[asterisk-users] PJSIP console messages with Zoiper
I'm running Asterisk 15.1.0 and in the process of converting my various SIP endpoints to use PJSIP. My Zoiper client causes the messages quoted below to show up on the CLI once per minute. Things seem to work OK, but I am curious because there seems to be no way to suppress the messages, and there are three per minute, clogging up the console. Thanks. b. ** snip ** -- Added contact 'sip:8005@64.184.17.216:51891;rinstance=21e198ab7d94a096' to AOR '8005' with expiration of 60 seconds == Contact 8005/sip:8005@64.184.17.216:51891;rinstance=21e198ab7d94a096 has been deleted == Contact 8005/sip:8005@64.184.17.216:51891;rinstance=21e198ab7d94a096 has been created -- Contact 8005/sip:8005@64.184.17.216:51891;rinstance=21e198ab7d94a096 is now Unknown. RTT: 0.000 msec -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PJSIP and Non Media Proxy
> On 6/11/2017, at 7:42 AM, Saint Michael wrote: > > I see here a big disconnect between Digium and the VOIP industry. 99% > of the VOIP entrepreneurs like me would need to avoid proxying the media. > Wow, that's quite a bold statement. I must be one of the 1% then because I'm a VoIP entrepreneur and I've had no need to do such thing. > I mean people like me buy and sale billion of minutes every day" > Granted, I don't "buy and sell billions of minutes every day"... By my calcs that would require some 11K+ channels running 24X7, and that's just for one billion per day! So I'm dubious of the claim, and that in turn makes me dubious of the quoted 99% figure too. Pete signature.asc Description: Message signed with OpenPGP -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PJSIP and Non Media Proxy
On Sun, Nov 5, 2017, at 02:42 PM, Saint Michael wrote: > Now that Joshua had the kindness to respond, I see here a big disconnect > between Digium and the VOIP industry. 99% of the VOIP entrepreneurs like > me > would need to avoid proxying the media. Would would Digium support and > bring in with such fanfare a channel like PJSIP that lacks the only thing > that 99% would need to do business in an efficient manner? I mean people > like me buy and sale billion of minutes every day, and most of my peers > gravitate towards Opensips and other solution that do not touch the > media. > Yesterday I had to roll back my sleeves and go back to the old sip > channel. > I would love to see Asterisk-PJSIP to find a way to act like a proxy. > This > would turn Asterisk into a real wholesale business tool, which is not, so > far. It's not the lack of this feature which drives people to using OpenSIPS or Kamailio for this use case. It's just fundamentally designed differently and better performant for that scenario. Asterisk isn't the best solution for everything everyone needs or wants, and that's okay. There are other projects (like those already mentioned) that are a better fit, and Asterisk can even play a part in there as an application server. I'm a firm believer in using the right tool for the right job even if it means that Asterisk isn't the right fit. Frustrated users are something I never want to see. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PJSIP and Non Media Proxy
Now that Joshua had the kindness to respond, I see here a big disconnect between Digium and the VOIP industry. 99% of the VOIP entrepreneurs like me would need to avoid proxying the media. Would would Digium support and bring in with such fanfare a channel like PJSIP that lacks the only thing that 99% would need to do business in an efficient manner? I mean people like me buy and sale billion of minutes every day, and most of my peers gravitate towards Opensips and other solution that do not touch the media. Yesterday I had to roll back my sleeves and go back to the old sip channel. I would love to see Asterisk-PJSIP to find a way to act like a proxy. This would turn Asterisk into a real wholesale business tool, which is not, so far. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PJSIP and Non Media Proxy
On Sun, Nov 5, 2017, at 07:16 AM, Saint Michael wrote: > Please correct me if I am wrong. With PJSIP there is no way for Asterisk > to > stay a OUT of the media path, while with the old SIP channel, using > directrtpsetup and directmedia, it just works. The issue I think is that > other servers do not accept reinvites or updates to redirect media, so > PJSIP will not be able to step out ever. Using the old sip channel, the > 200 > OK with SDP tells the calling side to talk direcly to the other side. > Is there a way to do this with PJSIP? There is no "directrtpsetup" equivalent in PJSIP. Even in chan_sip it was experimental and could break things depending on the codec payloads in use. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PJSIP and Non Media Proxy
Please correct me if I am wrong. With PJSIP there is no way for Asterisk to stay a OUT of the media path, while with the old SIP channel, using directrtpsetup and directmedia, it just works. The issue I think is that other servers do not accept reinvites or updates to redirect media, so PJSIP will not be able to step out ever. Using the old sip channel, the 200 OK with SDP tells the calling side to talk direcly to the other side. Is there a way to do this with PJSIP? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
