Gang I have stumbled over a strange issue with Asterisk 13.18.3 I have two interfaces, two different IP Addresses. One facing to the internet, and one facing to am internal voice lan. Therefore I defined two different transports and endpoints: [transport-udp-intern..
Author : Benoit Panizzon
Home » Articles posted by Benoit Panizzon (paged 4)
Dear List Its probably been more than a year now I switched from chan_sip to pjsip. pjsip works much cleaner than chan_sip. But! I have come across a Problem I was not able to solve with Asterisk Dialplan Logic. With pjsip an endpoint can have multi..
Dear List We are renewing our voicemail server and by this occasion I am migrating from chan_sip to pjsip. I have come to a problem I have not experienced on other pjsip examples. Switzerland was heavily SS7 based in the past. So usually you have a Netw..
Im struggling to find the correct RFC which exactly defines how a SIP Invite has to look like after a call has been diverted. Especially what the content of the To: header field has to be. Example call flow: Alice calls Bob who diverts to Carol. Al..
Hey List I sometimes use our asterisk server to do some debugging or other PBX and SBC. Now we have a case where a PBX is replying an incomming invite with 180 ringing immediately. It looks like the SBC does not accept this. According to my understand..
Dear List I try to get my clients to connect via TLS. First I did try Snom M9 phones. After looking at the Wireshark TLSv1 Handhake it became obvious, that the M9 only supports old RC4 and similar ciphers, that are not supported by openssl anymore…
List Asterisk 13.14.1 in use with pjsip stack. On the remote side is a SBC which performs some nat detection. I suppose this means the SBC listens from where it is getting RTP data and then replies to that ip. As long as the asterisk is initiating ..
List I have a still two connected DUNDI peers, but they seem to flap from time to time. A couple of years ago I was able to look up quite some, mostly free call numbers via DUNDI all over the world and I als saw incomming lookups. But not anymore..
Dear List I fear I stumbled over a bug in asterisk 13.14.1. My phones are roaming around, sometimes some are connecting from ipv6 enabled networks, another time they are not. If a connection is ipv6 I would prefer to use ipv6 to avoid ipv4-nat proble..
Dear fellow list readers This is the situation: ISDN Devices => Patton ISDN to SIP GW => Asterisk PJSIP The Patton GW resides on a dynamic IP address, so I cannot really use match=ip in the identify section. The Patton does not send a line paramet..