Re: [asterisk-users] PJSIP: 481 Call/Transaction Does Not Exist (only) for MESSAGE method
On Sun, 2019-02-17 at 17:31 -0500, Brian J. Murrell wrote: > I have a PJSIP trunk set up which works fine for voice. I can call > out > and I receive calls from it once it registers. > > What isn't working though is receiving MESSAGE (i.e. SIP SIMPLE) > events. It was working earlier today but I seem to have done > something > as I was enabling voice on the trunk to mess it up. On receiving of > a > MESSAGE, my Asterisk sends a 401 for the ITSP to authenticate it's > message, which it does, to which my Asterisk responds with a "481 > Call/Transaction Does Not Exist" and displays nothing at all in the > console. Nobody has any idea about this? :-( Cheers, b. signature.asc Description: This is a digitally signed message part -- _ -- 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] configure SRTP port range?
On Fri, Feb 22, 2019, at 2:48 PM, hw wrote: > > Hi, > > when trying to use SRTP, I can see UDP traffic from phones to the > asterisk server being dropped be the firewall on arbitrary ports. There is no separate port range used for SRTP, and Asterisk does not control the port that the phone uses for sending to Asterisk. That's up to the endpoint. > > Where do I configure the SRTP port range (like the rtp port range)? > > Why aren't the clients talking to each other directly but apparenty try > to send the SRTP traffic to the server? DIrect media with SRTP is not supported. All media when SRTP goes through Asterisk. -- Joshua C. Colp Digium - A Sangoma Company | 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] configure SRTP port range?
Hi, when trying to use SRTP, I can see UDP traffic from phones to the asterisk server being dropped be the firewall on arbitrary ports. Where do I configure the SRTP port range (like the rtp port range)? Why aren't the clients talking to each other directly but apparenty try to send the SRTP traffic to the server? That the traffic is being blocked by the firewall is probably the reason why I have no audio when using SRTP ... -- _ -- 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] Gigaset C610 IP error with PJSIP
Hi, We upgraded an Asterisk 11 server to 16.1.1, going from chan_sip to pjsip, on a site using Gigaset phones. They are registring well despite the fact that we get a lot of errors like [Feb 22 18:30:07] ERROR[1556]: pjproject: : sip_transport. Error processing 367 bytes packet from UDP 192.168.1.108:5060 : PJSIP syntax error exception when parsing 'To' header on line 4 col 51: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.250:5060;rport=5060;branch=z9hG4bKPjf89448cc-aeed-47be-b9a3-4e9b79f064bd From: ;tag=9a7c1775-fe8d-4c67-a8a4-c44e5d473742 To: ;tag=8`6b0664,gd9e,5b76,`9`5,b55d4e562653 Call-ID: fd359d3b-1102-4cd5-929a-46a04a95389d CSeq: 28722 NOTIFY Content-Length: 0 It seems that the comma in the tag is the origin of this behavior. Is this an Asterisk problem or a Gigaset one ? Daniel -- _ -- 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] SRTP with accounts in mysql database
On Friday 22 February 2019 at 18:05:26, hw wrote: > Hi, > > the ecnryption tutorial[1] says to add 'encryption=yes' into sip.conf > for a peer to use SRTP. > > I have all the account information in a mysql database in a table called > `sippeers` asterisk uses. The table doesn't seem to have a column for > this option. > > How can I specify it; where in the database do I put it? Can I just add > a column `ecryption` and put 'yes' (or no) into it? Yes - so long as you spell it correctly :) http://lists.digium.com/pipermail/asterisk-dev/2013-February/058581.html Antony. -- You can tell that the day just isn't going right when you find yourself using the telephone before the toilet. Please reply to the list; please *don't* CC me. -- _ -- 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] SRTP with accounts in mysql database
Hi, the ecnryption tutorial[1] says to add 'encryption=yes' into sip.conf for a peer to use SRTP. I have all the account information in a mysql database in a table called `sippeers` asterisk uses. The table doesn't seem to have a column for this option. How can I specify it; where in the database do I put it? Can I just add a column `ecryption` and put 'yes' (or no) into it? [1]: https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial -- _ -- 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] ARI set multiple channels vars at once
Hi all, we were wondering if there is a possibility to set multiple channels vars using ARI at once. Docu says it is not, but usually you need to set more than one variable. according docu: https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Channels+REST+API#Asterisk16ChannelsRESTAPI-setChannelVar If you use non staged channel Create it is possible to attach a set of variables. If you use staged dial, you first create a channel than you set the variables one at a time. It would be much easier if you could attach the channel vars to the channel create or to set multiple variables in one command. is there a way to do this and we haven't found it yet? many Thanks Jöran vinzens -- Jöran Vinzens - vinz...@sipgate.de sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois Steuernummer: 106/5724/7147, Umsatzsteuer-ID: DE219349391 www.sipgate.de - www.sipgate.co.uk -- _ -- 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
