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 18.104.22.1689i have configured the trunks like below100 in snom300200 in snom300300 in aastra6731i400 in x-litethe ca..
Howdy,Im looking at enabling autopause on one of my queues where my queue members are bad about leaving their desks without pausing.The problem I see is that when the queue pauses an Member it doesnt jump into the dialplan to do so which means my ha..
I have had numerous issues with PJSIP outbound calling in Asterisk 13.1.0and SIP.US SIP trunk. My Asterisk server is on EC2 and I have opened up the appropriate ports. The SIP clients can be anywhere on the Internet, including behind NATs.I am able..
Have you tried using tcpdump? Then analyze the pcap on wireshark?Mar..
everyone.We regularly get customers complaining about call quality issues. Most of the time it turns out to be their own broadband. Very occasionally server load. Does anyone have any advice or links to advice on measuring call quality?I’ve been play..
list,i have asterisk 11.15.0 and i have some trunks sip from my providerwe have some ip phone astra 6731ieach Ip-phone is configured with trunk and we callno ihave configured another trunk from the same provider in my asteriski can call all numbers j..