obviously), but it also kills the server.I put some code in my pbx_con..
listi need your help please regarding an issue with snom300 and aastra6731i using asterisk11.13.0asterisksnom 3008.7.3.25astra 6731i 220.127.116.119i have configured the trunks like below100 in snom300200 in snom300300 in aastra6731i400 in x-litethe ca..