similar to: Pass through registration / proxy

OK - I'll have to rethink how to solve this problem. Maybe I made some assumptions...here's what I'm trying to accomplish: I've been given a legacy PBX with SIP capabilities. I need to have SIP phones connect to Asterisk (for other features, part of the next step) which passes the calls through to the legacy PBX. And conversely, calls to that extension number on the legacy PBX

Hi all, we face a strange behavior while connecting an Asterisk16 instance with PJSIP to 2 providers: we receive error 401 Unauthorized, both of them having Kamailio as front-end. With other providers -we don't know if they run kamailio- registration is just fine. One of the provider took a pcap and told us that expiration was set to 0 that's why they don't accept the

Hello, I am beginning to front my Asterisk cluster with OpenSIPS/Kamailio and so far my biggest issue is the complete lack of quick-start-like documentation for either. Is there any place I can get a very simple HA configuration (telling me where the config files are, for starters, is a good thing) for OpenSIPS or Kamailio with the following features: (a) Support an arbitrarily large number of

Le 15/01/2020 à 19:50, C.Maj a écrit : > On 2020-01-15 11:24, Administrator wrote: > > 8<'s > >> One of the provider took a pcap and told us that expiration was set to 0 >> that's why they don't accept the registration. We took a pcap on our >> side when SIP packet goes out of our server and we see that the >> expiration parameter is setted to

Le 17/01/2020 à 11:54, Administrator a écrit : > > Le 15/01/2020 à 19:24, Administrator a écrit : >> Hi all, >> >> we face a strange behavior while connecting an Asterisk16 instance >> with PJSIP to 2 providers: we receive error 401 Unauthorized, both of >> them having Kamailio as front-end. With other providers -we don't >> know if they run

Hi Guys Is it possible to register Kamallio directly to our SIP provider then load balance the RTP to 2 asterisk servers? We cant do the registration from the asterisk boxes as we want to do it directly from Kamallio. Is this possible? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear List: Please correct me if I am wrong, but as I understand the requirement to connect an IP-PBX to the PSTN via a SIP trunking service provider (ITSP), a SIP proxy service and a SIP registration service are required local to the IP-PBX. Does Asterisk include this functionality and/or are there other open source projects providing these SIP services? Thanks a bunch! John --------------

Hi, I want to have Kamailio in front of one or more Asterisk boxes. I don't think it is necessary for Kamailio and Asterisk to register with one another. I'd like for PJSIP to recognise Kamailio by its IP address. I have two boxes, both have public IP addresses, they also have private IP addresses and can communicate with each other. I have a Snom phone accessing Kamailio via its

Hi all, Lately, I've seen an increase in the number of attacks against my system from the so-called "Friendly Scanner." When one of these script kiddies targets my server, all I see for symptoms is a few of my trunks become lagged due to server load and a stream of messages on the console that resemble this: [Aug 2 20:27:50] == Using SIP VIDEO CoS mark 6 [Aug 2 20:27:50] ==

Well, correct me if I'm wrong, but I would say this conversation you have posted is a bit outdated, now fail2ban can be used with asterisk security log https://wiki.asterisk.org/wiki/display/AST/Asterisk+Security+Event+Logger. On Thu, Aug 17, 2017, 4:53 AM Telium Technical Support <support at telium.ca> wrote: > Keep in mind that the attacks you are seeing in the log are ONLY the

OK, it stopped working. It turns out the transport and endpoints in PJSIP are ok. I can send an invite from my unregistered snom phone and I can see some activity in the CLI. However, when I dial from my snom to Kamailio and have it pass the message to asterisk, PJSIP seems to ignore the sip messages even though they are there. Is there something wrong in the invite that I'm missing? U

Hi all Have recently watched Matt Jordan's session on Kamailio World 2014 On slides 26-29 of his presentation (http://www.kamailio.org/events/2014-KamailioWorld/day1/09-Matt.Jordan-Asterisk12-And-PJSIP.pdf) he speaks about a (completely new, for me at least) approach to build scalable telephony systems, using N instances of Kamailio and N instances of Asterisk Are there any

Hi all, Strange request, I have a customer where we are putting an Asterisk PBX in front of a legacy (non-VoIP) PBX. One of the requirements it that the Asterisk PBX have 2 PRI ports (on towards the legacy PBX and one towards the carrier) with the ability to go to pass through should the Asterisk PBX (software or hardware level) fail. I did not see this feature in the Digium, Sangoma, Allo, or

Hello, We have a provider which is using Kamailio as front end. Our asterisk 13/chan_sip server has no problem to register and pass/receive calls form this provider. Now we want to move to asterisk 16/pjsip and face problem. Registration is OK but when we pass a call our INVITE never receive answer from the provider. We opened a ticket to their support but in the mean time we want to know

Hi all, I saw Matt Jordan's recent Kamailio world talk and was interested in the idea he proposed of stripping out authentication and registration from asterisk and letting Kamailio handle it. All of the tutorials I've seen (e.g. on asipto) show Kamailio forwarding registrations to asterisk. In order to do what Matt suggested would I be correct in assuming I would have to use the

Skipped content of type multipart/alternative-------------- next part -------------- _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

If this is a small site, I recommend you download the free version of SecAst (www.telium.ca <http://www.telium.ca> ) and replace fail2ban. SecAst does NOT use the log file, or regexes, to match etc.instead it talks to Asterisk through the AMI to extract security information. Messing with regexes is a losing battle, and the lag in reading logs can allow an attacker 100+ registration

Hi all, I need to register asterisk on an OpenSIPS SIP Proxy...The Registration is OK but my asterisk is sending OPTIONS packets to OpenSIPS and the SIP Proxy is not replying back...The issue is the UNKNOWN username that reside in the OPTIONS packet as you can see in the captured packets as you can see below: 1. U Asterisk_IP:5060 -> OPENSIPS_IP:5060 2. OPTIONS sip:OPENSIPS_IP

Hi, We have a Kamailio and Asterisk cluster, both machines being on a real 103.x IP address and also on a 172.x OpenVPN address. The problem is that when Kamailo receives a call from the VPN and forwards it to the Asterisk server on it's 103.x address, Asterisk never sees the call. If Kamailio receives a call from the VPN and forwards the call to the Asterisk server on it's 172.x

It looks like if I call Getvar and pass PJSIP_HEADERS() I can get the entire SIP header for a channel. I also read (on stackoverflow) that the PJSIP_HEADER function will only return the headers from the INVITE of the inbound channel. If that’s correct, how would I get the headers from the outbound channel (second leg of the bridged call) INVITE ? Or will PJSIP_HEADERS() in fact return the