Re: [asterisk-users] Dial() from the console?
On Wed, Jan 11, 2017 at 07:31:31AM -0500, Doug Lytle wrote: > >>> On Jan 11, 2017, at 7:20 AM, Thufir Hawat hawat.thu...@gmail.com wrote: > > >>> Can I dial directly from the asterisk console with the Dial() application? > > > console dial number@context Note, however, that it is a different thing. It uses a specific device: the "console". That is: the speaker and microphone of the system Asterisk is running on. For that to be available you need to have one of chan_alsa, chan_console or chan_oss loaded and working. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- 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] 256 bit SRTP ciphers in Asterisk 14.x , only works for outbound call ?
Greetings, If I understand correctly, Asterisk 14 introduced support for some new SRTP ciphers (including some 256 bit ones), previously only two 128 bit ciphers were supported. Using Asterisk 14, I was able to make a call from a softphone (Groundwire) with a 256 bit cipher suite on SRTP, which is great. However, I don’t see any way to specify with PJSIP (or chan sip) , the cipher suite which should be used when Asterisk calls the endpoint/phone. So I believe this means Asterisk would always use 128 bit SRTP to call the phone. Then, if you have 256 bit only ciphers allowed on the phone, the call fails. Perhaps this is just not documented, or may not be implemented yet. Anyone have a thought? Thank you,. Kevin Long -- _ -- 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 status check at DB level
On Wed, Jan 11, 2017, at 12:59 PM, Ahmed Munir wrote: > Thanks. > > But I was not able to find the records in 'ps_contacts' table. As per my > 'ps_aors' entries, I'm enabling and using only following fields below; > > id: 20010 > max_contacts: 1 > remove_existing: yes > qualify_frequecy: 10 What is your sorcery.conf? Have you configured it such that your contacts are put into the database? -- 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
Re: [asterisk-users] PJSIP status check at DB level
Thanks. But I was not able to find the records in 'ps_contacts' table. As per my 'ps_aors' entries, I'm enabling and using only following fields below; id: 20010 max_contacts: 1 remove_existing: yes qualify_frequecy: 10 Please advise if there any more parameters I may need to set in 'ps_aors' table. Date: Tue, 10 Jan 2017 10:18:50 -0400 > From: Joshua Colp > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] PJSIP status check at DB level > (Realtime) > Message-ID: > <1484057930.4162258.843096561.6368d...@webmail.messagingengine.com > > > Content-Type: text/plain; charset="utf-8" > > On Tue, Jan 10, 2017, at 10:11 AM, Ahmed Munir wrote: > > Hi, > > > > I would like to know how to check PJSIP status for endpoints at DB level > > (realtime) just like in chan_sip? Like on chan_sip when sip extension > > gets > > register regseconds, ipaddr, fullcontact & many more parameters get > > updated > > at DB level. > > > > The version of asterisk I'm using is 13.8 cert 4 > > Registered devices are added as contacts to the "ps_contacts" table. > This is because there can be multiple of them. You would need to look in > that table and make the association (based on AOR). > > -- > 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 > > -- Regards, Ahmed Munir Chohan -- _ -- 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] sip:p...@noname.com
On Wed, Jan 11, 2017, at 11:09 AM, Thufir Hawat wrote: > The SIP trace shows messages from what I took to be a suspicious > connection from sip:p...@noname.com so I added that IP address to IP > tables...but then anveo showed as unreachable so I removed that rule. > > Yes, I'm running fail2ban. > > What are these messages from sip:p...@noname.com? The domain name alone > set off alarm bells for me. (I was looking for my own registration > attempts when I turned on SIP debugging.) It's just a keepalive from their end to determine that your SIP server is still reachable and to keep open any NAT mappings so you are reachable. -- 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] hangup locked channels
Hello, When I run "core show channels verbose", I seen around 10 locked channels that are lasting hundreds of hours (I haven't restarted Asterisk for more than 7 weeks). I try to hangup those channels using "hangup request ", but nothing happens. What could I do (besides restarting Asterisk)? Thanks Dov -- _ -- 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] sip show [general]?
Hi You can do sip show settings On Jan 11, 2017 5:32 AM, "Thufir Hawat" wrote: > I appreciate that the console lets you see the details for a peer with > "sip show peer foo". Certainly, I can look in sip.conf to see the > [general] context, but can I output those settings, and only those > settings, to the console? > > > > thanks, > > Thufir > > -- > _ > -- 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 > -- _ -- 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] sip show [general]?
'sip show settings' may do what you want. On Wed, Jan 11, 2017 at 3:32 AM, Thufir Hawat wrote: > I appreciate that the console lets you see the details for a peer with > "sip show peer foo". Certainly, I can look in sip.conf to see the > [general] context, but can I output those settings, and only those > settings, to the console? > > > > thanks, > > Thufir > > -- > _ > -- 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 > -- A human being should be able to change a diaper, plan an invasion, butcher a hog, conn a ship, design a building, write a sonnet, balance accounts, build a wall, set a bone, comfort the dying, take orders, give orders, cooperate, act alone, solve equations, analyze a new problem, pitch manure, program a computer, cook a tasty meal, fight efficiently, die gallantly. Specialization is for insects. ---Heinlein -- _ -- 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] Dial() from the console?
On Wed, 11 Jan 2017, Doug Lytle wrote: On Jan 11, 2017, at 7:20 AM, Thufir Hawat hawat.thu...@gmail.com wrote: Can I dial directly from the asterisk console with the Dial() application? console dial number@context Thanks, that's much more intuitive :) -Thufir -- _ -- 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] sip:p...@noname.com
The SIP trace shows messages from what I took to be a suspicious connection from sip:p...@noname.com so I added that IP address to IP tables...but then anveo showed as unreachable so I removed that rule. Yes, I'm running fail2ban. What are these messages from sip:p...@noname.com? The domain name alone set off alarm bells for me. (I was looking for my own registration attempts when I turned on SIP debugging.) SIP trace: fqdn*CLI> fqdn*CLI> sip set debug on SIP Debugging enabled fqdn*CLI> <--- SIP read from UDP:67.212.84.21:5010 ---> OPTIONS sip:s...@xxx.xxx.xxx.xxx:5060 SIP/2.0 Via: SIP/2.0/UDP 67.212.84.21:5010;branch=0 From: sip:p...@noname.com;tag=uloc-5875e606-bf5-dea1e-52564b36-00fe47a3 To: sip:s...@xxx.xxx.xxx.xxx:5060 Call-ID: cb004ab7-97b14601-e7ade23@67.212.84.21 CSeq: 1 OPTIONS Content-Length: 0 <-> --- (7 headers 0 lines) --- Sending to 67.212.84.21:5010 (NAT) Looking for s in default (domain xxx.xxx.xxx.xxx) <--- Transmitting (NAT) to 67.212.84.21:5010 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 67.212.84.21:5010;branch=0;received=67.212.84.21;rport=5010 From: sip:p...@noname.com;tag=uloc-5875e606-bf5-dea1e-52564b36-00fe47a3 To: sip:s...@xxx.xxx.xxx.xxx:5060;tag=as5f595fce Call-ID: cb004ab7-97b14601-e7ade23@67.212.84.21 CSeq: 1 OPTIONS Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <> Scheduling destruction of SIP dialog 'cb004ab7-97b14601-e7ade23@67.212.84.21' in 32000 ms (Method: OPTIONS) Really destroying SIP dialog 'cb004ab7-90004601-06ade23@67.212.84.21' Method: OPTIONS Reliably Transmitting (NAT) to 67.212.84.21:5010: OPTIONS sip:sip.anveo.com SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK601302be;rport Max-Forwards: 70 From: "asterisk" ;tag=as194a0afc To: Contact: Call-ID: 6e15b7534a1b1e852464e02a5fca4...@xxx.xxx.xxx.xxx:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4 Date: Wed, 11 Jan 2017 14:56:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:67.212.84.21:5010 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK601302be;rport=5060;received=xxx.xxx.xxx.xxx From: "asterisk" ;tag=as194a0afc To: ;tag=a1766e4537c6d6082807422b1789bf43.b9ae Call-ID: 6e15b7534a1b1e852464e02a5fca4...@xxx.xxx.xxx.xxx:5060 CSeq: 102 OPTIONS Server: Anv Edge Proxy 3.5 Content-Length: 0 <-> --- (8 headers 0 lines) --- Really destroying SIP dialog '6e15b7534a1b1e852464e02a5fca4...@xxx.xxx.xxx.xxx:5060' Method: OPTIONS fqdn*CLI> sip set debug off SIP Debugging Disabled fqdn*CLI> fqdn*CLI> sip show peers Name/username HostDyn Forcerport ComediaACL Port Status Description anveo/1234567890 67.212.84.21Yes Yes5010 OK (78 ms) demo_alice(Unspecified)D Yes Yes0UNKNOWN demo_bob (Unspecified)D Yes Yes0UNKNOWN piter (Unspecified)D Yes Yes0UNKNOWN thufir(Unspecified)D Yes Yes0UNKNOWN 5 sip peers [Monitored: 1 online, 4 offline Unmonitored: 0 online, 0 offline] fqdn*CLI> fqdn*CLI> sip show peer anveo * Name : anveo Description : Secret : MD5Secret: Remote Secret: Context : from-anveo Record On feature : automon Record Off feature : automon Subscr.Cont. : Language : Tonezone : AMA flags: Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : Named Callgr : Nam. Pickupgr: MOH Suggest : Mailbox : VM Extension : asterisk LastMsgsSent : 0/0 Call limit : 0 Max forwards : 0 Dynamic : No Callerid : "" <> MaxCallBR: 384 kbps Expire : -1 Insecure : port,invite Force rport : Yes Symmetric RTP: Yes ACL : No DirectMedACL : No T.38 support : No T.38 EC mode : Unknown T.38 MaxDtgrm: 4294967295 DirectMedia : Yes PromiscRedir : No User=Phone : No Video Support: No Text Support : No Ign SDP ver : No Trust RPID : No Send RPID: No Path support : No Path : N/A TrustIDOutbnd: Legacy Subscriptions: Yes Overlap dial : Yes DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : sip.anveo.com Addr->IP : 67.212.84.21:5010 Defaddr->IP : (null) Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: 1234567890 SIP Options : (none) Codecs : (ulaw) Auto-Framing : No
Re: [asterisk-users] Dial() from the console?
>>> On Jan 11, 2017, at 7:20 AM, Thufir Hawat hawat.thu...@gmail.com wrote: >>> Can I dial directly from the asterisk console with the Dial() application? console dial number@context Doug -- _ -- 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] Dial() from the console?
Can I dial directly from the asterisk console with the Dial() application? or, is channel originate preferred: channel originate SIP/thufir extension 18003569377@outbound thanks, Thufir -- _ -- 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] sip show [general]?
I appreciate that the console lets you see the details for a peer with "sip show peer foo". Certainly, I can look in sip.conf to see the [general] context, but can I output those settings, and only those settings, to the console? thanks, Thufir -- _ -- 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
