[asterisk-users] x-ast-orig-host - How is this IP taken ?
Hi list, We have a strange behavior: a customer Snom300 behind a public FW has contact like contact : sip:user@x.y.39.147:2048;x-ast-orig-host=169.254.252.1:2048 The phone can place calls but not receive any. Also, qualify give unreachable which seems correct when looking the x-ast-orig-host IP. Problem is that the local IP of this phone is 192.168.1.75 Question: how asterisk sets this IP ? It looks for us like a FW issue as we have other customers with approaching local network organisation and which are not facing this problem. Thanks for any hint. -- 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] call replicating
Hello, provider responded the behavior is intentional from their side. So this should be fixed in asterisk. The pjsip cleanly does not do any unregistrations where it should. Marek 2020-06-07 12:30 GMT+02:00, Marek Greško : > Hello, > > I found the problem and also the workaround. > > Clearly, since it was working with chan_sip it should not be dialplan > problem, but sip stack problem. > > I have line=yes set up. After asterisk restart the old registration is > not unregistered and new one is registered with different line value. > Then incoming invites and qualify requests are sent to all the > registrations and there the problem lies. > > I am thinking of how could asterisk prevent such situations. > > 1. I think it should send unregistration requests on shutdown. > > 2. I think it should keep the database of active registrations and > unregister and reregister all of them during startup in case some of > them remain active after unclean shutdown. > > Also probably provider side should be fixed? > > Thanks for your insight. > > Marek > > > 2020-06-05 19:29 GMT+02:00, Doug Lytle : >> On 6/5/20 12:24 PM, Marek Greško wrote: >>> How can this behavior been overriden? I do not expect this is problem >>> on provider side, since it was working normally using chan_sip. >> >> Console output and dial plan snippets are always useful when diagnosing, >> >> 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
Re: [asterisk-users] CDR mysql: timeout when remote database unavailable
On Wednesday 10 June 2020 at 16:33:53, Doug Lytle wrote: > >>> Instead, the call still terminates if mysql cannot be reached. > > I just tested this, I'm using cdr_odbc, by shutting down mysql and I did > not experience the call being dropped. That sounds different to me. The OP is asking about a call which starts at a time when MySQL is not available. It sounds as though you made a call, then terminated MySQL, and the call did not terminate. > The console logged the mysql failure, but the call continued. > > You may want to consider moving to cdr_odbc instead. I certainly agree with that suggestion / recommendation. Antony. -- "Black holes are where God divided by zero." - Steven Wright 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
Re: [asterisk-users] CDR mysql: timeout when remote database unavailable
>>> Instead, the call still terminates if mysql cannot be reached. I just tested this, I'm using cdr_odbc, by shutting down mysql and I did not experience the call being dropped. The console logged the mysql failure, but the call continued. You may want to consider moving to cdr_odbc instead. 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
Re: [asterisk-users] CDR mysql: timeout when remote database unavailable
> On 2020-06-08 16:37, Sean Bright wrote: > > In the case of cdr_mysql, the connect timeout is configurable by putting > the following in cdr_mysql.conf: > > [global] > timeout = 5 ; Set connect timeout to 5 seconds OK, so i changed the timeout to 2 sec, but it does not have the desired effect. I expected, if mysql cannot be reached, after 2 seconds the call will progress normally. Instead, the call still terminates if mysql cannot be reached. Is it possible to set in asterisk, if mysql is unreachable then skip the step and simply ignore mysql logging ? -- _ -- 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] asterisk hints can be in multiple states; most sip NOTIFY dialogs only send one state
On 10/06/2020 15:40, Joshua C. Colp wrote: You wouldn't be able to access such information from ast_sip_presence_exten_state_to_str, that function is strictly for taking in instructions/data and producing the output. The user of it would need to pass in a value to turn on this new behavior. From that level the ast_sip_exten_state_data structure can optionally have a subscription, which itself has the endpoint that was used to establish the subscription. Ok, I'll look at that when I get around to moving to chan_pjsip. I'm very slow at changing working configurations. :) -- _ -- 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] Support or 12 Bogen HS201C analog phones
Any suggestions for an analog to SIP gateway that would support these devices. I need 12 phones supported. I cannot use any analog cards in this case - different buildings. Thanks, Jerry -- _ -- 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] asterisk hints can be in multiple states; most sip NOTIFY dialogs only send one state
On Wed, Jun 10, 2020 at 10:27 AM John Hughes wrote: > Asterisk can know that one of the attached phones is both "ringing" and > "on the phone". > > However the sip NOTIFY it sends out to interested parties can only > communicate one state, for example with pidf+xml it can either send > "Ringing" or "On the phone" and so it sends "Ringing". > > This makes the "busy lights" less than useful, if a call makes multiple > phones ring you can't tell, looking at the busy lights, which ones are > busy, and so less likely to answer. > > In the chan_sip configuration there is an option "notifyringing": > > notifyringing > > *notifyringing* enables or disables notifications for the RINGING state > when an extension is already INUSE. Only affects subscriptions using the > *dialog-info* event package. Option can be configured in the general > section only. It cannot be set per-peer. > > As the doc says this only applies to dialog-info style NOTIFY, not the > pidf+xml format my phones use. > > Here is a patch that makes notifyringing work for pidf+xml. > > Generalising it for other formats is left as an exercise for the reader. > > Of course chan_sip is obsolete. How might this be done for chan_pjsip? > Parts of the code are similar, but the layering is vastly different. How > could the ast_sip_presence_exten_state_to_str function in > res/res_pjsip/presence_xml.c get at the pjsip configuration? > You wouldn't be able to access such information from ast_sip_presence_exten_state_to_str, that function is strictly for taking in instructions/data and producing the output. The user of it would need to pass in a value to turn on this new behavior. From that level the ast_sip_exten_state_data structure can optionally have a subscription, which itself has the endpoint that was used to establish the subscription. -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at www.sangoma.com and 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] asterisk hints can be in multiple states; most sip NOTIFY dialogs only send one state
Asterisk can know that one of the attached phones is both "ringing" and "on the phone". However the sip NOTIFY it sends out to interested parties can only communicate one state, for example with pidf+xml it can either send "Ringing" or "On the phone" and so it sends "Ringing". This makes the "busy lights" less than useful, if a call makes multiple phones ring you can't tell, looking at the busy lights, which ones are busy, and so less likely to answer. In the chan_sip configuration there is an option "notifyringing": notifyringing *notifyringing* enables or disables notifications for the RINGING state when an extension is already INUSE. Only affects subscriptions using the *dialog-info* event package. Option can be configured in the general section only. It cannot be set per-peer. As the doc says this only applies to dialog-info style NOTIFY, not the pidf+xml format my phones use. Here is a patch that makes notifyringing work for pidf+xml. Generalising it for other formats is left as an exercise for the reader. Of course chan_sip is obsolete. How might this be done for chan_pjsip? Parts of the code are similar, but the layering is vastly different. How could the ast_sip_presence_exten_state_to_str function in res/res_pjsip/presence_xml.c get at the pjsip configuration? Description: make "notifyringing" work with pidf+xml If sip config specifies notifyringing=no and an extension is in a call then we send out "On a call" instead of "Ringing" so people can see who is not going to pick the call up. Author: John Hughes Last-Update: 2020-06-09 --- asterisk-13.14.1~dfsg.orig/channels/chan_sip.c +++ asterisk-13.14.1~dfsg/channels/chan_sip.c @@ -14966,7 +14966,10 @@ static void state_notify_build_xml(struc statestring = (sip_cfg.notifyringing) ? "early" : "confirmed"; local_state = NOTIFY_INUSE; pidfstate = "busy"; - pidfnote = "Ringing"; + if (subscribed == PIDF_XML && !sip_cfg.notifyringing) + pidfnote = "On the phone"; + else + pidfnote = "Ringing"; break; case AST_EXTENSION_RINGING: statestring = "early"; -- _ -- 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
